From 0bf79ef2c303cc70d036c9fb355aeb468e8efb62 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:08:52 -0600 Subject: ASoC: wm8903: init GPIOs during I2C probe not codec probe This allows the GPIOs to be available as soon as the I2C device has probed, which in turn enables machine drivers to request the GPIOs in their probe(), rather than deferring this to their ASoC machine init function, i.e. after the whole sound card has been constructed, and hence the WM8903 codec is available. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 86b8a29..f6a3fc5 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2,7 +2,7 @@ * wm8903.c -- WM8903 ALSA SoC Audio driver * * Copyright 2008 Wolfson Microelectronics - * Copyright 2011 NVIDIA, Inc. + * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown * @@ -116,6 +116,7 @@ static const struct reg_default wm8903_reg_defaults[] = { struct wm8903_priv { struct wm8903_platform_data *pdata; + struct device *dev; struct snd_soc_codec *codec; struct regmap *regmap; @@ -1774,7 +1775,6 @@ static int wm8903_gpio_request(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1782,8 +1782,8 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) val = (WM8903_GPn_FN_GPIO_INPUT << WM8903_GP1_FN_SHIFT) | WM8903_GP1_DIR; - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1793,10 +1793,9 @@ static int wm8903_gpio_direction_in(struct gpio_chip *chip, unsigned offset) static int wm8903_gpio_get(struct gpio_chip *chip, unsigned offset) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - int reg; + unsigned int reg; - reg = snd_soc_read(codec, WM8903_GPIO_CONTROL_1 + offset); + regmap_read(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, ®); return (reg & WM8903_GP1_LVL_MASK) >> WM8903_GP1_LVL_SHIFT; } @@ -1805,7 +1804,6 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; unsigned int mask, val; int ret; @@ -1813,8 +1811,8 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, val = (WM8903_GPn_FN_GPIO_OUTPUT << WM8903_GP1_FN_SHIFT) | (value << WM8903_GP2_LVL_SHIFT); - ret = snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - mask, val); + ret = regmap_update_bits(wm8903->regmap, + WM8903_GPIO_CONTROL_1 + offset, mask, val); if (ret < 0) return ret; @@ -1824,11 +1822,10 @@ static int wm8903_gpio_direction_out(struct gpio_chip *chip, static void wm8903_gpio_set(struct gpio_chip *chip, unsigned offset, int value) { struct wm8903_priv *wm8903 = gpio_to_wm8903(chip); - struct snd_soc_codec *codec = wm8903->codec; - snd_soc_update_bits(codec, WM8903_GPIO_CONTROL_1 + offset, - WM8903_GP1_LVL_MASK, - !!value << WM8903_GP1_LVL_SHIFT); + regmap_update_bits(wm8903->regmap, WM8903_GPIO_CONTROL_1 + offset, + WM8903_GP1_LVL_MASK, + !!value << WM8903_GP1_LVL_SHIFT); } static struct gpio_chip wm8903_template_chip = { @@ -1842,15 +1839,14 @@ static struct gpio_chip wm8903_template_chip = { .can_sleep = 1, }; -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; int ret; wm8903->gpio_chip = wm8903_template_chip; wm8903->gpio_chip.ngpio = WM8903_NUM_GPIO; - wm8903->gpio_chip.dev = codec->dev; + wm8903->gpio_chip.dev = wm8903->dev; if (pdata->gpio_base) wm8903->gpio_chip.base = pdata->gpio_base; @@ -1859,24 +1855,23 @@ static void wm8903_init_gpio(struct snd_soc_codec *codec) ret = gpiochip_add(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to add GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to add GPIOs: %d\n", ret); } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int ret; ret = gpiochip_remove(&wm8903->gpio_chip); if (ret != 0) - dev_err(codec->dev, "Failed to remove GPIOs: %d\n", ret); + dev_err(wm8903->dev, "Failed to remove GPIOs: %d\n", ret); } #else -static void wm8903_init_gpio(struct snd_soc_codec *codec) +static void wm8903_init_gpio(struct wm8903_priv *wm8903) { } -static void wm8903_free_gpio(struct snd_soc_codec *codec) +static void wm8903_free_gpio(struct wm8903_priv *wm8903) { } #endif @@ -2000,8 +1995,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); - wm8903_init_gpio(codec); - return ret; } @@ -2010,7 +2003,6 @@ static int wm8903_remove(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - wm8903_free_gpio(codec); wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); if (wm8903->irq) free_irq(wm8903->irq, codec); @@ -2130,6 +2122,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, GFP_KERNEL); if (wm8903 == NULL) return -ENOMEM; + wm8903->dev = &i2c->dev; wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap); if (IS_ERR(wm8903->regmap)) { @@ -2189,6 +2182,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, /* Reset the device */ regmap_write(wm8903->regmap, WM8903_SW_RESET_AND_ID, 0x8903); + wm8903_init_gpio(wm8903); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) @@ -2204,6 +2199,7 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct wm8903_priv *wm8903 = i2c_get_clientdata(client); + wm8903_free_gpio(wm8903); regmap_exit(wm8903->regmap); snd_soc_unregister_codec(&client->dev); -- cgit v0.10.2 From f51022f1aedc4d1a02d0dfa8fde47f6a8291f618 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:08:54 -0600 Subject: ASoC: tegra+wm8903: move all GPIO setup into probe Now that deferred probe exists, we can parse device tree and request GPIOs from probe(), rather than deferring this to the DAI link's init(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 0b0df49..a8a3103 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -245,80 +245,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - struct device_node *np = card->dev->of_node; - int ret; - - if (card->dev->platform_data) { - memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); - } else if (np) { - /* - * This part must be in init() rather than probe() in order to - * guarantee that the WM8903 has been probed, and hence its - * GPIO controller registered, which is a pre-condition for - * of_get_named_gpio() to be able to map the phandles in the - * properties to the controller node. Given this, all - * pdata handling is in init() for consistency. - */ - pdata->gpio_spkr_en = of_get_named_gpio(np, - "nvidia,spkr-en-gpios", 0); - pdata->gpio_hp_mute = of_get_named_gpio(np, - "nvidia,hp-mute-gpios", 0); - pdata->gpio_hp_det = of_get_named_gpio(np, - "nvidia,hp-det-gpios", 0); - pdata->gpio_int_mic_en = of_get_named_gpio(np, - "nvidia,int-mic-en-gpios", 0); - pdata->gpio_ext_mic_en = of_get_named_gpio(np, - "nvidia,ext-mic-en-gpios", 0); - } else { - dev_err(card->dev, "No platform data supplied\n"); - return -EINVAL; - } - - if (gpio_is_valid(pdata->gpio_spkr_en)) { - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); - if (ret) { - dev_err(card->dev, "cannot get spkr_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_SPKR_EN; - - gpio_direction_output(pdata->gpio_spkr_en, 0); - } - - if (gpio_is_valid(pdata->gpio_hp_mute)) { - ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); - if (ret) { - dev_err(card->dev, "cannot get hp_mute gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_HP_MUTE; - - gpio_direction_output(pdata->gpio_hp_mute, 1); - } - - if (gpio_is_valid(pdata->gpio_int_mic_en)) { - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get int_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_INT_MIC_EN; - - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); - } - - if (gpio_is_valid(pdata->gpio_ext_mic_en)) { - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); - if (ret) { - dev_err(card->dev, "cannot get ext_mic_en gpio\n"); - return ret; - } - machine->gpio_requested |= GPIO_EXT_MIC_EN; - - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); - } if (gpio_is_valid(pdata->gpio_hp_det)) { tegra_wm8903_hp_jack_gpio.gpio = pdata->gpio_hp_det; @@ -372,8 +298,10 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_wm8903; struct tegra_wm8903 *machine; + struct tegra_wm8903_platform_data *pdata; int ret; if (!pdev->dev.platform_data && !pdev->dev.of_node) { @@ -388,12 +316,42 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) ret = -ENOMEM; goto err; } + pdata = &machine->pdata; card->dev = &pdev->dev; platform_set_drvdata(pdev, card); snd_soc_card_set_drvdata(card, machine); - if (pdev->dev.of_node) { + if (pdev->dev.platform_data) { + memcpy(pdata, card->dev->platform_data, sizeof(*pdata)); + } else if (np) { + pdata->gpio_spkr_en = of_get_named_gpio(np, + "nvidia,spkr-en-gpios", 0); + if (pdata->gpio_spkr_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_mute = of_get_named_gpio(np, + "nvidia,hp-mute-gpios", 0); + if (pdata->gpio_hp_mute == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_hp_det = of_get_named_gpio(np, + "nvidia,hp-det-gpios", 0); + if (pdata->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_int_mic_en = of_get_named_gpio(np, + "nvidia,int-mic-en-gpios", 0); + if (pdata->gpio_int_mic_en == -ENODEV) + return -EPROBE_DEFER; + + pdata->gpio_ext_mic_en = of_get_named_gpio(np, + "nvidia,ext-mic-en-gpios", 0); + if (pdata->gpio_ext_mic_en == -ENODEV) + return -EPROBE_DEFER; + } + + if (np) { ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; @@ -404,8 +362,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) goto err; tegra_wm8903_dai.codec_name = NULL; - tegra_wm8903_dai.codec_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,audio-codec", 0); + tegra_wm8903_dai.codec_of_node = of_parse_phandle(np, + "nvidia,audio-codec", 0); if (!tegra_wm8903_dai.codec_of_node) { dev_err(&pdev->dev, "Property 'nvidia,audio-codec' missing or invalid\n"); @@ -414,8 +372,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } tegra_wm8903_dai.cpu_dai_name = NULL; - tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle( - pdev->dev.of_node, "nvidia,i2s-controller", 0); + tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(np, + "nvidia,i2s-controller", 0); if (!tegra_wm8903_dai.cpu_dai_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); @@ -442,6 +400,52 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } } + if (gpio_is_valid(pdata->gpio_spkr_en)) { + ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); + if (ret) { + dev_err(card->dev, "cannot get spkr_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_SPKR_EN; + + gpio_direction_output(pdata->gpio_spkr_en, 0); + } + + if (gpio_is_valid(pdata->gpio_hp_mute)) { + ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); + if (ret) { + dev_err(card->dev, "cannot get hp_mute gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_HP_MUTE; + + gpio_direction_output(pdata->gpio_hp_mute, 1); + } + + if (gpio_is_valid(pdata->gpio_int_mic_en)) { + ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get int_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_INT_MIC_EN; + + /* Disable int mic; enable signal is active-high */ + gpio_direction_output(pdata->gpio_int_mic_en, 0); + } + + if (gpio_is_valid(pdata->gpio_ext_mic_en)) { + ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); + if (ret) { + dev_err(card->dev, "cannot get ext_mic_en gpio\n"); + return ret; + } + machine->gpio_requested |= GPIO_EXT_MIC_EN; + + /* Enable ext mic; enable signal is active-low */ + gpio_direction_output(pdata->gpio_ext_mic_en, 0); + } + ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) goto err; -- cgit v0.10.2 From e2d287c179a12a6069bc3b746e2e34edcddf81b3 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:08:55 -0600 Subject: ASoC: tegra+wm8903: Use devm_gpio_request_one By using this function, the driver no longer needs to explicitly free the GPIOs. Hence, we can also remove the flags we use to track whether we allocated these GPIOs. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index a8a3103..5ef2063 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -50,10 +50,6 @@ #define DRV_NAME "tegra-snd-wm8903" -#define GPIO_SPKR_EN BIT(0) -#define GPIO_HP_MUTE BIT(1) -#define GPIO_INT_MIC_EN BIT(2) -#define GPIO_EXT_MIC_EN BIT(3) #define GPIO_HP_DET BIT(4) struct tegra_wm8903 { @@ -401,49 +397,41 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } if (gpio_is_valid(pdata->gpio_spkr_en)) { - ret = gpio_request(pdata->gpio_spkr_en, "spkr_en"); + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_spkr_en, + GPIOF_OUT_INIT_LOW, "spkr_en"); if (ret) { dev_err(card->dev, "cannot get spkr_en gpio\n"); return ret; } - machine->gpio_requested |= GPIO_SPKR_EN; - - gpio_direction_output(pdata->gpio_spkr_en, 0); } if (gpio_is_valid(pdata->gpio_hp_mute)) { - ret = gpio_request(pdata->gpio_hp_mute, "hp_mute"); + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_hp_mute, + GPIOF_OUT_INIT_HIGH, "hp_mute"); if (ret) { dev_err(card->dev, "cannot get hp_mute gpio\n"); return ret; } - machine->gpio_requested |= GPIO_HP_MUTE; - - gpio_direction_output(pdata->gpio_hp_mute, 1); } if (gpio_is_valid(pdata->gpio_int_mic_en)) { - ret = gpio_request(pdata->gpio_int_mic_en, "int_mic_en"); + /* Disable int mic; enable signal is active-high */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_int_mic_en, + GPIOF_OUT_INIT_LOW, "int_mic_en"); if (ret) { dev_err(card->dev, "cannot get int_mic_en gpio\n"); return ret; } - machine->gpio_requested |= GPIO_INT_MIC_EN; - - /* Disable int mic; enable signal is active-high */ - gpio_direction_output(pdata->gpio_int_mic_en, 0); } if (gpio_is_valid(pdata->gpio_ext_mic_en)) { - ret = gpio_request(pdata->gpio_ext_mic_en, "ext_mic_en"); + /* Enable ext mic; enable signal is active-low */ + ret = devm_gpio_request_one(&pdev->dev, pdata->gpio_ext_mic_en, + GPIOF_OUT_INIT_LOW, "ext_mic_en"); if (ret) { dev_err(card->dev, "cannot get ext_mic_en gpio\n"); return ret; } - machine->gpio_requested |= GPIO_EXT_MIC_EN; - - /* Enable ext mic; enable signal is active-low */ - gpio_direction_output(pdata->gpio_ext_mic_en, 0); } ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); @@ -469,21 +457,11 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - struct tegra_wm8903_platform_data *pdata = &machine->pdata; if (machine->gpio_requested & GPIO_HP_DET) snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); - if (machine->gpio_requested & GPIO_EXT_MIC_EN) - gpio_free(pdata->gpio_ext_mic_en); - if (machine->gpio_requested & GPIO_INT_MIC_EN) - gpio_free(pdata->gpio_int_mic_en); - if (machine->gpio_requested & GPIO_HP_MUTE) - gpio_free(pdata->gpio_hp_mute); - if (machine->gpio_requested & GPIO_SPKR_EN) - gpio_free(pdata->gpio_spkr_en); - machine->gpio_requested = 0; snd_soc_unregister_card(card); -- cgit v0.10.2 From e44fbbd45896e684d44391aaf881dd3e36bd1a16 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:08:56 -0600 Subject: ASoC: tegra+wm8903: unconditionally free jack GPIOs in remove The headphone jack GPIOs are added/initialized in the DAI link's init() method, and hence in theory may not always have been added before remove() is called in some unusual cases. In order to prevent calling snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the code kept track of the initialization state to avoid the free call when necessary. However, it appears that snd_soc_jack_free_gpios() is robust in the face of being called without snd_soc_jack_add_gpios() first succeeding, so there is little point manually tracking this information. Hence, remove the tracking code. Almost all other machine drivers already operate this way. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 5ef2063..9059525 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -50,12 +50,9 @@ #define DRV_NAME "tegra-snd-wm8903" -#define GPIO_HP_DET BIT(4) - struct tegra_wm8903 { struct tegra_wm8903_platform_data pdata; struct tegra_asoc_utils_data util_data; - int gpio_requested; }; static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, @@ -252,7 +249,6 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) snd_soc_jack_add_gpios(&tegra_wm8903_hp_jack, 1, &tegra_wm8903_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_jack_new(codec, "Mic Jack", SND_JACK_MICROPHONE, @@ -458,10 +454,8 @@ static int __devexit tegra_wm8903_driver_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, - 1, - &tegra_wm8903_hp_jack_gpio); + snd_soc_jack_free_gpios(&tegra_wm8903_hp_jack, 1, + &tegra_wm8903_hp_jack_gpio); snd_soc_unregister_card(card); -- cgit v0.10.2 From aef9a37c01a63a132d43d65d231dfe515d0f918a Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:09:51 -0600 Subject: ASoC: tegra+alc5632: move all GPIO setup into probe Now that deferred probe exists, we can parse device tree and request GPIOs from probe(), rather than deferring this to the DAI link's init(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 32de700..facf6f0 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -1,5 +1,5 @@ /* - * tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver +* tegra_alc5632.c -- Toshiba AC100(PAZ00) machine ASoC driver * * Copyright (C) 2011 The AC100 Kernel Team * Copyright (C) 2012 - NVIDIA, Inc. @@ -110,7 +110,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct device_node *np = codec->card->dev->of_node; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET, @@ -119,8 +118,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) ARRAY_SIZE(tegra_alc5632_hs_jack_pins), tegra_alc5632_hs_jack_pins); - machine->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); - if (gpio_is_valid(machine->gpio_hp_det)) { tegra_alc5632_hp_jack_gpio.gpio = machine->gpio_hp_det; snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, @@ -159,6 +156,7 @@ static struct snd_soc_card snd_soc_tegra_alc5632 = { static __devinit int tegra_alc5632_probe(struct platform_device *pdev) { + struct device_node *np = pdev->dev.of_node; struct snd_soc_card *card = &snd_soc_tegra_alc5632; struct tegra_alc5632 *alc5632; int ret; @@ -181,6 +179,10 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } + alc5632->gpio_hp_det = of_get_named_gpio(np, "nvidia,hp-det-gpios", 0); + if (alc5632->gpio_hp_det == -ENODEV) + return -EPROBE_DEFER; + ret = snd_soc_of_parse_card_name(card, "nvidia,model"); if (ret) goto err; -- cgit v0.10.2 From 9f6328d910ef8df8176ed433aa2de037eba1f656 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:09:52 -0600 Subject: ASoC: tegra+alc5632: unconditionally free jack GPIOs in remove The headphone jack GPIOs are added/initialized in the DAI link's init() method, and hence in theory may not always have been added before remove() is called in some unusual cases. In order to prevent calling snd_soc_jack_free_gpios() if snd_soc_jack_add_gpios() had not been, the code kept track of the initialization state to avoid the free call when necessary. However, it appears that snd_soc_jack_free_gpios() is robust in the face of being called without snd_soc_jack_add_gpios() first succeeding, so there is little point manually tracking this information. Hence, remove the tracking code. All other machine drivers already operate this way. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index facf6f0..1566957 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -33,11 +33,8 @@ #define DRV_NAME "tegra-alc5632" -#define GPIO_HP_DET BIT(0) - struct tegra_alc5632 { struct tegra_asoc_utils_data util_data; - int gpio_requested; int gpio_hp_det; }; @@ -123,7 +120,6 @@ static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) snd_soc_jack_add_gpios(&tegra_alc5632_hs_jack, 1, &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested |= GPIO_HP_DET; } snd_soc_dapm_force_enable_pin(dapm, "MICBIAS1"); @@ -236,11 +232,8 @@ static int __devexit tegra_alc5632_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(card); - if (machine->gpio_requested & GPIO_HP_DET) - snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, - 1, - &tegra_alc5632_hp_jack_gpio); - machine->gpio_requested = 0; + snd_soc_jack_free_gpios(&tegra_alc5632_hs_jack, 1, + &tegra_alc5632_hp_jack_gpio); snd_soc_unregister_card(card); -- cgit v0.10.2 From 14df415a38234aa483219335bc6c1ee899b85e10 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:08:53 -0600 Subject: ASoC: tegra+wm8903: simplify gpio tests in widget callbacks By the time any widget callbacks could be called, if the GPIO ID they will manipulate is valid, it must have already been requested, or the card would have failed to probe or initialize. So, testing for GPIO validity is equivalent to testing whether the GPIO was successfully requested at this point in the code. Making this change will allow later patches to remove the gpio_requested variable. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 9059525..1fd6a41 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -153,7 +153,7 @@ static int tegra_wm8903_event_int_spk(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_SPKR_EN)) + if (!gpio_is_valid(pdata->gpio_spkr_en)) return 0; gpio_set_value_cansleep(pdata->gpio_spkr_en, @@ -170,7 +170,7 @@ static int tegra_wm8903_event_hp(struct snd_soc_dapm_widget *w, struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); struct tegra_wm8903_platform_data *pdata = &machine->pdata; - if (!(machine->gpio_requested & GPIO_HP_MUTE)) + if (!gpio_is_valid(pdata->gpio_hp_mute)) return 0; gpio_set_value_cansleep(pdata->gpio_hp_mute, -- cgit v0.10.2 From b350ecbe4c2e4639ed3a716ec67accb744e4417d Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Tue, 22 May 2012 16:11:19 -0600 Subject: ASoC: tegra+wm8903: remove non-DT support for Seaboard In kernel 3.6, Seaboard will only be supported when booting using device tree; the board files are being removed. Hence, remove the non-DT support for Seaboard and derivatives Kaen and Aebl from the audio driver. Harmony is the only remaining board supported by this driver when not using DT. This support is currently scheduled for removal in 3.7. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 1fd6a41..b75e0e8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -28,8 +28,6 @@ * */ -#include - #include #include #include @@ -196,37 +194,6 @@ static const struct snd_soc_dapm_route harmony_audio_map[] = { {"IN1L", NULL, "Mic Jack"}, }; -static const struct snd_soc_dapm_route seaboard_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route kaen_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "ROP"}, - {"Int Spk", NULL, "RON"}, - {"Int Spk", NULL, "LOP"}, - {"Int Spk", NULL, "LON"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN2R", NULL, "Mic Jack"}, -}; - -static const struct snd_soc_dapm_route aebl_audio_map[] = { - {"Headphone Jack", NULL, "HPOUTR"}, - {"Headphone Jack", NULL, "HPOUTL"}, - {"Int Spk", NULL, "LINEOUTR"}, - {"Int Spk", NULL, "LINEOUTL"}, - {"Mic Jack", NULL, "MICBIAS"}, - {"IN1R", NULL, "Mic Jack"}, -}; - static const struct snd_kcontrol_new tegra_wm8903_controls[] = { SOC_DAPM_PIN_SWITCH("Int Spk"), }; @@ -377,19 +344,8 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) tegra_wm8903_dai.platform_of_node = tegra_wm8903_dai.cpu_dai_of_node; } else { - if (machine_is_harmony()) { - card->dapm_routes = harmony_audio_map; - card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); - } else if (machine_is_seaboard()) { - card->dapm_routes = seaboard_audio_map; - card->num_dapm_routes = ARRAY_SIZE(seaboard_audio_map); - } else if (machine_is_kaen()) { - card->dapm_routes = kaen_audio_map; - card->num_dapm_routes = ARRAY_SIZE(kaen_audio_map); - } else { - card->dapm_routes = aebl_audio_map; - card->num_dapm_routes = ARRAY_SIZE(aebl_audio_map); - } + card->dapm_routes = harmony_audio_map; + card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); } if (gpio_is_valid(pdata->gpio_spkr_en)) { -- cgit v0.10.2 From 656baaebf92ae9b16644c7e10a273d8dfe1ba1f6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 23 May 2012 12:39:07 +0100 Subject: ASoC: codecs: Refresh copyrights for Wolfson drivers Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a75c376..52f0a19 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -1,7 +1,7 @@ /* * wm2000.c -- WM2000 ALSA Soc Audio driver * - * Copyright 2008-2010 Wolfson Microelectronics PLC. + * Copyright 2008-2011 Wolfson Microelectronics PLC. * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm5100-tables.c b/sound/soc/codecs/wm5100-tables.c index e167207..e239f4b 100644 --- a/sound/soc/codecs/wm5100-tables.c +++ b/sound/soc/codecs/wm5100-tables.c @@ -1,7 +1,7 @@ /* * wm5100-tables.c -- WM5100 ALSA SoC Audio driver data * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index cb6d537..3823af3 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1,7 +1,7 @@ /* * wm5100.c -- WM5100 ALSA SoC Audio driver * - * Copyright 2011 Wolfson Microelectronics plc + * Copyright 2011-2 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index 555ee14..e782a5a 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1,7 +1,7 @@ /* * wm8350.c -- WM8350 ALSA SoC audio driver * - * Copyright (C) 2007, 2008 Wolfson Microelectronics PLC. + * Copyright (C) 2007-12 Wolfson Microelectronics PLC. * * Author: Liam Girdwood * diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 5dc31eb..5d277a9 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1,7 +1,7 @@ /* * wm8400.c -- WM8400 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * Author: Mark Brown * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 2112851..7c68226 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -1,7 +1,7 @@ /* * wm8580.c -- WM8580 ALSA Soc Audio driver * - * Copyright 2008, 2009 Wolfson Microelectronics PLC. + * Copyright 2008-11 Wolfson Microelectronics PLC. * * This program is free software; you can redistribute it and/or modify it * under the terms of the GNU General Public License as published by the diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 9d1b9b02..bb1d269 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -2,6 +2,7 @@ * wm8731.c -- WM8731 ALSA SoC Audio driver * * Copyright 2005 Openedhand Ltd. + * Copyright 2006-12 Wolfson Microelectronics, plc * * Author: Richard Purdie * diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 6e849cb..35f3d23 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -1,7 +1,7 @@ /* * wm8741.c -- WM8741 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-1 Wolfson Microelectronics plc * * Author: Ian Lartey * diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index a26482c..13bff87 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1,7 +1,7 @@ /* * wm8753.c -- WM8753 ALSA Soc Audio driver * - * Copyright 2003 Wolfson Microelectronics PLC. + * Copyright 2003-11 Wolfson Microelectronics PLC. * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index a19db5a..879c356 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -1,7 +1,7 @@ /* * wm8776.c -- WM8776 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 6bd1b76..c088020 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -1,7 +1,7 @@ /* * wm8804.c -- WM8804 S/PDIF transceiver driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-11 Wolfson Microelectronics plc * * Author: Dimitris Papastamos * diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index f6a3fc5..304b5cf 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,7 +1,7 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008 Wolfson Microelectronics + * Copyright 2008-11 Wolfson Microelectronics * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d..db94d10 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1,7 +1,7 @@ /* * wm8904.c -- WM8904 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 8bc659d..96518ac 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -1,6 +1,8 @@ /* * wm8960.c -- WM8960 ALSA SoC Audio driver * + * Copyright 2007-11 Wolfson Microelectronics, plc + * * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 05ea7c2..01edbcc 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -1,6 +1,8 @@ /* * wm8961.c -- WM8961 ALSA SoC Audio driver * + * Copyright 2009-10 Wolfson Microelectronics, plc + * * Author: Mark Brown * * This program is free software; you can redistribute it and/or modify diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 0cfce99..27da4d7 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1,7 +1,7 @@ /* * wm8962.c -- WM8962 ALSA SoC Audio driver * - * Copyright 2010 Wolfson Microelectronics plc + * Copyright 2010-2 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 36acfcc..9fd80d6 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1,7 +1,7 @@ /* * wm8993.c -- WM8993 ALSA SoC audio driver * - * Copyright 2009, 2010 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 993639d..5d4d7df 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1,7 +1,7 @@ /* * wm8994.c -- WM8994 ALSA SoC Audio driver * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8af422e..efc4e9d 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1,7 +1,7 @@ /* * wm8996.c - WM8996 audio codec interface * - * Copyright 2011 Wolfson Microelectronics PLC. + * Copyright 2011-2 Wolfson Microelectronics PLC. * Author: Mark Brown * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 9328270..2de74e1 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -3,7 +3,7 @@ * * Author: Mark Brown * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License version 2 as diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index 4b263b6..2c2346f 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -1,7 +1,7 @@ /* * ALSA SoC WM9090 driver * - * Copyright 2009, 2010 Wolfson Microelectronics + * Copyright 2009-12 Wolfson Microelectronics * * Author: Mark Brown * diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index a154141..099e6ec 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -1,7 +1,7 @@ /* * wm9712.c -- ALSA Soc WM9712 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-12 Wolfson Microelectronics PLC. * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index 2d22cc7..3eb19fb 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1,7 +1,7 @@ /* * wm9713.c -- ALSA Soc WM9713 codec support * - * Copyright 2006 Wolfson Microelectronics PLC. + * Copyright 2006-10 Wolfson Microelectronics PLC. * Author: Liam Girdwood * * This program is free software; you can redistribute it and/or modify it diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index dfe957a..61baa48 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -1,7 +1,7 @@ /* * wm_hubs.c -- WM8993/4 common code * - * Copyright 2009 Wolfson Microelectronics plc + * Copyright 2009-12 Wolfson Microelectronics plc * * Author: Mark Brown * -- cgit v0.10.2 From 1aad779fccdbb4d79af7b9de93dfd2bfe807e052 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 24 May 2012 15:26:03 +0200 Subject: ALSA: pcm: Add debug-print helper function Adds a function getting the stream-name as a string for a specific stream. Signed-off-by: Ola Lilja Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 0d11128..a55d5db 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1073,4 +1073,15 @@ static inline void snd_pcm_limit_isa_dma_size(int dma, size_t *max) const char *snd_pcm_format_name(snd_pcm_format_t format); +/** + * Get a string naming the direction of a stream + */ +static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream) +{ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + return "Playback"; + else + return "Capture"; +} + #endif /* __SOUND_PCM_H */ -- cgit v0.10.2 From d7e7eb91551ad99244b989d71d092cb0375648fa Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 24 May 2012 15:26:25 +0200 Subject: ASoC: core: Add widget SND_SOC_DAPM_CLOCK_SUPPLY Adds a supply-widget variant for connection to the clock-framework. This widget-type corresponds to the variant for regulators. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e3833d9..05559e5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -229,6 +229,10 @@ struct device; { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert, \ .event = wevent, .event_flags = wflags} +#define SND_SOC_DAPM_CLOCK_SUPPLY(wname) \ +{ .id = snd_soc_dapm_clock_supply, .name = wname, \ + .reg = SND_SOC_NOPM, .event = dapm_clock_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } /* generic widgets */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ @@ -245,6 +249,7 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ #define SOC_DAPM_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ @@ -327,6 +332,8 @@ int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -432,6 +439,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ snd_soc_dapm_regulator_supply, /* external regulator */ + snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ snd_soc_dapm_aif_out, /* audio interface output */ snd_soc_dapm_siggen, /* signal generator */ @@ -537,6 +545,8 @@ struct snd_soc_dapm_widget { struct list_head dirty; int inputs; int outputs; + + struct clk *clk; }; struct snd_soc_dapm_update { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 90ee77d..3bb7a6f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include #include #include +#include #include #include #include @@ -51,6 +52,7 @@ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_supply] = 1, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_micbias] = 2, [snd_soc_dapm_dai_link] = 2, [snd_soc_dapm_dai] = 3, @@ -92,6 +94,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_aif_out] = 10, [snd_soc_dapm_dai] = 10, [snd_soc_dapm_dai_link] = 11, + [snd_soc_dapm_clock_supply] = 12, [snd_soc_dapm_regulator_supply] = 12, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_post] = 13, @@ -391,6 +394,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, case snd_soc_dapm_vmid: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -764,6 +768,7 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -850,6 +855,7 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, switch (widget->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: return 0; default: break; @@ -996,6 +1002,24 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for clock supply widget. + */ +int dapm_clock_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + if (!w->clk) + return -EIO; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + return clk_enable(w->clk); + } else { + clk_disable(w->clk); + return 0; + } +} +EXPORT_SYMBOL_GPL(dapm_clock_event); + static int dapm_widget_power_check(struct snd_soc_dapm_widget *w) { if (w->power_checked) @@ -1487,6 +1511,7 @@ static void dapm_widget_set_power(struct snd_soc_dapm_widget *w, bool power, switch (w->id) { case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: /* Supplies can't affect their outputs, only their inputs */ break; default: @@ -1587,6 +1612,7 @@ static int dapm_power_widgets(struct snd_soc_dapm_context *dapm, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) d->target_bias_level = SND_SOC_BIAS_STANDBY; @@ -1941,6 +1967,7 @@ static ssize_t dapm_widget_show(struct device *dev, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", w->name, w->power ? "On":"Off"); @@ -2187,6 +2214,7 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, case snd_soc_dapm_post: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: case snd_soc_dapm_aif_in: case snd_soc_dapm_aif_out: case snd_soc_dapm_dai: @@ -2873,6 +2901,15 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, return NULL; } break; + case snd_soc_dapm_clock_supply: + w->clk = clk_get(dapm->dev, w->name); + if (IS_ERR(w->clk)) { + ret = PTR_ERR(w->clk); + dev_err(dapm->dev, "Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; default: break; } @@ -2924,6 +2961,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_clock_supply: w->power_check = dapm_supply_check_power; break; case snd_soc_dapm_dai: -- cgit v0.10.2 From 01a0c1139c2bd075d005253093e7060022c5d0cb Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 24 May 2012 15:26:32 +0200 Subject: ASoC: Ux500: Add platform-driver Add platform-driver handling all DMA-activities. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 44cf434..1d38515 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -12,3 +12,10 @@ menuconfig SND_SOC_UX500 config SND_SOC_UX500_PLAT_MSP_I2S tristate depends on SND_SOC_UX500 + +config SND_SOC_UX500_PLAT_DMA + tristate "Platform - DB8500 (DMA)" + depends on SND_SOC_UX500 + select SND_SOC_DMAENGINE_PCM + help + Say Y if you want to enable the Ux500 platform-driver. diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 19974c5..4634bf0 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -2,3 +2,6 @@ snd-soc-ux500-plat-msp-i2s-objs := ux500_msp_dai.o ux500_msp_i2s.o obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o + +snd-soc-ux500-plat-dma-objs := ux500_pcm.o +obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c new file mode 100644 index 0000000..66b080e --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.c @@ -0,0 +1,318 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja , + * Roger Nilsson + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include + +#include +#include +#include +#include + +#include + +#include +#include +#include +#include + +#include "ux500_msp_i2s.h" +#include "ux500_pcm.h" + +static struct snd_pcm_hardware ux500_pcm_hw_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_PLAYBACK, + .rate_max = UX500_PLATFORM_MAX_RATE_PLAYBACK, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static struct snd_pcm_hardware ux500_pcm_hw_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_U16_BE, + .rates = SNDRV_PCM_RATE_KNOT, + .rate_min = UX500_PLATFORM_MIN_RATE_CAPTURE, + .rate_max = UX500_PLATFORM_MAX_RATE_CAPTURE, + .channels_min = UX500_PLATFORM_MIN_CHANNELS, + .channels_max = UX500_PLATFORM_MAX_CHANNELS, + .buffer_bytes_max = UX500_PLATFORM_BUFFER_BYTES_MAX, + .period_bytes_min = UX500_PLATFORM_PERIODS_BYTES_MIN, + .period_bytes_max = UX500_PLATFORM_PERIODS_BYTES_MAX, + .periods_min = UX500_PLATFORM_PERIODS_MIN, + .periods_max = UX500_PLATFORM_PERIODS_MAX, +}; + +static void ux500_pcm_dma_hw_free(struct device *dev, + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + + if (runtime->dma_area == NULL) + return; + + if (buf != &substream->dma_buffer) { + dma_free_coherent(buf->dev.dev, buf->bytes, buf->area, + buf->addr); + kfree(runtime->dma_buffer_p); + } + + snd_pcm_set_runtime_buffer(substream, NULL); +} + +static int ux500_pcm_open(struct snd_pcm_substream *substream) +{ + int stream_id = substream->pstr->stream; + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct device *dev = dai->dev; + int ret; + struct ux500_msp_dma_params *dma_params; + u16 per_data_width, mem_data_width; + struct stedma40_chan_cfg *dma_cfg; + + dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, + snd_pcm_stream_str(substream)); + + dev_dbg(dev, "%s: Set runtime hwparams.\n", __func__); + if (stream_id == SNDRV_PCM_STREAM_PLAYBACK) + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_playback); + else + snd_soc_set_runtime_hwparams(substream, + &ux500_pcm_hw_capture); + + /* ensure that buffer size is a multiple of period size */ + ret = snd_pcm_hw_constraint_integer(runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) { + dev_err(dev, "%s: Error: snd_pcm_hw_constraints failed (%d)\n", + __func__, ret); + return ret; + } + + dev_dbg(dev, "%s: Set hw-struct for %s.\n", __func__, + snd_pcm_stream_str(substream)); + runtime->hw = (stream_id == SNDRV_PCM_STREAM_PLAYBACK) ? + ux500_pcm_hw_playback : ux500_pcm_hw_capture; + + mem_data_width = STEDMA40_HALFWORD_WIDTH; + + dma_params = snd_soc_dai_get_dma_data(dai, substream); + switch (dma_params->data_size) { + case 32: + per_data_width = STEDMA40_WORD_WIDTH; + break; + case 16: + per_data_width = STEDMA40_HALFWORD_WIDTH; + break; + case 8: + per_data_width = STEDMA40_BYTE_WIDTH; + break; + default: + per_data_width = STEDMA40_WORD_WIDTH; + dev_warn(rtd->platform->dev, + "%s: Unknown data-size (%d)! Assuming 32 bits.\n", + __func__, dma_params->data_size); + } + + dma_cfg = dma_params->dma_cfg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + dma_cfg->src_info.data_width = mem_data_width; + dma_cfg->dst_info.data_width = per_data_width; + } else { + dma_cfg->src_info.data_width = per_data_width; + dma_cfg->dst_info.data_width = mem_data_width; + } + + + ret = snd_dmaengine_pcm_open(substream, stedma40_filter, dma_cfg); + if (ret) { + dev_dbg(dai->dev, + "%s: ERROR: snd_dmaengine_pcm_open failed (%d)!\n", + __func__, ret); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_cfg); + + return 0; +} + +static int ux500_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *dai = rtd->cpu_dai; + + dev_dbg(dai->dev, "%s: Enter\n", __func__); + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int ux500_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_dma_buffer *buf = runtime->dma_buffer_p; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret = 0; + int size; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + size = params_buffer_bytes(hw_params); + + if (buf) { + if (buf->bytes >= size) + goto out; + ux500_pcm_dma_hw_free(NULL, substream); + } + + if (substream->dma_buffer.area != NULL && + substream->dma_buffer.bytes >= size) { + buf = &substream->dma_buffer; + } else { + buf = kmalloc(sizeof(struct snd_dma_buffer), GFP_KERNEL); + if (!buf) + goto nomem; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = NULL; + buf->area = dma_alloc_coherent(NULL, size, &buf->addr, + GFP_KERNEL); + buf->bytes = size; + buf->private_data = NULL; + + if (!buf->area) + goto free; + } + snd_pcm_set_runtime_buffer(substream, buf); + ret = 1; + out: + runtime->dma_bytes = size; + return ret; + + free: + kfree(buf); + nomem: + return -ENOMEM; +} + +static int ux500_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter\n", __func__); + + ux500_pcm_dma_hw_free(NULL, substream); + + return 0; +} + +static int ux500_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + dev_dbg(rtd->platform->dev, "%s: Enter.\n", __func__); + + return dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); +} + +static struct snd_pcm_ops ux500_pcm_ops = { + .open = ux500_pcm_open, + .close = ux500_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = ux500_pcm_hw_params, + .hw_free = ux500_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = ux500_pcm_mmap +}; + +int ux500_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm *pcm = rtd->pcm; + + dev_dbg(rtd->platform->dev, "%s: Enter (id = '%s').\n", __func__, + pcm->id); + + pcm->info_flags = 0; + + return 0; +} + +static struct snd_soc_platform_driver ux500_pcm_soc_drv = { + .ops = &ux500_pcm_ops, + .pcm_new = ux500_pcm_new, +}; + +static int __devexit ux500_pcm_drv_probe(struct platform_device *pdev) +{ + int ret; + + ret = snd_soc_register_platform(&pdev->dev, &ux500_pcm_soc_drv); + if (ret < 0) { + dev_err(&pdev->dev, + "%s: ERROR: Failed to register platform '%s' (%d)!\n", + __func__, pdev->name, ret); + return ret; + } + + return 0; +} + +static int __devinit ux500_pcm_drv_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver ux500_pcm_driver = { + .driver = { + .name = "ux500-pcm", + .owner = THIS_MODULE, + }, + + .probe = ux500_pcm_drv_probe, + .remove = __devexit_p(ux500_pcm_drv_remove), +}; +module_platform_driver(ux500_pcm_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/ux500/ux500_pcm.h b/sound/soc/ux500/ux500_pcm.h new file mode 100644 index 0000000..77ed44d --- /dev/null +++ b/sound/soc/ux500/ux500_pcm.h @@ -0,0 +1,35 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja , + * Roger Nilsson + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ +#ifndef UX500_PCM_H +#define UX500_PCM_H + +#include + +#include + +#define UX500_PLATFORM_MIN_RATE_PLAYBACK 8000 +#define UX500_PLATFORM_MAX_RATE_PLAYBACK 48000 +#define UX500_PLATFORM_MIN_RATE_CAPTURE 8000 +#define UX500_PLATFORM_MAX_RATE_CAPTURE 48000 + +#define UX500_PLATFORM_MIN_CHANNELS 1 +#define UX500_PLATFORM_MAX_CHANNELS 8 + +#define UX500_PLATFORM_PERIODS_BYTES_MIN 128 +#define UX500_PLATFORM_PERIODS_BYTES_MAX (64 * PAGE_SIZE) +#define UX500_PLATFORM_PERIODS_MIN 2 +#define UX500_PLATFORM_PERIODS_MAX 48 +#define UX500_PLATFORM_BUFFER_BYTES_MAX (2048 * PAGE_SIZE) + +#endif -- cgit v0.10.2 From 5514efdfe0384576ef38c66b1672b6826696fbf3 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Mon, 28 May 2012 23:29:36 -0700 Subject: ASoC: fsi: use dmaengine helper functions This patch used dmaengine helper functions instead of using hand setting. And reduced local variables Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index 2ef9853..fcaa6b8 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1089,13 +1089,10 @@ static void fsi_dma_do_tasklet(unsigned long data) { struct fsi_stream *io = (struct fsi_stream *)data; struct fsi_priv *fsi = fsi_stream_to_priv(io); - struct dma_chan *chan; struct snd_soc_dai *dai; struct dma_async_tx_descriptor *desc; - struct scatterlist sg; struct snd_pcm_runtime *runtime; enum dma_data_direction dir; - dma_cookie_t cookie; int is_play = fsi_stream_is_play(fsi, io); int len; dma_addr_t buf; @@ -1104,7 +1101,6 @@ static void fsi_dma_do_tasklet(unsigned long data) return; dai = fsi_get_dai(io->substream); - chan = io->chan; runtime = io->substream->runtime; dir = is_play ? DMA_TO_DEVICE : DMA_FROM_DEVICE; len = samples_to_bytes(runtime, io->period_samples); @@ -1112,14 +1108,8 @@ static void fsi_dma_do_tasklet(unsigned long data) dma_sync_single_for_device(dai->dev, buf, len, dir); - sg_init_table(&sg, 1); - sg_set_page(&sg, pfn_to_page(PFN_DOWN(buf)), - len , offset_in_page(buf)); - sg_dma_address(&sg) = buf; - sg_dma_len(&sg) = len; - - desc = dmaengine_prep_slave_sg(chan, &sg, 1, dir, - DMA_PREP_INTERRUPT | DMA_CTRL_ACK); + desc = dmaengine_prep_slave_single(io->chan, buf, len, dir, + DMA_PREP_INTERRUPT | DMA_CTRL_ACK); if (!desc) { dev_err(dai->dev, "dmaengine_prep_slave_sg() fail\n"); return; @@ -1128,13 +1118,12 @@ static void fsi_dma_do_tasklet(unsigned long data) desc->callback = fsi_dma_complete; desc->callback_param = io; - cookie = desc->tx_submit(desc); - if (cookie < 0) { + if (dmaengine_submit(desc) < 0) { dev_err(dai->dev, "tx_submit() fail\n"); return; } - dma_async_issue_pending(chan); + dma_async_issue_pending(io->chan); /* * FIXME -- cgit v0.10.2 From b1226dc59d55ecde7fc9a338d8cb2a313821fac0 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Thu, 24 May 2012 23:56:19 -0700 Subject: ASoC: fsi: use PIO handler if DMA handler was invalid PIO handler is not good performance, but works on all platform. So, switch to PIO handler if DMA handler was invalid case. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index fcaa6b8..53486ff 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -247,7 +247,7 @@ struct fsi_priv { struct fsi_stream_handler { int (*init)(struct fsi_priv *fsi, struct fsi_stream *io); int (*quit)(struct fsi_priv *fsi, struct fsi_stream *io); - int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io); + int (*probe)(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev); int (*transfer)(struct fsi_priv *fsi, struct fsi_stream *io); int (*remove)(struct fsi_priv *fsi, struct fsi_stream *io); void (*start_stop)(struct fsi_priv *fsi, struct fsi_stream *io, @@ -571,16 +571,16 @@ static int fsi_stream_transfer(struct fsi_stream *io) #define fsi_stream_stop(fsi, io)\ fsi_stream_handler_call(io, start_stop, fsi, io, 0) -static int fsi_stream_probe(struct fsi_priv *fsi) +static int fsi_stream_probe(struct fsi_priv *fsi, struct device *dev) { struct fsi_stream *io; int ret1, ret2; io = &fsi->playback; - ret1 = fsi_stream_handler_call(io, probe, fsi, io); + ret1 = fsi_stream_handler_call(io, probe, fsi, io, dev); io = &fsi->capture; - ret2 = fsi_stream_handler_call(io, probe, fsi, io); + ret2 = fsi_stream_handler_call(io, probe, fsi, io, dev); if (ret1 < 0) return ret1; @@ -1173,7 +1173,7 @@ static void fsi_dma_push_start_stop(struct fsi_priv *fsi, struct fsi_stream *io, fsi_master_mask_set(master, CLK_RST, clk, (enable) ? clk : 0); } -static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) +static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io, struct device *dev) { dma_cap_mask_t mask; @@ -1181,8 +1181,19 @@ static int fsi_dma_probe(struct fsi_priv *fsi, struct fsi_stream *io) dma_cap_set(DMA_SLAVE, mask); io->chan = dma_request_channel(mask, fsi_dma_filter, &io->slave); - if (!io->chan) - return -EIO; + if (!io->chan) { + + /* switch to PIO handler */ + if (fsi_stream_is_play(fsi, io)) + fsi->playback.handler = &fsi_pio_push_handler; + else + fsi->capture.handler = &fsi_pio_pop_handler; + + dev_info(dev, "switch handler (dma => pio)\n"); + + /* probe again */ + return fsi_stream_probe(fsi, dev); + } tasklet_init(&io->tasklet, fsi_dma_do_tasklet, (unsigned long)io); @@ -1672,7 +1683,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsia.master = master; master->fsia.info = &info->port_a; fsi_handler_init(&master->fsia); - ret = fsi_stream_probe(&master->fsia); + ret = fsi_stream_probe(&master->fsia, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIA stream probe failed\n"); goto exit_iounmap; @@ -1683,7 +1694,7 @@ static int fsi_probe(struct platform_device *pdev) master->fsib.master = master; master->fsib.info = &info->port_b; fsi_handler_init(&master->fsib); - ret = fsi_stream_probe(&master->fsib); + ret = fsi_stream_probe(&master->fsib, &pdev->dev); if (ret < 0) { dev_err(&pdev->dev, "FSIB stream probe failed\n"); goto exit_fsia; -- cgit v0.10.2 From 14a95fe865c0b2ede6f386f52413f6396c010833 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Mon, 28 May 2012 22:09:02 +0300 Subject: ASoC: tlv320aic3x: Change Class-D amplifier gain control name ALSA mixers cannot classify this "Class-D Amplifier Gain" speaker output gain control as a playback control. Fix this by changing the name as "Class-D Playback Volume". Signed-off-by: Jarkko Nikula Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..58ef59d 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -368,7 +368,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { static DECLARE_TLV_DB_SCALE(classd_amp_tlv, 0, 600, 0); static const struct snd_kcontrol_new aic3x_classd_amp_gain_ctrl = - SOC_DOUBLE_TLV("Class-D Amplifier Gain", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); + SOC_DOUBLE_TLV("Class-D Playback Volume", CLASSD_CTRL, 6, 4, 3, 0, classd_amp_tlv); /* Left DAC Mux */ static const struct snd_kcontrol_new aic3x_left_dac_mux_controls = -- cgit v0.10.2 From 0561c1bf354c4a8230a1e0ada43362f54e60b2f0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 30 May 2012 13:20:17 +0100 Subject: ASoC: ac97: Remove empty remove() function Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c index 2023c74..ea06b83 100644 --- a/sound/soc/codecs/ac97.c +++ b/sound/soc/codecs/ac97.c @@ -91,11 +91,6 @@ static int ac97_soc_probe(struct snd_soc_codec *codec) return 0; } -static int ac97_soc_remove(struct snd_soc_codec *codec) -{ - return 0; -} - #ifdef CONFIG_PM static int ac97_soc_suspend(struct snd_soc_codec *codec) { @@ -119,7 +114,6 @@ static struct snd_soc_codec_driver soc_codec_dev_ac97 = { .write = ac97_write, .read = ac97_read, .probe = ac97_soc_probe, - .remove = ac97_soc_remove, .suspend = ac97_soc_suspend, .resume = ac97_soc_resume, }; -- cgit v0.10.2 From 51cc7ed3e378a60a3413a7e424f536e4dec3f39d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 May 2012 14:48:07 +0100 Subject: ASoC: wm2000: Add register readability information Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 52f0a19..78a148f 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -691,9 +691,39 @@ static int wm2000_resume(struct snd_soc_codec *codec) #define wm2000_resume NULL #endif +static bool wm2000_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case WM2000_REG_SYS_START: + case WM2000_REG_SPEECH_CLARITY: + case WM2000_REG_SYS_WATCHDOG: + case WM2000_REG_ANA_VMID_PD_TIME: + case WM2000_REG_ANA_VMID_PU_TIME: + case WM2000_REG_CAT_FLTR_INDX: + case WM2000_REG_CAT_GAIN_0: + case WM2000_REG_SYS_STATUS: + case WM2000_REG_SYS_MODE_CNTRL: + case WM2000_REG_SYS_START0: + case WM2000_REG_SYS_START1: + case WM2000_REG_ID1: + case WM2000_REG_ID2: + case WM2000_REG_REVISON: + case WM2000_REG_SYS_CTL1: + case WM2000_REG_SYS_CTL2: + case WM2000_REG_ANC_STAT: + case WM2000_REG_IF_CTL: + return true; + default: + return false; + } +} + static const struct regmap_config wm2000_regmap = { .reg_bits = 8, .val_bits = 8, + + .max_register = WM2000_REG_IF_CTL, + .readable_reg = wm2000_readable_reg, }; static int wm2000_probe(struct snd_soc_codec *codec) -- cgit v0.10.2 From 210cb67cb5b9f9a23b7ce91de50bab357440ba9d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 8 May 2012 17:46:36 +0100 Subject: ASoC: io: Use dev_get_regmap() if driver doesn't provide a regmap Less error prone and one less line of code in drivers. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 4d8dc6a..44d0174 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -142,6 +142,8 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, case SND_SOC_REGMAP: /* Device has made its own regmap arrangements */ codec->using_regmap = true; + if (!codec->control_data) + codec->control_data = dev_get_regmap(codec->dev, NULL); ret = regmap_get_val_bytes(codec->control_data); /* Errors are legitimate for non-integer byte multiples */ -- cgit v0.10.2 From bc92657a11c0982783979bbb84ceaf58ba222124 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 25 May 2012 18:22:11 -0600 Subject: ASoC: make snd_soc_dai_link more symmetrical Prior to this patch, the CPU side of a DAI link was specified using a single name. Often, this was the result of calling dev_name() on the device providing the DAI, but in the case of a CPU DAI driver that provided multiple DAIs, it needed to mix together both the device name and some device-relative name, in order to form a single globally unique name. However, the CODEC side of the DAI link was specified using separate fields for device (name or OF node) and device-relative DAI name. This patch allows the CPU side of a DAI link to be specified in the same way as the CODEC side, separating concepts of device and device-relative DAI name. I believe this will be important in multi-codec and/or dynamic PCM scenarios, where a single CPU driver provides multiple DAIs, while also booting using device tree, with accompanying desire not to hard-code the CPU side device's name into the original .cpu_dai_name field. Ideally, both the CPU DAI and CODEC DAI loops in soc_bind_dai_link() would now be identical. However, two things prevent that at present: 1) The need to save rtd->codec for the CODEC side, which means we have to search for the CODEC explicitly, and not just the CODEC side DAI. 2) Since we know the CODEC side DAI is part of a codec, and not just a standalone DAI, it's slightly more efficient to convert .codec_name/ .codec_of_node into a codec first, and then compare each DAI's .codec field, since this avoids strcmp() on each DAI's CODEC's name within the loop. However, the two loops are essentially semantically equivalent. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index c703871..23c4efb 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -785,13 +785,36 @@ struct snd_soc_dai_link { /* config - must be set by machine driver */ const char *name; /* Codec name */ const char *stream_name; /* Stream name */ - const char *codec_name; /* for multi-codec */ - const struct device_node *codec_of_node; - const char *platform_name; /* for multi-platform */ - const struct device_node *platform_of_node; + /* + * You MAY specify the link's CPU-side device, either by device name, + * or by DT/OF node, but not both. If this information is omitted, + * the CPU-side DAI is matched using .cpu_dai_name only, which hence + * must be globally unique. These fields are currently typically used + * only for codec to codec links, or systems using device tree. + */ + const char *cpu_name; + const struct device_node *cpu_of_node; + /* + * You MAY specify the DAI name of the CPU DAI. If this information is + * omitted, the CPU-side DAI is matched using .cpu_name/.cpu_of_node + * only, which only works well when that device exposes a single DAI. + */ const char *cpu_dai_name; - const struct device_node *cpu_dai_of_node; + /* + * You MUST specify the link's codec, either by device name, or by + * DT/OF node, but not both. + */ + const char *codec_name; + const struct device_node *codec_of_node; + /* You MUST specify the DAI name within the codec */ const char *codec_dai_name; + /* + * You MAY specify the link's platform/PCM/DMA driver, either by + * device name, or by DT/OF node, but not both. Some forms of link + * do not need a platform. + */ + const char *platform_name; + const struct device_node *platform_of_node; int be_id; /* optional ID for machine driver BE identification */ const struct snd_soc_pcm_stream *params; diff --git a/sound/soc/mxs/mxs-sgtl5000.c b/sound/soc/mxs/mxs-sgtl5000.c index 3e6e876..215113b 100644 --- a/sound/soc/mxs/mxs-sgtl5000.c +++ b/sound/soc/mxs/mxs-sgtl5000.c @@ -133,7 +133,7 @@ static int __devinit mxs_sgtl5000_probe_dt(struct platform_device *pdev) mxs_sgtl5000_dai[i].codec_name = NULL; mxs_sgtl5000_dai[i].codec_of_node = codec_np; mxs_sgtl5000_dai[i].cpu_dai_name = NULL; - mxs_sgtl5000_dai[i].cpu_dai_of_node = saif_np[i]; + mxs_sgtl5000_dai[i].cpu_of_node = saif_np[i]; mxs_sgtl5000_dai[i].platform_name = NULL; mxs_sgtl5000_dai[i].platform_of_node = saif_np[i]; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b37ee80..ec83505 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -812,13 +812,15 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) /* Find CPU DAI from registered DAIs*/ list_for_each_entry(cpu_dai, &dai_list, list) { - if (dai_link->cpu_dai_of_node) { - if (cpu_dai->dev->of_node != dai_link->cpu_dai_of_node) - continue; - } else { - if (strcmp(cpu_dai->name, dai_link->cpu_dai_name)) - continue; - } + if (dai_link->cpu_of_node && + (cpu_dai->dev->of_node != dai_link->cpu_of_node)) + continue; + if (dai_link->cpu_name && + strcmp(dev_name(cpu_dai->dev), dai_link->cpu_name)) + continue; + if (dai_link->cpu_dai_name && + strcmp(cpu_dai->name, dai_link->cpu_dai_name)) + continue; rtd->cpu_dai = cpu_dai; } @@ -3346,6 +3348,12 @@ int snd_soc_register_card(struct snd_soc_card *card) link->name); return -EINVAL; } + /* Codec DAI name must be specified */ + if (!link->codec_dai_name) { + dev_err(card->dev, "codec_dai_name not set for %s\n", + link->name); + return -EINVAL; + } /* * Platform may be specified by either name or OF node, but @@ -3358,12 +3366,24 @@ int snd_soc_register_card(struct snd_soc_card *card) } /* - * CPU DAI must be specified by 1 of name or OF node, - * not both or neither. + * CPU device may be specified by either name or OF node, but + * can be left unspecified, and will be matched based on DAI + * name alone.. + */ + if (link->cpu_name && link->cpu_of_node) { + dev_err(card->dev, + "Neither/both cpu name/of_node are set for %s\n", + link->name); + return -EINVAL; + } + /* + * At least one of CPU DAI name or CPU device name/node must be + * specified */ - if (!!link->cpu_dai_name == !!link->cpu_dai_of_node) { + if (!link->cpu_dai_name && + !(link->cpu_name || link->cpu_of_node)) { dev_err(card->dev, - "Neither/both cpu_dai name/of_node are set for %s\n", + "Neither cpu_dai_name nor cpu_name/of_node are set for %s\n", link->name); return -EINVAL; } diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 1566957..417b09b 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -197,16 +197,16 @@ static __devinit int tegra_alc5632_probe(struct platform_device *pdev) goto err; } - tegra_alc5632_dai.cpu_dai_of_node = of_parse_phandle( + tegra_alc5632_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_alc5632_dai.cpu_dai_of_node) { + if (!tegra_alc5632_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; goto err; } - tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_dai_of_node; + tegra_alc5632_dai.platform_of_node = tegra_alc5632_dai.cpu_of_node; ret = tegra_asoc_utils_init(&alc5632->util_data, &pdev->dev); if (ret) diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 4e77026..02bd5a8 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -157,9 +157,9 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) goto err; } - tegra_wm8753_dai.cpu_dai_of_node = of_parse_phandle( + tegra_wm8753_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!tegra_wm8753_dai.cpu_dai_of_node) { + if (!tegra_wm8753_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -167,7 +167,7 @@ static __devinit int tegra_wm8753_driver_probe(struct platform_device *pdev) } tegra_wm8753_dai.platform_of_node = - tegra_wm8753_dai.cpu_dai_of_node; + tegra_wm8753_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index b75e0e8..1fd71e5 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -331,9 +331,9 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) } tegra_wm8903_dai.cpu_dai_name = NULL; - tegra_wm8903_dai.cpu_dai_of_node = of_parse_phandle(np, + tegra_wm8903_dai.cpu_of_node = of_parse_phandle(np, "nvidia,i2s-controller", 0); - if (!tegra_wm8903_dai.cpu_dai_of_node) { + if (!tegra_wm8903_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -342,7 +342,7 @@ static __devinit int tegra_wm8903_driver_probe(struct platform_device *pdev) tegra_wm8903_dai.platform_name = NULL; tegra_wm8903_dai.platform_of_node = - tegra_wm8903_dai.cpu_dai_of_node; + tegra_wm8903_dai.cpu_of_node; } else { card->dapm_routes = harmony_audio_map; card->num_dapm_routes = ARRAY_SIZE(harmony_audio_map); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 4a8d5b6..5815430 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -162,9 +162,9 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) } trimslice_tlv320aic23_dai.cpu_dai_name = NULL; - trimslice_tlv320aic23_dai.cpu_dai_of_node = of_parse_phandle( + trimslice_tlv320aic23_dai.cpu_of_node = of_parse_phandle( pdev->dev.of_node, "nvidia,i2s-controller", 0); - if (!trimslice_tlv320aic23_dai.cpu_dai_of_node) { + if (!trimslice_tlv320aic23_dai.cpu_of_node) { dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing or invalid\n"); ret = -EINVAL; @@ -173,7 +173,7 @@ static __devinit int tegra_snd_trimslice_probe(struct platform_device *pdev) trimslice_tlv320aic23_dai.platform_name = NULL; trimslice_tlv320aic23_dai.platform_of_node = - trimslice_tlv320aic23_dai.cpu_dai_of_node; + trimslice_tlv320aic23_dai.cpu_of_node; } ret = tegra_asoc_utils_init(&trimslice->util_data, &pdev->dev); -- cgit v0.10.2 From 6c9d8cf6372ed2995a3d982f5c1f966e842101cc Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 31 May 2012 15:18:01 +0100 Subject: ASoC: core: Add single controls with specified range of values Control type added for cases where a specific range of values within a register are required for control. Added convenience macros: SOC_SINGLE_RANGE SOC_SINGLE_RANGE_TLV Added accessor implementations: snd_soc_info_volsw_range snd_soc_put_volsw_range snd_soc_get_volsw_range Signed-off-by: Michal Hajduk Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/include/sound/soc.h b/include/sound/soc.h index 23c4efb..e4348d2 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -47,6 +47,13 @@ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ .put = snd_soc_put_volsw, \ .private_value = SOC_SINGLE_VALUE(reg, shift, max, invert) } +#define SOC_SINGLE_RANGE(xname, xreg, xshift, xmin, xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, .get = snd_soc_get_volsw_range, \ + .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_SINGLE_TLV(xname, reg, shift, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -67,6 +74,16 @@ {.reg = xreg, .rreg = xreg, \ .shift = xshift, .rshift = xshift, \ .max = xmax, .min = xmin} } +#define SOC_SINGLE_RANGE_TLV(xname, xreg, xshift, xmin, xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = (unsigned long)&(struct soc_mixer_control) \ + {.reg = xreg, .shift = xshift, .min = xmin,\ + .max = xmax, .platform_max = xmax, .invert = xinvert} } #define SOC_DOUBLE(xname, reg, shift_left, shift_right, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw, \ @@ -460,6 +477,12 @@ int snd_soc_get_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo); +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol); int snd_soc_limit_volume(struct snd_soc_codec *codec, const char *name, int max); int snd_soc_bytes_info(struct snd_kcontrol *kcontrol, diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ec83505..3d803f3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2792,6 +2792,104 @@ int snd_soc_put_volsw_s8(struct snd_kcontrol *kcontrol, EXPORT_SYMBOL_GPL(snd_soc_put_volsw_s8); /** + * snd_soc_info_volsw_range - single mixer info callback with range. + * @kcontrol: mixer control + * @uinfo: control element information + * + * Callback to provide information, within a range, about a single + * mixer control. + * + * returns 0 for success. + */ +int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + int platform_max; + int min = mc->min; + + if (!mc->platform_max) + mc->platform_max = mc->max; + platform_max = mc->platform_max; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = platform_max - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_info_volsw_range); + +/** + * snd_soc_put_volsw_range - single mixer put value callback with range. + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to set the value, within a range, for a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + unsigned int val, val_mask; + + val = ((ucontrol->value.integer.value[0] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + return snd_soc_update_bits_locked(codec, reg, val_mask, val); +} +EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); + +/** + * snd_soc_get_volsw_range - single mixer get callback with range + * @kcontrol: mixer control + * @ucontrol: control element information + * + * Callback to get the value, within a range, of a single mixer control. + * + * Returns 0 for success. + */ +int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int reg = mc->reg; + unsigned int shift = mc->shift; + int min = mc->min; + int max = mc->max; + unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + + ucontrol->value.integer.value[0] = + (snd_soc_read(codec, reg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[0] = + max - ucontrol->value.integer.value[0]; + ucontrol->value.integer.value[0] = + ucontrol->value.integer.value[0] - min; + + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); + +/** * snd_soc_limit_volume - Set new limit to an existing volume control. * * @codec: where to look for the control -- cgit v0.10.2 From f59fef441753cdd07ffe7268b0801ec48cac7b1d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 30 Apr 2012 20:26:41 +0100 Subject: ASoC: wm8350: Convert to direct regmap API usage Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index e782a5a..d26c8ae 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -71,20 +71,6 @@ struct wm8350_data { int fll_freq_in; }; -static unsigned int wm8350_codec_read(struct snd_soc_codec *codec, - unsigned int reg) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_read(wm8350, reg); -} - -static int wm8350_codec_write(struct snd_soc_codec *codec, unsigned int reg, - unsigned int value) -{ - struct wm8350 *wm8350 = codec->control_data; - return wm8350_reg_write(wm8350, reg, value); -} - /* * Ramp OUT1 PGA volume to minimise pops at stream startup and shutdown. */ @@ -1519,7 +1505,9 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350; + codec->control_data = wm8350->regmap; + + snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); @@ -1629,8 +1617,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm8350 = { .remove = wm8350_codec_remove, .suspend = wm8350_suspend, .resume = wm8350_resume, - .read = wm8350_codec_read, - .write = wm8350_codec_write, .set_bias_level = wm8350_set_bias_level, .controls = wm8350_snd_controls, -- cgit v0.10.2 From 695594f1b79d3b88e99e28f06afaab32c4d65853 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Jun 2012 08:14:13 +0100 Subject: ASoC: dapm: Use devm_clk_get() Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3bb7a6f..a66379a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2902,7 +2902,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: - w->clk = clk_get(dapm->dev, w->name); + w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); dev_err(dapm->dev, "Failed to request %s: %d\n", -- cgit v0.10.2 From ec02995adad5a7b428f46c1a87fae1bc93d6dfe3 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Jun 2012 08:16:20 +0100 Subject: ASoC: dapm: Bodge for lack of a widely available clk API Reported-by: Stephen Rothwell Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index a66379a..39e8c2f 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1011,12 +1011,14 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, if (!w->clk) return -EIO; +#ifdef CONFIG_HAVE_CLK if (SND_SOC_DAPM_EVENT_ON(event)) { return clk_enable(w->clk); } else { clk_disable(w->clk); return 0; } +#endif } EXPORT_SYMBOL_GPL(dapm_clock_event); @@ -2902,6 +2904,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: +#ifdef CONFIG_HAVE_CLK w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); @@ -2909,6 +2912,9 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, w->name, ret); return NULL; } +#else + return NULL; +#endif break; default: break; -- cgit v0.10.2 From 014e5b56702575c5cd8ffc4b1a7924cfdfe0f1ea Mon Sep 17 00:00:00 2001 From: Richard Zhao Date: Mon, 4 Jun 2012 09:42:53 +0800 Subject: ASoC: fsl_ssi: convert to use devm_clk_get Signed-off-by: Richard Zhao Acked-by: Timur Tabi Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 4ed2afd..b10a427 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -725,7 +725,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = clk_get(&pdev->dev, NULL); + ssi_private->clk = devm_clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); @@ -842,10 +842,8 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) { + if (ssi_private->ssi_on_imx) clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); - } error_irq: free_irq(ssi_private->irq, ssi_private); @@ -871,7 +869,6 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) { platform_device_unregister(ssi_private->imx_pcm_pdev); clk_disable_unprepare(ssi_private->clk); - clk_put(ssi_private->clk); } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); -- cgit v0.10.2 From 7376bde8945fe20d35aa51f493a7e43b60a39dbe Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Sun, 3 Jun 2012 22:50:18 +0530 Subject: ASoC: cs42l52: Remove version.h header file inclusion version.h header file is no longer needed. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index a710941..ec03abc 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -14,7 +14,6 @@ #include #include -#include #include #include #include -- cgit v0.10.2 From 2bce133c3b00020f4bc146cea94ff5d4de9a8a0f Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Sun, 3 Jun 2012 22:58:40 +0530 Subject: ASoC: lm49453: Remove version.h header file inclusion version.h header file is no longer required. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..c1bc945 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -12,7 +12,6 @@ #include #include -#include #include #include #include -- cgit v0.10.2 From 2f989f7e9f5f9ba97535fa58f4240ec250d6b2df Mon Sep 17 00:00:00 2001 From: M R Swami Reddy Date: Fri, 1 Jun 2012 22:21:54 +0530 Subject: ASoC: Support TI Isabelle Audio driver ASoC: Support TI Isabelle Audio driver The Isabelle Audio IC is a complete low power high fidelity CODEC with integrated ADCs, DACs, decimation and interpolation filters, PLL, and power providers. This device supports 2 analog and 2 digital microphone channels, a mono earpiece driver, stereo class G headphone drivers with ultra low power and best SNR in the industry, stereo Class D speaker drivers, and 2 high performance Line drivers. The below patch is a basic driver code for TI Isabelle audio codec. The functionalities like headset detection, etc., will be included incrementally in the up-coming patches. Signed-off-by: Vishwas A Deshpande Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1e1613a..8b879c7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C select SND_SOC_DFBMCS320 + select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C select SND_SOC_LM49453 if I2C @@ -225,6 +226,9 @@ config SND_SOC_DFBMCS320 config SND_SOC_DMIC tristate +config SND_SOC_ISABELLE + tristate + config SND_SOC_LM49453 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fc27fec..e50811b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,7 @@ snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o +snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o snd-soc-lm4857-objs := lm4857.o @@ -134,6 +135,7 @@ obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o obj-$(CONFIG_SND_SOC_LM4857) += snd-soc-lm4857.o diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c new file mode 100644 index 0000000..b6921a8 --- /dev/null +++ b/sound/soc/codecs/isabelle.c @@ -0,0 +1,1179 @@ +/* + * isabelle.c - Low power high fidelity audio codec driver + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * + * Initially based on sound/soc/codecs/twl6040.c + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "isabelle.h" + + +/* Register default values for ISABELLE driver. */ +static struct reg_default isabelle_reg_defs[] = { + { 0, 0x00 }, + { 1, 0x00 }, + { 2, 0x00 }, + { 3, 0x00 }, + { 4, 0x00 }, + { 5, 0x00 }, + { 6, 0x00 }, + { 7, 0x00 }, + { 8, 0x00 }, + { 9, 0x00 }, + { 10, 0x00 }, + { 11, 0x00 }, + { 12, 0x00 }, + { 13, 0x00 }, + { 14, 0x00 }, + { 15, 0x00 }, + { 16, 0x00 }, + { 17, 0x00 }, + { 18, 0x00 }, + { 19, 0x00 }, + { 20, 0x00 }, + { 21, 0x02 }, + { 22, 0x02 }, + { 23, 0x02 }, + { 24, 0x02 }, + { 25, 0x0F }, + { 26, 0x8F }, + { 27, 0x0F }, + { 28, 0x8F }, + { 29, 0x00 }, + { 30, 0x00 }, + { 31, 0x00 }, + { 32, 0x00 }, + { 33, 0x00 }, + { 34, 0x00 }, + { 35, 0x00 }, + { 36, 0x00 }, + { 37, 0x00 }, + { 38, 0x00 }, + { 39, 0x00 }, + { 40, 0x00 }, + { 41, 0x00 }, + { 42, 0x00 }, + { 43, 0x00 }, + { 44, 0x00 }, + { 45, 0x00 }, + { 46, 0x00 }, + { 47, 0x00 }, + { 48, 0x00 }, + { 49, 0x00 }, + { 50, 0x00 }, + { 51, 0x00 }, + { 52, 0x00 }, + { 53, 0x00 }, + { 54, 0x00 }, + { 55, 0x00 }, + { 56, 0x00 }, + { 57, 0x00 }, + { 58, 0x00 }, + { 59, 0x00 }, + { 60, 0x00 }, + { 61, 0x00 }, + { 62, 0x00 }, + { 63, 0x00 }, + { 64, 0x00 }, + { 65, 0x00 }, + { 66, 0x00 }, + { 67, 0x00 }, + { 68, 0x00 }, + { 69, 0x90 }, + { 70, 0x90 }, + { 71, 0x90 }, + { 72, 0x00 }, + { 73, 0x00 }, + { 74, 0x00 }, + { 75, 0x00 }, + { 76, 0x00 }, + { 77, 0x00 }, + { 78, 0x00 }, + { 79, 0x00 }, + { 80, 0x00 }, + { 81, 0x00 }, + { 82, 0x00 }, + { 83, 0x00 }, + { 84, 0x00 }, + { 85, 0x07 }, + { 86, 0x00 }, + { 87, 0x00 }, + { 88, 0x00 }, + { 89, 0x07 }, + { 90, 0x80 }, + { 91, 0x07 }, + { 92, 0x07 }, + { 93, 0x00 }, + { 94, 0x00 }, + { 95, 0x00 }, + { 96, 0x00 }, + { 97, 0x00 }, + { 98, 0x00 }, + { 99, 0x00 }, +}; + +static const char *isabelle_rx1_texts[] = {"VRX1", "ARX1"}; +static const char *isabelle_rx2_texts[] = {"VRX2", "ARX2"}; + +static const struct soc_enum isabelle_rx1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 3, 1, isabelle_rx1_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 5, 1, isabelle_rx1_texts), +}; + +static const struct soc_enum isabelle_rx2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VOICE_HPF_CFG_REG, 2, 1, isabelle_rx2_texts), + SOC_ENUM_SINGLE(ISABELLE_AUDIO_HPF_CFG_REG, 4, 1, isabelle_rx2_texts), +}; + +/* Headset DAC playback switches */ +static const struct snd_kcontrol_new rx1_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx1_enum); + +static const struct snd_kcontrol_new rx2_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_rx2_enum); + +/* TX input selection */ +static const char *isabelle_atx_texts[] = {"AMIC1", "DMIC"}; +static const char *isabelle_vtx_texts[] = {"AMIC2", "DMIC"}; + +static const struct soc_enum isabelle_atx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 7, 1, isabelle_atx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_atx_texts), +}; + +static const struct soc_enum isabelle_vtx_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 6, 1, isabelle_vtx_texts), + SOC_ENUM_SINGLE(ISABELLE_DMIC_CFG_REG, 0, 1, isabelle_vtx_texts), +}; + +static const struct snd_kcontrol_new atx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_atx_enum); + +static const struct snd_kcontrol_new vtx_mux_controls = + SOC_DAPM_ENUM("Route", isabelle_vtx_enum); + +/* Left analog microphone selection */ +static const char *isabelle_amic1_texts[] = { + "Main Mic", "Headset Mic", "Aux/FM Left"}; + +/* Left analog microphone selection */ +static const char *isabelle_amic2_texts[] = {"Sub Mic", "Aux/FM Right"}; + +static const struct soc_enum isabelle_amic1_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 5, + ARRAY_SIZE(isabelle_amic1_texts), + isabelle_amic1_texts), +}; + +static const struct soc_enum isabelle_amic2_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_AMIC_CFG_REG, 4, + ARRAY_SIZE(isabelle_amic2_texts), + isabelle_amic2_texts), +}; + +static const struct snd_kcontrol_new amic1_control = + SOC_DAPM_ENUM("Route", isabelle_amic1_enum); + +static const struct snd_kcontrol_new amic2_control = + SOC_DAPM_ENUM("Route", isabelle_amic2_enum); + +static const char *isabelle_st_audio_texts[] = {"ATX1", "ATX2"}; + +static const char *isabelle_st_voice_texts[] = {"VTX1", "VTX2"}; + +static const struct soc_enum isabelle_st_audio_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA1_CFG_REG, 7, 1, + isabelle_st_audio_texts), + SOC_ENUM_SINGLE(ISABELLE_ATX_STPGA2_CFG_REG, 7, 1, + isabelle_st_audio_texts), +}; + +static const struct soc_enum isabelle_st_voice_enum[] = { + SOC_ENUM_SINGLE(ISABELLE_VTX_STPGA1_CFG_REG, 7, 1, + isabelle_st_voice_texts), + SOC_ENUM_SINGLE(ISABELLE_VTX2_STPGA2_CFG_REG, 7, 1, + isabelle_st_voice_texts), +}; + +static const struct snd_kcontrol_new st_audio_control = + SOC_DAPM_ENUM("Route", isabelle_st_audio_enum); + +static const struct snd_kcontrol_new st_voice_control = + SOC_DAPM_ENUM("Route", isabelle_st_voice_enum); + +/* Mixer controls */ +static const struct snd_kcontrol_new isabelle_hs_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1L Playback Switch", ISABELLE_HSDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hs_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC1R Playback Switch", ISABELLE_HSDRV_CFG1_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HSDRV_CFG1_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_HFLPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_HFLPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_hf_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2R Playback Switch", ISABELLE_HFRPGA_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_HFRPGA_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_ep_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC2L Playback Switch", ISABELLE_EARDRV_CFG1_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_EARDRV_CFG1_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_left_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3L Playback Switch", ISABELLE_LINEAMP_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("APGA1 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_aux_right_mixer_controls[] = { +SOC_DAPM_SINGLE("DAC3R Playback Switch", ISABELLE_LINEAMP_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("APGA2 Playback Switch", ISABELLE_LINEAMP_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga1_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA1LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 5, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 4, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2L_IN_SEL_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga2_right_mixer_controls[] = { +SOC_DAPM_SINGLE("USNC Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA2R_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_left_mixer_controls[] = { +SOC_DAPM_SINGLE("RX1 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 7, 1, 0), +SOC_DAPM_SINGLE("RX3 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 6, 1, 0), +SOC_DAPM_SINGLE("RX5 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 5, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_dpga3_right_mixer_controls[] = { +SOC_DAPM_SINGLE("RX2 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 3, 1, 0), +SOC_DAPM_SINGLE("RX4 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 2, 1, 0), +SOC_DAPM_SINGLE("RX6 Playback Switch", ISABELLE_DPGA3LR_IN_SEL_REG, 1, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx1_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx2_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx3_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 3, 1, 0), +SOC_DAPM_SINGLE("DL3 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 2, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx4_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 1, 1, 0), +SOC_DAPM_SINGLE("DL4 Playback Switch", ISABELLE_RX_INPUT_CFG_REG, 0, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx5_mixer_controls[] = { +SOC_DAPM_SINGLE("ST1 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 7, 1, 0), +SOC_DAPM_SINGLE("DL5 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 6, 1, 0), +}; + +static const struct snd_kcontrol_new isabelle_rx6_mixer_controls[] = { +SOC_DAPM_SINGLE("ST2 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 5, 1, 0), +SOC_DAPM_SINGLE("DL6 Playback Switch", ISABELLE_RX_INPUT_CFG2_REG, 4, 1, 0), +}; + +static const struct snd_kcontrol_new ep_path_enable_control = + SOC_DAPM_SINGLE("Switch", ISABELLE_EARDRV_CFG2_REG, 0, 1, 0); + +/* TLV Declarations */ +static const DECLARE_TLV_DB_SCALE(mic_amp_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(afm_amp_tlv, -3300, 300, 0); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -1200, 200, 0); +static const DECLARE_TLV_DB_SCALE(hf_tlv, -5000, 200, 0); + +/* from -63 to 0 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(dpga_tlv, -6300, 100, 1); + +/* from -63 to 9 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(rx_tlv, -6300, 100, 1); + +static const DECLARE_TLV_DB_SCALE(st_tlv, -2700, 300, 1); +static const DECLARE_TLV_DB_SCALE(tx_tlv, -600, 100, 0); + +static const struct snd_kcontrol_new isabelle_snd_controls[] = { + SOC_DOUBLE_TLV("Headset Playback Volume", ISABELLE_HSDRV_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_DOUBLE_R_TLV("Handsfree Playback Volume", + ISABELLE_HFLPGA_CFG_REG, ISABELLE_HFRPGA_CFG_REG, + 0, 0x1F, 0, hf_tlv), + SOC_DOUBLE_TLV("Aux Playback Volume", ISABELLE_LINEAMP_GAIN_REG, + 4, 0, 0xF, 0, dac_tlv), + SOC_SINGLE_TLV("Earpiece Playback Volume", ISABELLE_EARDRV_CFG1_REG, + 0, 0xF, 0, dac_tlv), + + SOC_DOUBLE_TLV("Aux FM Volume", ISABELLE_APGA_GAIN_REG, 4, 0, 0xF, 0, + afm_amp_tlv), + SOC_SINGLE_TLV("Mic1 Capture Volume", ISABELLE_MIC1_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + SOC_SINGLE_TLV("Mic2 Capture Volume", ISABELLE_MIC2_GAIN_REG, 3, 0x1F, + 0, mic_amp_tlv), + + SOC_DOUBLE_R_TLV("DPGA1 Volume", ISABELLE_DPGA1L_GAIN_REG, + ISABELLE_DPGA1R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA2 Volume", ISABELLE_DPGA2L_GAIN_REG, + ISABELLE_DPGA2R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + SOC_DOUBLE_R_TLV("DPGA3 Volume", ISABELLE_DPGA3L_GAIN_REG, + ISABELLE_DPGA3R_GAIN_REG, 0, 0x3F, 0, dpga_tlv), + + SOC_SINGLE_TLV("Sidetone Audio TX1 Volume", + ISABELLE_ATX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Audio TX2 Volume", + ISABELLE_ATX_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX1 Volume", + ISABELLE_VTX_STPGA1_CFG_REG, 0, 0xF, 0, st_tlv), + SOC_SINGLE_TLV("Sidetone Voice TX2 Volume", + ISABELLE_VTX2_STPGA2_CFG_REG, 0, 0xF, 0, st_tlv), + + SOC_SINGLE_TLV("Audio TX1 Volume", ISABELLE_ATX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Audio TX2 Volume", ISABELLE_ATX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX1 Volume", ISABELLE_VTX1_DPGA_REG, 4, 0xF, 0, + tx_tlv), + SOC_SINGLE_TLV("Voice TX2 Volume", ISABELLE_VTX2_DPGA_REG, 4, 0xF, 0, + tx_tlv), + + SOC_SINGLE_TLV("RX1 DPGA Volume", ISABELLE_RX1_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX2 DPGA Volume", ISABELLE_RX2_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX3 DPGA Volume", ISABELLE_RX3_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX4 DPGA Volume", ISABELLE_RX4_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX5 DPGA Volume", ISABELLE_RX5_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + SOC_SINGLE_TLV("RX6 DPGA Volume", ISABELLE_RX6_DPGA_REG, 0, 0x3F, 0, + rx_tlv), + + SOC_SINGLE("Headset Noise Gate", ISABELLE_HS_NG_CFG1_REG, 7, 1, 0), + SOC_SINGLE("Handsfree Noise Gate", ISABELLE_HF_NG_CFG1_REG, 7, 1, 0), + + SOC_SINGLE("ATX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 6, 1, 0), + SOC_SINGLE("ARX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 5, 1, 0), + SOC_SINGLE("ARX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 4, 1, 0), + SOC_SINGLE("ARX3 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("ARX4 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + SOC_SINGLE("ARX5 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 1, 1, 0), + SOC_SINGLE("ARX6 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 0, 1, 0), + SOC_SINGLE("VRX1 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 3, 1, 0), + SOC_SINGLE("VRX2 Filter Bypass Switch", ISABELLE_AUDIO_HPF_CFG_REG, + 2, 1, 0), + + SOC_SINGLE("ATX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 7, 1, 0), + SOC_SINGLE("ATX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 6, 1, 0), + SOC_SINGLE("VTX1 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 5, 1, 0), + SOC_SINGLE("VTX2 Filter Enable Switch", ISABELLE_ALU_TX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX1 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 5, 1, 0), + SOC_SINGLE("RX2 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 4, 1, 0), + SOC_SINGLE("RX3 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 3, 1, 0), + SOC_SINGLE("RX4 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 2, 1, 0), + SOC_SINGLE("RX5 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 1, 1, 0), + SOC_SINGLE("RX6 Filter Enable Switch", ISABELLE_ALU_RX_EN_REG, + 0, 1, 0), + + SOC_SINGLE("ULATX12 Capture Switch", ISABELLE_ULATX12_INTF_CFG_REG, + 7, 1, 0), + + SOC_SINGLE("DL12 Playback Switch", ISABELLE_DL12_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL34 Playback Switch", ISABELLE_DL34_INTF_CFG_REG, + 7, 1, 0), + SOC_SINGLE("DL56 Playback Switch", ISABELLE_DL56_INTF_CFG_REG, + 7, 1, 0), + + /* DMIC Switch */ + SOC_SINGLE("DMIC Switch", ISABELLE_DMIC_CFG_REG, 0, 1, 0), +}; + +static const struct snd_soc_dapm_widget isabelle_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("MAINMIC"), + SND_SOC_DAPM_INPUT("HSMIC"), + SND_SOC_DAPM_INPUT("SUBMIC"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + SND_SOC_DAPM_INPUT("DMICDAT"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HSOL"), + SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("HFL"), + SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), + SND_SOC_DAPM_OUTPUT("LINEOUT1"), + SND_SOC_DAPM_OUTPUT("LINEOUT2"), + + SND_SOC_DAPM_PGA("DL1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL3", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL4", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL5", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DL6", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog input muxes for the capture amplifiers */ + SND_SOC_DAPM_MUX("Analog Left Capture Route", + SND_SOC_NOPM, 0, 0, &amic1_control), + SND_SOC_DAPM_MUX("Analog Right Capture Route", + SND_SOC_NOPM, 0, 0, &amic2_control), + + SND_SOC_DAPM_MUX("Sidetone Audio Playback", SND_SOC_NOPM, 0, 0, + &st_audio_control), + SND_SOC_DAPM_MUX("Sidetone Voice Playback", SND_SOC_NOPM, 0, 0, + &st_voice_control), + + /* AIF */ + SND_SOC_DAPM_AIF_IN("INTF1_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 7, 0), + SND_SOC_DAPM_AIF_IN("INTF2_SDI", NULL, 0, ISABELLE_INTF_EN_REG, 6, 0), + + SND_SOC_DAPM_AIF_OUT("INTF1_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 5, 0), + SND_SOC_DAPM_AIF_OUT("INTF2_SDO", NULL, 0, ISABELLE_INTF_EN_REG, 4, 0), + + SND_SOC_DAPM_OUT_DRV("ULATX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULATX2", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("ULVTX2", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Capture PGAs */ + SND_SOC_DAPM_PGA("MicAmp1", ISABELLE_AMIC_CFG_REG, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("MicAmp2", ISABELLE_AMIC_CFG_REG, 4, 0, NULL, 0), + + /* Auxiliary FM PGAs */ + SND_SOC_DAPM_PGA("APGA1", ISABELLE_APGA_CFG_REG, 7, 0, NULL, 0), + SND_SOC_DAPM_PGA("APGA2", ISABELLE_APGA_CFG_REG, 6, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1", "Left Front Capture", + ISABELLE_AMIC_CFG_REG, 7, 0), + SND_SOC_DAPM_ADC("ADC2", "Right Front Capture", + ISABELLE_AMIC_CFG_REG, 6, 0), + + /* Microphone Bias */ + SND_SOC_DAPM_SUPPLY("Headset Mic Bias", ISABELLE_ABIAS_CFG_REG, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Main Mic Bias", ISABELLE_ABIAS_CFG_REG, + 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic1 Bias", + ISABELLE_DBIAS_CFG_REG, 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Digital Mic2 Bias", + ISABELLE_DBIAS_CFG_REG, 2, 0, NULL, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Headset Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_left_mixer_controls, + ARRAY_SIZE(isabelle_hs_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Headset Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hs_right_mixer_controls, + ARRAY_SIZE(isabelle_hs_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Left Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_left_mixer_controls, + ARRAY_SIZE(isabelle_hf_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Handsfree Right Mixer", SND_SOC_NOPM, 0, 0, + isabelle_hf_right_mixer_controls, + ARRAY_SIZE(isabelle_hf_right_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_left_mixer_controls, + ARRAY_SIZE(isabelle_aux_left_mixer_controls)), + SND_SOC_DAPM_MIXER("LINEOUT2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_aux_right_mixer_controls, + ARRAY_SIZE(isabelle_aux_right_mixer_controls)), + SND_SOC_DAPM_MIXER("Earphone Mixer", SND_SOC_NOPM, 0, 0, + isabelle_ep_mixer_controls, + ARRAY_SIZE(isabelle_ep_mixer_controls)), + + SND_SOC_DAPM_MIXER("DPGA1L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA1R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga1_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga1_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA2R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga2_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga2_right_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3L Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_left_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_left_mixer_controls)), + SND_SOC_DAPM_MIXER("DPGA3R Mixer", SND_SOC_NOPM, 0, 0, + isabelle_dpga3_right_mixer_controls, + ARRAY_SIZE(isabelle_dpga3_right_mixer_controls)), + + SND_SOC_DAPM_MIXER("RX1 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx1_mixer_controls, + ARRAY_SIZE(isabelle_rx1_mixer_controls)), + SND_SOC_DAPM_MIXER("RX2 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx2_mixer_controls, + ARRAY_SIZE(isabelle_rx2_mixer_controls)), + SND_SOC_DAPM_MIXER("RX3 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx3_mixer_controls, + ARRAY_SIZE(isabelle_rx3_mixer_controls)), + SND_SOC_DAPM_MIXER("RX4 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx4_mixer_controls, + ARRAY_SIZE(isabelle_rx4_mixer_controls)), + SND_SOC_DAPM_MIXER("RX5 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx5_mixer_controls, + ARRAY_SIZE(isabelle_rx5_mixer_controls)), + SND_SOC_DAPM_MIXER("RX6 Mixer", SND_SOC_NOPM, 0, 0, + isabelle_rx6_mixer_controls, + ARRAY_SIZE(isabelle_rx6_mixer_controls)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", "Headset Playback", ISABELLE_DAC_CFG_REG, + 5, 0), + SND_SOC_DAPM_DAC("DAC1R", "Headset Playback", ISABELLE_DAC_CFG_REG, + 4, 0), + SND_SOC_DAPM_DAC("DAC2L", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 3, 0), + SND_SOC_DAPM_DAC("DAC2R", "Handsfree Playback", ISABELLE_DAC_CFG_REG, + 2, 0), + SND_SOC_DAPM_DAC("DAC3L", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 1, 0), + SND_SOC_DAPM_DAC("DAC3R", "Lineout Playback", ISABELLE_DAC_CFG_REG, + 0, 0), + + /* Analog Playback PGAs */ + SND_SOC_DAPM_PGA("Sidetone Audio PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("Sidetone Voice PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Left PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("HF Right PGA", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA1R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA2R", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("DPGA3R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Analog Playback Mux */ + SND_SOC_DAPM_MUX("RX1 Playback", ISABELLE_ALU_RX_EN_REG, 5, 0, + &rx1_mux_controls), + SND_SOC_DAPM_MUX("RX2 Playback", ISABELLE_ALU_RX_EN_REG, 4, 0, + &rx2_mux_controls), + + /* TX Select */ + SND_SOC_DAPM_MUX("ATX Select", ISABELLE_TX_INPUT_CFG_REG, + 7, 0, &atx_mux_controls), + SND_SOC_DAPM_MUX("VTX Select", ISABELLE_TX_INPUT_CFG_REG, + 6, 0, &vtx_mux_controls), + + SND_SOC_DAPM_SWITCH("Earphone Playback", SND_SOC_NOPM, 0, 0, + &ep_path_enable_control), + + /* Output Drivers */ + SND_SOC_DAPM_OUT_DRV("HS Left Driver", ISABELLE_HSDRV_CFG2_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HS Right Driver", ISABELLE_HSDRV_CFG2_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT1 Left Driver", ISABELLE_LINEAMP_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("LINEOUT2 Right Driver", ISABELLE_LINEAMP_CFG_REG, + 0, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Earphone Driver", ISABELLE_EARDRV_CFG2_REG, + 1, 0, NULL, 0), + + SND_SOC_DAPM_OUT_DRV("HF Left Driver", ISABELLE_HFDRV_CFG_REG, + 1, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HF Right Driver", ISABELLE_HFDRV_CFG_REG, + 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route isabelle_intercon[] = { + /* Interface mapping */ + { "DL1", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF1_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF1_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF1_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF1_SDI" }, + + { "DL1", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL2", "DL12 Playback Switch", "INTF2_SDI" }, + { "DL3", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL4", "DL34 Playback Switch", "INTF2_SDI" }, + { "DL5", "DL56 Playback Switch", "INTF2_SDI" }, + { "DL6", "DL56 Playback Switch", "INTF2_SDI" }, + + /* Input side mapping */ + { "Sidetone Audio PGA", NULL, "Sidetone Audio Playback" }, + { "Sidetone Voice PGA", NULL, "Sidetone Voice Playback" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Audio PGA" }, + + { "RX1 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX1 Mixer", "DL1 Playback Switch", "DL1" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Audio PGA" }, + + { "RX2 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX2 Mixer", "DL2 Playback Switch", "DL2" }, + + { "RX3 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX3 Mixer", "DL3 Playback Switch", "DL3" }, + + { "RX4 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX4 Mixer", "DL4 Playback Switch", "DL4" }, + + { "RX5 Mixer", "ST1 Playback Switch", "Sidetone Voice PGA" }, + { "RX5 Mixer", "DL5 Playback Switch", "DL5" }, + + { "RX6 Mixer", "ST2 Playback Switch", "Sidetone Voice PGA" }, + { "RX6 Mixer", "DL6 Playback Switch", "DL6" }, + + /* Capture path */ + { "Analog Left Capture Route", "Headset Mic", "HSMIC" }, + { "Analog Left Capture Route", "Main Mic", "MAINMIC" }, + { "Analog Left Capture Route", "Aux/FM Left", "LINEIN1" }, + + { "Analog Right Capture Route", "Sub Mic", "SUBMIC" }, + { "Analog Right Capture Route", "Aux/FM Right", "LINEIN2" }, + + { "MicAmp1", NULL, "Analog Left Capture Route" }, + { "MicAmp2", NULL, "Analog Right Capture Route" }, + + { "ADC1", NULL, "MicAmp1" }, + { "ADC2", NULL, "MicAmp2" }, + + { "ATX Select", "AMIC1", "ADC1" }, + { "ATX Select", "DMIC", "DMICDAT" }, + { "ATX Select", "AMIC2", "ADC2" }, + + { "VTX Select", "AMIC1", "ADC1" }, + { "VTX Select", "DMIC", "DMICDAT" }, + { "VTX Select", "AMIC2", "ADC2" }, + + { "ULATX1", "ATX1 Filter Enable Switch", "ATX Select" }, + { "ULATX1", "ATX1 Filter Bypass Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Enable Switch", "ATX Select" }, + { "ULATX2", "ATX2 Filter Bypass Switch", "ATX Select" }, + + { "ULVTX1", "VTX1 Filter Enable Switch", "VTX Select" }, + { "ULVTX1", "VTX1 Filter Bypass Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Enable Switch", "VTX Select" }, + { "ULVTX2", "VTX2 Filter Bypass Switch", "VTX Select" }, + + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF1_SDO", "ULATX12 Capture Switch", "ULATX2" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX1" }, + { "INTF2_SDO", "ULATX12 Capture Switch", "ULATX2" }, + + { "INTF1_SDO", NULL, "ULVTX1" }, + { "INTF1_SDO", NULL, "ULVTX2" }, + { "INTF2_SDO", NULL, "ULVTX1" }, + { "INTF2_SDO", NULL, "ULVTX2" }, + + /* AFM Path */ + { "APGA1", NULL, "LINEIN1" }, + { "APGA2", NULL, "LINEIN2" }, + + { "RX1 Playback", "VRX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "ARX1 Filter Bypass Switch", "RX1 Mixer" }, + { "RX1 Playback", "RX1 Filter Enable Switch", "RX1 Mixer" }, + + { "RX2 Playback", "VRX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "ARX2 Filter Bypass Switch", "RX2 Mixer" }, + { "RX2 Playback", "RX2 Filter Enable Switch", "RX2 Mixer" }, + + { "RX3 Playback", "ARX3 Filter Bypass Switch", "RX3 Mixer" }, + { "RX3 Playback", "RX3 Filter Enable Switch", "RX3 Mixer" }, + + { "RX4 Playback", "ARX4 Filter Bypass Switch", "RX4 Mixer" }, + { "RX4 Playback", "RX4 Filter Enable Switch", "RX4 Mixer" }, + + { "RX5 Playback", "ARX5 Filter Bypass Switch", "RX5 Mixer" }, + { "RX5 Playback", "RX5 Filter Enable Switch", "RX5 Mixer" }, + + { "RX6 Playback", "ARX6 Filter Bypass Switch", "RX6 Mixer" }, + { "RX6 Playback", "RX6 Filter Enable Switch", "RX6 Mixer" }, + + { "DPGA1L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA1L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA1L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA1R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA1R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA1R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA1L", NULL, "DPGA1L Mixer" }, + { "DPGA1R", NULL, "DPGA1R Mixer" }, + + { "DAC1L", NULL, "DPGA1L" }, + { "DAC1R", NULL, "DPGA1R" }, + + { "DPGA2L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA2L Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA2L Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + { "DPGA2L Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA2R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA2R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA2L", NULL, "DPGA2L Mixer" }, + { "DPGA2R", NULL, "DPGA2R Mixer" }, + + { "DAC2L", NULL, "DPGA2L" }, + { "DAC2R", NULL, "DPGA2R" }, + + { "DPGA3L Mixer", "RX1 Playback Switch", "RX1 Playback" }, + { "DPGA3L Mixer", "RX3 Playback Switch", "RX3 Playback" }, + { "DPGA3L Mixer", "RX5 Playback Switch", "RX5 Playback" }, + + { "DPGA3R Mixer", "RX2 Playback Switch", "RX2 Playback" }, + { "DPGA3R Mixer", "RX4 Playback Switch", "RX4 Playback" }, + { "DPGA3R Mixer", "RX6 Playback Switch", "RX6 Playback" }, + + { "DPGA3L", NULL, "DPGA3L Mixer" }, + { "DPGA3R", NULL, "DPGA3R Mixer" }, + + { "DAC3L", NULL, "DPGA3L" }, + { "DAC3R", NULL, "DPGA3R" }, + + { "Headset Left Mixer", "DAC1L Playback Switch", "DAC1L" }, + { "Headset Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Headset Right Mixer", "DAC1R Playback Switch", "DAC1R" }, + { "Headset Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HS Left Driver", NULL, "Headset Left Mixer" }, + { "HS Right Driver", NULL, "Headset Right Mixer" }, + + { "HSOL", NULL, "HS Left Driver" }, + { "HSOR", NULL, "HS Right Driver" }, + + /* Earphone playback path */ + { "Earphone Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Earphone Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Earphone Playback", "Switch", "Earphone Mixer" }, + { "Earphone Driver", NULL, "Earphone Playback" }, + { "EP", NULL, "Earphone Driver" }, + + { "Handsfree Left Mixer", "DAC2L Playback Switch", "DAC2L" }, + { "Handsfree Left Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "Handsfree Right Mixer", "DAC2R Playback Switch", "DAC2R" }, + { "Handsfree Right Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "HF Left PGA", NULL, "Handsfree Left Mixer" }, + { "HF Right PGA", NULL, "Handsfree Right Mixer" }, + + { "HF Left Driver", NULL, "HF Left PGA" }, + { "HF Right Driver", NULL, "HF Right PGA" }, + + { "HFL", NULL, "HF Left Driver" }, + { "HFR", NULL, "HF Right Driver" }, + + { "LINEOUT1 Mixer", "DAC3L Playback Switch", "DAC3L" }, + { "LINEOUT1 Mixer", "APGA1 Playback Switch", "APGA1" }, + + { "LINEOUT2 Mixer", "DAC3R Playback Switch", "DAC3R" }, + { "LINEOUT2 Mixer", "APGA2 Playback Switch", "APGA2" }, + + { "LINEOUT1 Driver", NULL, "LINEOUT1 Mixer" }, + { "LINEOUT2 Driver", NULL, "LINEOUT2 Mixer" }, + + { "LINEOUT1", NULL, "LINEOUT1 Driver" }, + { "LINEOUT2", NULL, "LINEOUT2 Driver" }, +}; + +static int isabelle_hs_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC1_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_hf_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC2_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_line_mute(struct snd_soc_dai *dai, int mute) +{ + snd_soc_update_bits(dai->codec, ISABELLE_DAC3_SOFTRAMP_REG, + BIT(4), (mute ? BIT(4) : 0)); + + return 0; +} + +static int isabelle_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, BIT(0)); + break; + + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ISABELLE_PWR_EN_REG, + ISABELLE_CHIP_EN, 0); + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int isabelle_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u16 aif = 0; + unsigned int fs_val = 0; + + switch (params_rate(params)) { + case 8000: + fs_val = ISABELLE_FS_RATE_8; + break; + case 11025: + fs_val = ISABELLE_FS_RATE_11; + break; + case 12000: + fs_val = ISABELLE_FS_RATE_12; + break; + case 16000: + fs_val = ISABELLE_FS_RATE_16; + break; + case 22050: + fs_val = ISABELLE_FS_RATE_22; + break; + case 24000: + fs_val = ISABELLE_FS_RATE_24; + break; + case 32000: + fs_val = ISABELLE_FS_RATE_32; + break; + case 44100: + fs_val = ISABELLE_FS_RATE_44; + break; + case 48000: + fs_val = ISABELLE_FS_RATE_48; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_FS_RATE_CFG_REG, + ISABELLE_FS_RATE_MASK, fs_val); + + /* bit size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= ISABELLE_AIF_LENGTH_20; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= ISABELLE_AIF_LENGTH_32; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + ISABELLE_AIF_LENGTH_MASK, aif); + + return 0; +} + +static int isabelle_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + unsigned int aif_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif_val &= ~ISABELLE_AIF_MS; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif_val |= ISABELLE_AIF_MS; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif_val |= ISABELLE_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif_val |= ISABELLE_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_PDM: + aif_val |= ISABELLE_PDM_MODE; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ISABELLE_INTF_CFG_REG, + (ISABELLE_AIF_MS | ISABELLE_AIF_FMT_MASK), aif_val); + + return 0; +} + +/* Rates supported by Isabelle driver */ +#define ISABELLE_RATES SNDRV_PCM_RATE_8000_48000 + +/* Formates supported by Isabelle driver. */ +#define ISABELLE_FORMATS (SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops isabelle_hs_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hs_mute, +}; + +static struct snd_soc_dai_ops isabelle_hf_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_hf_mute, +}; + +static struct snd_soc_dai_ops isabelle_line_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, + .digital_mute = isabelle_line_mute, +}; + +static struct snd_soc_dai_ops isabelle_ul_dai_ops = { + .hw_params = isabelle_hw_params, + .set_fmt = isabelle_set_dai_fmt, +}; + +/* ISABELLE dai structure */ +struct snd_soc_dai_driver isabelle_dai[] = { + { + .name = "isabelle-dl1", + .playback = { + .stream_name = "Headset Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hs_dai_ops, + }, + { + .name = "isabelle-dl2", + .playback = { + .stream_name = "Handsfree Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_hf_dai_ops, + }, + { + .name = "isabelle-lineout", + .playback = { + .stream_name = "Lineout Playback", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_line_dai_ops, + }, + { + .name = "isabelle-ul", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = ISABELLE_RATES, + .formats = ISABELLE_FORMATS, + }, + .ops = &isabelle_ul_dai_ops, + }, +}; + +static int isabelle_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + + codec->control_data = dev_get_regmap(codec->dev, NULL); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_isabelle = { + .probe = isabelle_probe, + .set_bias_level = isabelle_set_bias_level, + .controls = isabelle_snd_controls, + .num_controls = ARRAY_SIZE(isabelle_snd_controls), + .dapm_widgets = isabelle_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(isabelle_dapm_widgets), + .dapm_routes = isabelle_intercon, + .num_dapm_routes = ARRAY_SIZE(isabelle_intercon), + .idle_bias_off = true, +}; + +static const struct regmap_config isabelle_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = ISABELLE_MAX_REGISTER, + .reg_defaults = isabelle_reg_defs, + .num_reg_defaults = ARRAY_SIZE(isabelle_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *isabelle_regmap; + int ret = 0; + + i2c_set_clientdata(i2c, isabelle_regmap); + + isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); + if (IS_ERR(isabelle_regmap)) { + ret = PTR_ERR(isabelle_regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + ret = snd_soc_register_codec(&i2c->dev, + &soc_codec_dev_isabelle, isabelle_dai, + ARRAY_SIZE(isabelle_dai)); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); + regmap_exit(dev_get_regmap(&i2c->dev, NULL)); + return ret; + } + + return ret; +} + +static int __devexit isabelle_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + regmap_exit(dev_get_regmap(&client->dev, NULL)); + return 0; +} + +static const struct i2c_device_id isabelle_i2c_id[] = { + { "isabelle", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, isabelle_i2c_id); + +static struct i2c_driver isabelle_i2c_driver = { + .driver = { + .name = "isabelle", + .owner = THIS_MODULE, + }, + .probe = isabelle_i2c_probe, + .remove = __devexit_p(isabelle_i2c_remove), + .id_table = isabelle_i2c_id, +}; + +module_i2c_driver(isabelle_i2c_driver); + +MODULE_DESCRIPTION("ASoC ISABELLE driver"); +MODULE_AUTHOR("Vishwas A Deshpande "); +MODULE_AUTHOR("M R Swami Reddy "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/isabelle.h b/sound/soc/codecs/isabelle.h new file mode 100644 index 0000000..96d839a --- /dev/null +++ b/sound/soc/codecs/isabelle.h @@ -0,0 +1,143 @@ +/* + * isabelle.h - Low power high fidelity audio codec driver header file + * + * Copyright (c) 2012 Texas Instruments, Inc + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + */ + +#ifndef _ISABELLE_H +#define _ISABELLE_H + +#include + +/* ISABELLE REGISTERS */ + +#define ISABELLE_PWR_CFG_REG 0x01 +#define ISABELLE_PWR_EN_REG 0x02 +#define ISABELLE_PS_EN1_REG 0x03 +#define ISABELLE_INT1_STATUS_REG 0x04 +#define ISABELLE_INT1_MASK_REG 0x05 +#define ISABELLE_INT2_STATUS_REG 0x06 +#define ISABELLE_INT2_MASK_REG 0x07 +#define ISABELLE_HKCTL1_REG 0x08 +#define ISABELLE_HKCTL2_REG 0x09 +#define ISABELLE_HKCTL3_REG 0x0A +#define ISABELLE_ACCDET_STATUS_REG 0x0B +#define ISABELLE_BUTTON_ID_REG 0x0C +#define ISABELLE_PLL_CFG_REG 0x10 +#define ISABELLE_PLL_EN_REG 0x11 +#define ISABELLE_FS_RATE_CFG_REG 0x12 +#define ISABELLE_INTF_CFG_REG 0x13 +#define ISABELLE_INTF_EN_REG 0x14 +#define ISABELLE_ULATX12_INTF_CFG_REG 0x15 +#define ISABELLE_DL12_INTF_CFG_REG 0x16 +#define ISABELLE_DL34_INTF_CFG_REG 0x17 +#define ISABELLE_DL56_INTF_CFG_REG 0x18 +#define ISABELLE_ATX_STPGA1_CFG_REG 0x19 +#define ISABELLE_ATX_STPGA2_CFG_REG 0x1A +#define ISABELLE_VTX_STPGA1_CFG_REG 0x1B +#define ISABELLE_VTX2_STPGA2_CFG_REG 0x1C +#define ISABELLE_ATX1_DPGA_REG 0x1D +#define ISABELLE_ATX2_DPGA_REG 0x1E +#define ISABELLE_VTX1_DPGA_REG 0x1F +#define ISABELLE_VTX2_DPGA_REG 0x20 +#define ISABELLE_TX_INPUT_CFG_REG 0x21 +#define ISABELLE_RX_INPUT_CFG_REG 0x22 +#define ISABELLE_RX_INPUT_CFG2_REG 0x23 +#define ISABELLE_VOICE_HPF_CFG_REG 0x24 +#define ISABELLE_AUDIO_HPF_CFG_REG 0x25 +#define ISABELLE_RX1_DPGA_REG 0x26 +#define ISABELLE_RX2_DPGA_REG 0x27 +#define ISABELLE_RX3_DPGA_REG 0x28 +#define ISABELLE_RX4_DPGA_REG 0x29 +#define ISABELLE_RX5_DPGA_REG 0x2A +#define ISABELLE_RX6_DPGA_REG 0x2B +#define ISABELLE_ALU_TX_EN_REG 0x2C +#define ISABELLE_ALU_RX_EN_REG 0x2D +#define ISABELLE_IIR_RESYNC_REG 0x2E +#define ISABELLE_ABIAS_CFG_REG 0x30 +#define ISABELLE_DBIAS_CFG_REG 0x31 +#define ISABELLE_MIC1_GAIN_REG 0x32 +#define ISABELLE_MIC2_GAIN_REG 0x33 +#define ISABELLE_AMIC_CFG_REG 0x34 +#define ISABELLE_DMIC_CFG_REG 0x35 +#define ISABELLE_APGA_GAIN_REG 0x36 +#define ISABELLE_APGA_CFG_REG 0x37 +#define ISABELLE_TX_GAIN_DLY_REG 0x38 +#define ISABELLE_RX_GAIN_DLY_REG 0x39 +#define ISABELLE_RX_PWR_CTRL_REG 0x3A +#define ISABELLE_DPGA1LR_IN_SEL_REG 0x3B +#define ISABELLE_DPGA1L_GAIN_REG 0x3C +#define ISABELLE_DPGA1R_GAIN_REG 0x3D +#define ISABELLE_DPGA2L_IN_SEL_REG 0x3E +#define ISABELLE_DPGA2R_IN_SEL_REG 0x3F +#define ISABELLE_DPGA2L_GAIN_REG 0x40 +#define ISABELLE_DPGA2R_GAIN_REG 0x41 +#define ISABELLE_DPGA3LR_IN_SEL_REG 0x42 +#define ISABELLE_DPGA3L_GAIN_REG 0x43 +#define ISABELLE_DPGA3R_GAIN_REG 0x44 +#define ISABELLE_DAC1_SOFTRAMP_REG 0x45 +#define ISABELLE_DAC2_SOFTRAMP_REG 0x46 +#define ISABELLE_DAC3_SOFTRAMP_REG 0x47 +#define ISABELLE_DAC_CFG_REG 0x48 +#define ISABELLE_EARDRV_CFG1_REG 0x49 +#define ISABELLE_EARDRV_CFG2_REG 0x4A +#define ISABELLE_HSDRV_GAIN_REG 0x4B +#define ISABELLE_HSDRV_CFG1_REG 0x4C +#define ISABELLE_HSDRV_CFG2_REG 0x4D +#define ISABELLE_HS_NG_CFG1_REG 0x4E +#define ISABELLE_HS_NG_CFG2_REG 0x4F +#define ISABELLE_LINEAMP_GAIN_REG 0x50 +#define ISABELLE_LINEAMP_CFG_REG 0x51 +#define ISABELLE_HFL_VOL_CTRL_REG 0x52 +#define ISABELLE_HFL_SFTVOL_CTRL_REG 0x53 +#define ISABELLE_HFL_LIM_CTRL_1_REG 0x54 +#define ISABELLE_HFL_LIM_CTRL_2_REG 0x55 +#define ISABELLE_HFR_VOL_CTRL_REG 0x56 +#define ISABELLE_HFR_SFTVOL_CTRL_REG 0x57 +#define ISABELLE_HFR_LIM_CTRL_1_REG 0x58 +#define ISABELLE_HFR_LIM_CTRL_2_REG 0x59 +#define ISABELLE_HF_MODE_REG 0x5A +#define ISABELLE_HFLPGA_CFG_REG 0x5B +#define ISABELLE_HFRPGA_CFG_REG 0x5C +#define ISABELLE_HFDRV_CFG_REG 0x5D +#define ISABELLE_PDMOUT_CFG1_REG 0x5E +#define ISABELLE_PDMOUT_CFG2_REG 0x5F +#define ISABELLE_PDMOUT_L_WM_REG 0x60 +#define ISABELLE_PDMOUT_R_WM_REG 0x61 +#define ISABELLE_HF_NG_CFG1_REG 0x62 +#define ISABELLE_HF_NG_CFG2_REG 0x63 + +/* ISABELLE_PWR_EN_REG (0x02h) */ +#define ISABELLE_CHIP_EN BIT(0) + +/* ISABELLE DAI FORMATS */ +#define ISABELLE_AIF_FMT_MASK 0x70 +#define ISABELLE_I2S_MODE 0x0 +#define ISABELLE_LEFT_J_MODE 0x1 +#define ISABELLE_PDM_MODE 0x2 + +#define ISABELLE_AIF_LENGTH_MASK 0x30 +#define ISABELLE_AIF_LENGTH_20 0x00 +#define ISABELLE_AIF_LENGTH_32 0x10 + +#define ISABELLE_AIF_MS 0x80 + +#define ISABELLE_FS_RATE_MASK 0xF +#define ISABELLE_FS_RATE_8 0x0 +#define ISABELLE_FS_RATE_11 0x1 +#define ISABELLE_FS_RATE_12 0x2 +#define ISABELLE_FS_RATE_16 0x4 +#define ISABELLE_FS_RATE_22 0x5 +#define ISABELLE_FS_RATE_24 0x6 +#define ISABELLE_FS_RATE_32 0x8 +#define ISABELLE_FS_RATE_44 0x9 +#define ISABELLE_FS_RATE_48 0xA + +#define ISABELLE_MAX_REGISTER 0xFF + +#endif -- cgit v0.10.2 From 5eba8ec37fe8cfed4cacff56f9025b756cc43faa Mon Sep 17 00:00:00 2001 From: MR Swami Reddy Date: Mon, 4 Jun 2012 17:44:54 +0530 Subject: ASoC: isabelle: Remove regmap_exit() With devm_ APIs regmap_exit() not needed, so remove regmap_exit(). Signed-off-by: Vishwas A Deshpande Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index b6921a8..bcc77ef 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1141,7 +1141,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, ARRAY_SIZE(isabelle_dai)); if (ret < 0) { dev_err(&i2c->dev, "Failed to register codec: %d\n", ret); - regmap_exit(dev_get_regmap(&i2c->dev, NULL)); return ret; } @@ -1151,7 +1150,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, static int __devexit isabelle_i2c_remove(struct i2c_client *client) { snd_soc_unregister_codec(&client->dev); - regmap_exit(dev_get_regmap(&client->dev, NULL)); return 0; } -- cgit v0.10.2 From 165961efc03159631eadc086877704c7778ac356 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Jun 2012 10:44:23 +0100 Subject: ASoC: dapm: The clock API is even less consistent than thought Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 39e8c2f..7365fed 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2904,7 +2904,7 @@ snd_soc_dapm_new_control(struct snd_soc_dapm_context *dapm, } break; case snd_soc_dapm_clock_supply: -#ifdef CONFIG_HAVE_CLK +#ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); if (IS_ERR(w->clk)) { ret = PTR_ERR(w->clk); -- cgit v0.10.2 From 571f6a7f07e9dda6c9795398747278e52368c88a Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Mon, 4 Jun 2012 13:19:41 -0500 Subject: ASoC: cs42l73: Convert to devm_regmap_init_i2c() Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index e0d45fd..2c08c4c 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1362,11 +1362,11 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, i2c_set_clientdata(i2c_client, cs42l73); - cs42l73->regmap = regmap_init_i2c(i2c_client, &cs42l73_regmap); + cs42l73->regmap = devm_regmap_init_i2c(i2c_client, &cs42l73_regmap); if (IS_ERR(cs42l73->regmap)) { ret = PTR_ERR(cs42l73->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } /* initialize codec */ ret = regmap_read(cs42l73->regmap, CS42L73_DEVID_AB, ®); @@ -1384,13 +1384,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L73 Device ID (%X). Expected %X\n", devid, CS42L73_DEVID); - goto err_regmap; + return ret; } ret = regmap_read(cs42l73->regmap, CS42L73_REVID, ®); if (ret < 0) { dev_err(&i2c_client->dev, "Get Revision ID failed\n"); - goto err_regmap; + return ret;; } dev_info(&i2c_client->dev, @@ -1402,23 +1402,13 @@ static __devinit int cs42l73_i2c_probe(struct i2c_client *i2c_client, &soc_codec_dev_cs42l73, cs42l73_dai, ARRAY_SIZE(cs42l73_dai)); if (ret < 0) - goto err_regmap; + return ret; return 0; - -err_regmap: - regmap_exit(cs42l73->regmap); - -err: - return ret; } static __devexit int cs42l73_i2c_remove(struct i2c_client *client) { - struct cs42l73_private *cs42l73 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(cs42l73->regmap); - return 0; } -- cgit v0.10.2 From 134b2f576b9144223dd5b59a496218e3aadaf56b Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Mon, 4 Jun 2012 13:19:42 -0500 Subject: ASoC: cs42l52: Convert to devm_regmap_init_i2c() Signed-off-by: Brian Austin Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index ec03abc..628daf6 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1216,11 +1216,11 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; cs42l52->dev = &i2c_client->dev; - cs42l52->regmap = regmap_init_i2c(i2c_client, &cs42l52_regmap); + cs42l52->regmap = devm_regmap_init_i2c(i2c_client, &cs42l52_regmap); if (IS_ERR(cs42l52->regmap)) { ret = PTR_ERR(cs42l52->regmap); dev_err(&i2c_client->dev, "regmap_init() failed: %d\n", ret); - goto err; + return ret; } i2c_set_clientdata(i2c_client, cs42l52); @@ -1242,7 +1242,7 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, dev_err(&i2c_client->dev, "CS42L52 Device ID (%X). Expected %X\n", devid, CS42L52_CHIP_ID); - goto err_regmap; + return ret; } regcache_cache_only(cs42l52->regmap, true); @@ -1250,23 +1250,13 @@ static int cs42l52_i2c_probe(struct i2c_client *i2c_client, ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_dev_cs42l52, &cs42l52_dai, 1); if (ret < 0) - goto err_regmap; + return ret; return 0; - -err_regmap: - regmap_exit(cs42l52->regmap); - -err: - return ret; } static int cs42l52_i2c_remove(struct i2c_client *client) { - struct cs42l52_private *cs42l52 = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(cs42l52->regmap); - return 0; } -- cgit v0.10.2 From 61dc479e99d4d74c6113656dc50babed90a384c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Jun 2012 11:08:45 +0100 Subject: Revert "ASoC: fsl_ssi: convert to use devm_clk_get" This reverts commit 014e5b56702575c5cd8ffc4b1a7924cfdfe0f1ea since PowerPC doesn't use clkdev and hasn't implemented devm_clk_get() itself. Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b10a427..4ed2afd 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -725,7 +725,7 @@ static int __devinit fsl_ssi_probe(struct platform_device *pdev) u32 dma_events[2]; ssi_private->ssi_on_imx = true; - ssi_private->clk = devm_clk_get(&pdev->dev, NULL); + ssi_private->clk = clk_get(&pdev->dev, NULL); if (IS_ERR(ssi_private->clk)) { ret = PTR_ERR(ssi_private->clk); dev_err(&pdev->dev, "could not get clock: %d\n", ret); @@ -842,8 +842,10 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); + } error_irq: free_irq(ssi_private->irq, ssi_private); @@ -869,6 +871,7 @@ static int fsl_ssi_remove(struct platform_device *pdev) if (ssi_private->ssi_on_imx) { platform_device_unregister(ssi_private->imx_pcm_pdev); clk_disable_unprepare(ssi_private->clk); + clk_put(ssi_private->clk); } snd_soc_unregister_dai(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); -- cgit v0.10.2 From cd86e3ce304189fbdb144622245d0da9189551a1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Jun 2012 18:20:21 +0100 Subject: ASoC: lm59453: Unconstify dai_driver The core fills in some blanks which makes it annoying to do the right thing and constify the calls in the core. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index c1bc945..99b0a9d 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1357,7 +1357,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- cgit v0.10.2 From a265367ccbe72010757a56e5776fcf9a49370181 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 31 May 2012 14:47:46 +0100 Subject: ASoC: max98095: Staticise non-exported functions and export jack detect Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 35179e2..7cd508e 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2216,7 +2216,7 @@ static irqreturn_t max98095_report_jack(int irq, void *data) return IRQ_HANDLED; } -int max98095_jack_detect_enable(struct snd_soc_codec *codec) +static int max98095_jack_detect_enable(struct snd_soc_codec *codec) { struct max98095_priv *max98095 = snd_soc_codec_get_drvdata(codec); int ret = 0; @@ -2245,7 +2245,7 @@ int max98095_jack_detect_enable(struct snd_soc_codec *codec) return ret; } -int max98095_jack_detect_disable(struct snd_soc_codec *codec) +static int max98095_jack_detect_disable(struct snd_soc_codec *codec) { int ret = 0; @@ -2286,6 +2286,7 @@ int max98095_jack_detect(struct snd_soc_codec *codec, max98095_report_jack(client->irq, codec); return 0; } +EXPORT_SYMBOL_GPL(max98095_jack_detect); #ifdef CONFIG_PM static int max98095_suspend(struct snd_soc_codec *codec) -- cgit v0.10.2 From 40820105d4caef1489edd56e9dc2b85871b65308 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Thu, 7 Jun 2012 15:38:37 +0300 Subject: ASoC: isabelle: using an uninitialized variable We should set "isabelle_regmap" before using it. GCC complains. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index bcc77ef..0d62f3b 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1126,8 +1126,6 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, struct regmap *isabelle_regmap; int ret = 0; - i2c_set_clientdata(i2c, isabelle_regmap); - isabelle_regmap = devm_regmap_init_i2c(i2c, &isabelle_regmap_config); if (IS_ERR(isabelle_regmap)) { ret = PTR_ERR(isabelle_regmap); @@ -1135,6 +1133,7 @@ static int __devinit isabelle_i2c_probe(struct i2c_client *i2c, ret); return ret; } + i2c_set_clientdata(i2c, isabelle_regmap); ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_isabelle, isabelle_dai, -- cgit v0.10.2 From d392dead724935ad45c42e1a802d0f1de478c0d2 Mon Sep 17 00:00:00 2001 From: M R Swami Reddy Date: Thu, 7 Jun 2012 18:37:54 +0530 Subject: ASoC: MAINTAINERS: Add maintainer for TI Isabelle Audio driver Vishwas and I support the TI Isabelle audio driver. Signed-off-by: Vishwas A Deshpande Signed-off-by: M R Swami Reddy Signed-off-by: Mark Brown diff --git a/MAINTAINERS b/MAINTAINERS index 55f0fda..d052664 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6723,9 +6723,11 @@ F: include/linux/tifm.h TI LM49xxx FAMILY ASoC CODEC DRIVERS M: M R Swami Reddy +M: Vishwas A Deshpande L: alsa-devel@alsa-project.org (moderated for non-subscribers) S: Maintained F: sound/soc/codecs/lm49453* +F: sound/soc/codecs/isabelle* TI TWL4030 SERIES SOC CODEC DRIVER M: Peter Ujfalusi -- cgit v0.10.2 From 9515c1010c98347ec92d923bd3e23793fa6dc6fe Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 6 Jun 2012 17:15:07 -0600 Subject: ASoC: tegra: add .stream_name to CPU DAIs This is certainly required if the I2S and SPDIF controllers are converted to be CODECs, and is probably good practice irrespective. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 0c7af63..9d5d470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -261,12 +261,14 @@ static const struct snd_soc_dai_ops tegra20_i2s_dai_ops = { static const struct snd_soc_dai_driver tegra20_i2s_dai_template = { .probe = tegra20_i2s_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index f9b5741..ffbd99c 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -181,6 +181,7 @@ static struct snd_soc_dai_driver tegra20_spdif_dai = { .name = DRV_NAME, .probe = tegra20_spdif_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 8596032..9c5c0e6 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -320,12 +320,14 @@ static struct snd_soc_dai_ops tegra30_i2s_dai_ops = { static const struct snd_soc_dai_driver tegra30_i2s_dai_template = { .probe = tegra30_i2s_probe, .playback = { + .stream_name = "Playback", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { + .stream_name = "Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_96000, -- cgit v0.10.2 From 408dafc4235e393036708126057e4d643f579486 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 6 Jun 2012 17:15:48 -0600 Subject: ASoC: tegra: statically define DAI link format Define the DAI format statically in the dai_link, rather than executing code to set it each time the hw params are set. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 1fd71e5..087d3d8 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -58,7 +58,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); @@ -86,24 +85,6 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -240,6 +221,9 @@ static struct snd_soc_dai_link tegra_wm8903_dai = { .codec_dai_name = "wm8903-hifi", .init = tegra_wm8903_init, .ops = &tegra_wm8903_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_tegra_wm8903 = { diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 5815430..62bb805 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -52,7 +52,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_codec *codec = rtd->codec; struct snd_soc_card *card = codec->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); @@ -68,24 +67,6 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, return err; } - err = snd_soc_dai_set_fmt(codec_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "codec_dai fmt not set\n"); - return err; - } - - err = snd_soc_dai_set_fmt(cpu_dai, - SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS); - if (err < 0) { - dev_err(card->dev, "cpu_dai fmt not set\n"); - return err; - } - err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk, SND_SOC_CLOCK_IN); if (err < 0) { @@ -121,6 +102,9 @@ static struct snd_soc_dai_link trimslice_tlv320aic23_dai = { .cpu_dai_name = "tegra20-i2s.0", .codec_dai_name = "tlv320aic23-hifi", .ops = &trimslice_asoc_ops, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, }; static struct snd_soc_card snd_soc_trimslice = { -- cgit v0.10.2 From 40db77a0c4223d0b87c4b61ae38760d47593b7a5 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 6 Jun 2012 17:15:49 -0600 Subject: ASoC: tegra: remove usage of rtd->codec rtd->codec_dai->codec can be used instead. This is a slight step along the way to not needing the rtd->codec field any more. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra_alc5632.c b/sound/soc/tegra/tegra_alc5632.c index 417b09b..d684df2 100644 --- a/sound/soc/tegra/tegra_alc5632.c +++ b/sound/soc/tegra/tegra_alc5632.c @@ -43,7 +43,7 @@ static int tegra_alc5632_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_alc5632 *alc5632 = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -105,7 +105,8 @@ static const struct snd_kcontrol_new tegra_alc5632_controls[] = { static int tegra_alc5632_asoc_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct tegra_alc5632 *machine = snd_soc_card_get_drvdata(codec->card); diff --git a/sound/soc/tegra/tegra_wm8753.c b/sound/soc/tegra/tegra_wm8753.c index 02bd5a8..ea9166d 100644 --- a/sound/soc/tegra/tegra_wm8753.c +++ b/sound/soc/tegra/tegra_wm8753.c @@ -57,7 +57,7 @@ static int tegra_wm8753_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8753 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; diff --git a/sound/soc/tegra/tegra_wm8903.c b/sound/soc/tegra/tegra_wm8903.c index 087d3d8..08b5fef 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -58,7 +58,7 @@ static int tegra_wm8903_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); int srate, mclk; @@ -181,7 +181,8 @@ static const struct snd_kcontrol_new tegra_wm8903_controls[] = { static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; struct snd_soc_card *card = codec->card; struct tegra_wm8903 *machine = snd_soc_card_get_drvdata(card); diff --git a/sound/soc/tegra/trimslice.c b/sound/soc/tegra/trimslice.c index 62bb805..e69a4f7 100644 --- a/sound/soc/tegra/trimslice.c +++ b/sound/soc/tegra/trimslice.c @@ -52,7 +52,7 @@ static int trimslice_asoc_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_card *card = codec->card; struct tegra_trimslice *trimslice = snd_soc_card_get_drvdata(card); int srate, mclk; -- cgit v0.10.2 From c92a40e3a163b6708e0dd82ba4612f79df846912 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 6 Jun 2012 17:15:05 -0600 Subject: ASoC: tegra: use DAI's not card's dev for dev_err This is the actual device of the I2S or SPDIF controller reporting the problem. If a future change converts these controllers to be CODECs, then there may be no pcm associated with the substream, so this change avoids a crash. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 9d5d470..647daf6 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -138,7 +138,7 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); u32 reg; int ret, sample_size, srate, i2sclock, bitcnt; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index ffbd99c..f774a2d 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -77,7 +77,7 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); int ret, spdifclock; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 9c5c0e6..2327f62 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -181,7 +181,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct device *dev = substream->pcm->card->dev; + struct device *dev = dai->dev; struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); u32 val; int ret, sample_size, srate, i2sclock, bitcnt; -- cgit v0.10.2 From 0f163546a772d62250f59bad6a9338a0e3a2605c Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Wed, 6 Jun 2012 17:15:06 -0600 Subject: ASoC: tegra: use regmap more directly Stop open-coding the caching of the ctrl registers; instead, use regmap_update_bits() to update parts of the register from different places. The removal of the open-coded cache will allow controls to be created which touch registers, which will be necessary if any of these modules are converted to CODECs. Get rid of tegra*_read/write; just call regmap_read/write directly. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 647daf6..c5fc6b1 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -46,18 +46,6 @@ #define DRV_NAME "tegra20-i2s" -static inline void tegra20_i2s_write(struct tegra20_i2s *i2s, u32 reg, u32 val) -{ - regmap_write(i2s->regmap, reg, val); -} - -static inline u32 tegra20_i2s_read(struct tegra20_i2s *i2s, u32 reg) -{ - u32 val; - regmap_read(i2s->regmap, reg, &val); - return val; -} - static int tegra20_i2s_runtime_suspend(struct device *dev) { struct tegra20_i2s *i2s = dev_get_drvdata(dev); @@ -85,6 +73,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -93,10 +82,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_MASTER_ENABLE; + val = TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -104,33 +93,35 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | - TEGRA20_I2S_CTRL_LRCK_MASK); + mask |= TEGRA20_I2S_CTRL_BIT_FORMAT_MASK | + TEGRA20_I2S_CTRL_LRCK_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_R_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_DSP; + val |= TEGRA20_I2S_CTRL_LRCK_R_LOW; break; case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_I2S; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_RJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA20_I2S_CTRL_BIT_FORMAT_LJM; + val |= TEGRA20_I2S_CTRL_LRCK_L_LOW; break; default: return -EINVAL; } + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + return 0; } @@ -140,27 +131,32 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 reg; + unsigned int mask, val; int ret, sample_size, srate, i2sclock, bitcnt; - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_BIT_SIZE_MASK; + mask = TEGRA20_I2S_CTRL_BIT_SIZE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_16; + val = TEGRA20_I2S_CTRL_BIT_SIZE_16; sample_size = 16; break; case SNDRV_PCM_FORMAT_S24_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_24; + val = TEGRA20_I2S_CTRL_BIT_SIZE_24; sample_size = 24; break; case SNDRV_PCM_FORMAT_S32_LE: - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_BIT_SIZE_32; + val = TEGRA20_I2S_CTRL_BIT_SIZE_32; sample_size = 32; break; default: return -EINVAL; } + mask |= TEGRA20_I2S_CTRL_FIFO_FORMAT_MASK; + val |= TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; + + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, mask, val); + srate = params_rate(params); /* Final "* 2" required by Tegra hardware */ @@ -175,42 +171,44 @@ static int tegra20_i2s_hw_params(struct snd_pcm_substream *substream, bitcnt = (i2sclock / (2 * srate)) - 1; if (bitcnt < 0 || bitcnt > TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_MASK_US) return -EINVAL; - reg = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; + val = bitcnt << TEGRA20_I2S_TIMING_CHANNEL_BIT_COUNT_SHIFT; if (i2sclock % (2 * srate)) - reg |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; + val |= TEGRA20_I2S_TIMING_NON_SYM_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_TIMING, reg); + regmap_write(i2s->regmap, TEGRA20_I2S_TIMING, val); - tegra20_i2s_write(i2s, TEGRA20_I2S_FIFO_SCR, - TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | - TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); + regmap_write(i2s->regmap, TEGRA20_I2S_FIFO_SCR, + TEGRA20_I2S_FIFO_SCR_FIFO2_ATN_LVL_FOUR_SLOTS | + TEGRA20_I2S_FIFO_SCR_FIFO1_ATN_LVL_FOUR_SLOTS); return 0; } static void tegra20_i2s_start_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO1_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, + TEGRA20_I2S_CTRL_FIFO1_ENABLE); } static void tegra20_i2s_stop_playback(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO1_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO1_ENABLE, 0); } static void tegra20_i2s_start_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl |= TEGRA20_I2S_CTRL_FIFO2_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, + TEGRA20_I2S_CTRL_FIFO2_ENABLE); } static void tegra20_i2s_stop_capture(struct tegra20_i2s *i2s) { - i2s->reg_ctrl &= ~TEGRA20_I2S_CTRL_FIFO2_ENABLE; - tegra20_i2s_write(i2s, TEGRA20_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA20_I2S_CTRL, + TEGRA20_I2S_CTRL_FIFO2_ENABLE, 0); } static int tegra20_i2s_trigger(struct snd_pcm_substream *substream, int cmd, @@ -414,8 +412,6 @@ static __devinit int tegra20_i2s_platform_probe(struct platform_device *pdev) i2s->playback_dma_data.width = 32; i2s->playback_dma_data.req_sel = dma_ch; - i2s->reg_ctrl = TEGRA20_I2S_CTRL_FIFO_FORMAT_PACKED; - pm_runtime_enable(&pdev->dev); if (!pm_runtime_enabled(&pdev->dev)) { ret = tegra20_i2s_runtime_resume(&pdev->dev); diff --git a/sound/soc/tegra/tegra20_i2s.h b/sound/soc/tegra/tegra20_i2s.h index a57efc6..c27069d 100644 --- a/sound/soc/tegra/tegra20_i2s.h +++ b/sound/soc/tegra/tegra20_i2s.h @@ -158,7 +158,6 @@ struct tegra20_i2s { struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index f774a2d..5c33c61 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -37,19 +37,6 @@ #define DRV_NAME "tegra20-spdif" -static inline void tegra20_spdif_write(struct tegra20_spdif *spdif, u32 reg, - u32 val) -{ - regmap_write(spdif->regmap, reg, val); -} - -static inline u32 tegra20_spdif_read(struct tegra20_spdif *spdif, u32 reg) -{ - u32 val; - regmap_read(spdif->regmap, reg, &val); - return val; -} - static int tegra20_spdif_runtime_suspend(struct device *dev) { struct tegra20_spdif *spdif = dev_get_drvdata(dev); @@ -79,19 +66,22 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; int ret, spdifclock; - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_PACK; - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_PACK; - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val = TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; } + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, mask, val); + switch (params_rate(params)) { case 32000: spdifclock = 4096000; @@ -129,14 +119,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, static void tegra20_spdif_start_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl |= TEGRA20_SPDIF_CTRL_TX_EN; - tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, + TEGRA20_SPDIF_CTRL_TX_EN); } static void tegra20_spdif_stop_playback(struct tegra20_spdif *spdif) { - spdif->reg_ctrl &= ~TEGRA20_SPDIF_CTRL_TX_EN; - tegra20_spdif_write(spdif, TEGRA20_SPDIF_CTRL, spdif->reg_ctrl); + regmap_update_bits(spdif->regmap, TEGRA20_SPDIF_CTRL, + TEGRA20_SPDIF_CTRL_TX_EN, 0); } static int tegra20_spdif_trigger(struct snd_pcm_substream *substream, int cmd, diff --git a/sound/soc/tegra/tegra20_spdif.h b/sound/soc/tegra/tegra20_spdif.h index ed75652..b48d699 100644 --- a/sound/soc/tegra/tegra20_spdif.h +++ b/sound/soc/tegra/tegra20_spdif.h @@ -465,7 +465,6 @@ struct tegra20_spdif { struct tegra_pcm_dma_params capture_dma_data; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 2327f62..b68e27a 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -44,18 +44,6 @@ #define DRV_NAME "tegra30-i2s" -static inline void tegra30_i2s_write(struct tegra30_i2s *i2s, u32 reg, u32 val) -{ - regmap_write(i2s->regmap, reg, val); -} - -static inline u32 tegra30_i2s_read(struct tegra30_i2s *i2s, u32 reg) -{ - u32 val; - regmap_read(i2s->regmap, reg, &val); - return val; -} - static int tegra30_i2s_runtime_suspend(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); @@ -128,6 +116,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); + unsigned int mask, val; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -136,10 +125,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_MASTER_ENABLE; + val = TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; @@ -147,33 +136,37 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - i2s->reg_ctrl &= ~(TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | - TEGRA30_I2S_CTRL_LRCK_MASK); + mask |= TEGRA30_I2S_CTRL_FRAME_FORMAT_MASK | + TEGRA30_I2S_CTRL_LRCK_MASK; switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_DSP_B: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_R_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_FSYNC; + val |= TEGRA30_I2S_CTRL_LRCK_R_LOW; break; case SND_SOC_DAIFMT_I2S: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_RIGHT_J: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; case SND_SOC_DAIFMT_LEFT_J: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_LRCK_L_LOW; + val |= TEGRA30_I2S_CTRL_FRAME_FORMAT_LRCK; + val |= TEGRA30_I2S_CTRL_LRCK_L_LOW; break; default: return -EINVAL; } + pm_runtime_get_sync(dai->dev); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + pm_runtime_put(dai->dev); + return 0; } @@ -183,22 +176,24 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - u32 val; + unsigned int mask, val, reg; int ret, sample_size, srate, i2sclock, bitcnt; if (params_channels(params) != 2) return -EINVAL; - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_BIT_SIZE_MASK; + mask = TEGRA30_I2S_CTRL_BIT_SIZE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_BIT_SIZE_16; + val = TEGRA30_I2S_CTRL_BIT_SIZE_16; sample_size = 16; break; default: return -EINVAL; } + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, mask, val); + srate = params_rate(params); /* Final "* 2" required by Tegra hardware */ @@ -219,7 +214,7 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, if (i2sclock % (2 * srate)) val |= TEGRA30_I2S_TIMING_NON_SYM_ENABLE; - tegra30_i2s_write(i2s, TEGRA30_I2S_TIMING, val); + regmap_write(i2s->regmap, TEGRA30_I2S_TIMING, val); val = (0 << TEGRA30_AUDIOCIF_CTRL_FIFO_THRESHOLD_SHIFT) | (1 << TEGRA30_AUDIOCIF_CTRL_AUDIO_CHANNELS_SHIFT) | @@ -229,15 +224,17 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_RX_CTRL, val); + reg = TEGRA30_I2S_CIF_RX_CTRL; } else { val |= TEGRA30_AUDIOCIF_CTRL_DIRECTION_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CIF_TX_CTRL, val); + reg = TEGRA30_I2S_CIF_RX_CTRL; } + regmap_write(i2s->regmap, reg, val); + val = (1 << TEGRA30_I2S_OFFSET_RX_DATA_OFFSET_SHIFT) | (1 << TEGRA30_I2S_OFFSET_TX_DATA_OFFSET_SHIFT); - tegra30_i2s_write(i2s, TEGRA30_I2S_OFFSET, val); + regmap_write(i2s->regmap, TEGRA30_I2S_OFFSET, val); return 0; } @@ -245,29 +242,31 @@ static int tegra30_i2s_hw_params(struct snd_pcm_substream *substream, static void tegra30_i2s_start_playback(struct tegra30_i2s *i2s) { tegra30_ahub_enable_tx_fifo(i2s->playback_fifo_cif); - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, + TEGRA30_I2S_CTRL_XFER_EN_TX); } static void tegra30_i2s_stop_playback(struct tegra30_i2s *i2s) { tegra30_ahub_disable_tx_fifo(i2s->playback_fifo_cif); - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_TX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_TX, 0); } static void tegra30_i2s_start_capture(struct tegra30_i2s *i2s) { tegra30_ahub_enable_rx_fifo(i2s->capture_fifo_cif); - i2s->reg_ctrl |= TEGRA30_I2S_CTRL_XFER_EN_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, + TEGRA30_I2S_CTRL_XFER_EN_RX); } static void tegra30_i2s_stop_capture(struct tegra30_i2s *i2s) { tegra30_ahub_disable_rx_fifo(i2s->capture_fifo_cif); - i2s->reg_ctrl &= ~TEGRA30_I2S_CTRL_XFER_EN_RX; - tegra30_i2s_write(i2s, TEGRA30_I2S_CTRL, i2s->reg_ctrl); + regmap_update_bits(i2s->regmap, TEGRA30_I2S_CTRL, + TEGRA30_I2S_CTRL_XFER_EN_RX, 0); } static int tegra30_i2s_trigger(struct snd_pcm_substream *substream, int cmd, diff --git a/sound/soc/tegra/tegra30_i2s.h b/sound/soc/tegra/tegra30_i2s.h index 91adf29..34dc47b 100644 --- a/sound/soc/tegra/tegra30_i2s.h +++ b/sound/soc/tegra/tegra30_i2s.h @@ -236,7 +236,6 @@ struct tegra30_i2s { enum tegra30_ahub_txcif playback_fifo_cif; struct tegra_pcm_dma_params playback_dma_data; struct regmap *regmap; - u32 reg_ctrl; }; #endif -- cgit v0.10.2 From 7d116684945459e98538c797dca37c54ddd89906 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 10:21:16 +0800 Subject: ASoC: wm8903: Convert to devm_regmap_init_i2c() Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 304b5cf..3abd450 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -2124,7 +2124,7 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, return -ENOMEM; wm8903->dev = &i2c->dev; - wm8903->regmap = regmap_init_i2c(i2c, &wm8903_regmap); + wm8903->regmap = devm_regmap_init_i2c(i2c, &wm8903_regmap); if (IS_ERR(wm8903->regmap)) { ret = PTR_ERR(wm8903->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2191,7 +2191,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, return 0; err: - regmap_exit(wm8903->regmap); return ret; } @@ -2200,7 +2199,6 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) struct wm8903_priv *wm8903 = i2c_get_clientdata(client); wm8903_free_gpio(wm8903); - regmap_exit(wm8903->regmap); snd_soc_unregister_codec(&client->dev); return 0; -- cgit v0.10.2 From 8cb28fd6d1e98fe4cf232d7803093a3b7b46e969 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:57:50 +0800 Subject: ASoC: wm8904: Convert to module_i2c_driver() Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index db94d10..02bc2ca 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2313,23 +2313,7 @@ static struct i2c_driver wm8904_i2c_driver = { .id_table = wm8904_i2c_id, }; -static int __init wm8904_modinit(void) -{ - int ret = 0; - ret = i2c_add_driver(&wm8904_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register wm8904 I2C driver: %d\n", - ret); - } - return ret; -} -module_init(wm8904_modinit); - -static void __exit wm8904_exit(void) -{ - i2c_del_driver(&wm8904_i2c_driver); -} -module_exit(wm8904_exit); +module_i2c_driver(wm8904_i2c_driver); MODULE_DESCRIPTION("ASoC WM8904 driver"); MODULE_AUTHOR("Mark Brown "); -- cgit v0.10.2 From d633edd95dc938f3f5f0d4e431932f4ca042bffb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:26:01 +0800 Subject: ASoC: wm8904: Convert to devm_regmap_init_i2c() Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 02bc2ca..560a9a47 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2263,7 +2263,7 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, if (wm8904 == NULL) return -ENOMEM; - wm8904->regmap = regmap_init_i2c(i2c, &wm8904_regmap); + wm8904->regmap = devm_regmap_init_i2c(i2c, &wm8904_regmap); if (IS_ERR(wm8904->regmap)) { ret = PTR_ERR(wm8904->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -2283,15 +2283,12 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, return 0; err: - regmap_exit(wm8904->regmap); return ret; } static __devexit int wm8904_i2c_remove(struct i2c_client *client) { - struct wm8904_priv *wm8904 = i2c_get_clientdata(client); snd_soc_unregister_codec(&client->dev); - regmap_exit(wm8904->regmap); return 0; } -- cgit v0.10.2 From 679d7abdc7543e56abc41b8f4858f31a91259b29 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 7 Jun 2012 14:00:21 +0200 Subject: ASoC: codecs: Add AB8500 codec-driver Add codec-driver for ST-Ericsson AB8500 mixed-signal ASIC. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 8b879c7..f63776d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -12,6 +12,7 @@ config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" select SND_SOC_88PM860X if MFD_88PM860X select SND_SOC_L3 + select SND_SOC_AB8500_CODEC if ABX500_CORE select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI @@ -132,6 +133,9 @@ config SND_SOC_WM_HUBS default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y default m if SND_SOC_WM8993=m || SND_SOC_WM8994=m +config SND_SOC_AB8500_CODEC + tristate + config SND_SOC_AC97_CODEC tristate select SND_AC97_CODEC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e50811b..fc93b4b 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -1,4 +1,5 @@ snd-soc-88pm860x-objs := 88pm860x-codec.o +snd-soc-ab8500-codec-objs := ab8500-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o @@ -109,6 +110,7 @@ snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o +obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c new file mode 100644 index 0000000..95dc7d5 --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.c @@ -0,0 +1,2521 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja , + * Kristoffer Karlsson , + * Roger Nilsson , + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto , + * Mikko Sarmanne , + * Jarmo K. Kuronen , + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + +#include "ab8500-codec.h" + +/* Macrocell value definitions */ +#define CLK_32K_OUT2_DISABLE 0x01 +#define INACTIVE_RESET_AUDIO 0x02 +#define ENABLE_AUDIO_CLK_TO_AUDIO_BLK 0x10 +#define ENABLE_VINTCORE12_SUPPLY 0x04 +#define GPIO27_DIR_OUTPUT 0x04 +#define GPIO29_DIR_OUTPUT 0x10 +#define GPIO31_DIR_OUTPUT 0x40 + +/* Macrocell register definitions */ +#define AB8500_CTRL3_REG 0x0200 +#define AB8500_GPIO_DIR4_REG 0x1013 + +/* Nr of FIR/IIR-coeff banks in ANC-block */ +#define AB8500_NR_OF_ANC_COEFF_BANKS 2 + +/* Minimum duration to keep ANC IIR Init bit high or +low before proceeding with the configuration sequence */ +#define AB8500_ANC_SM_DELAY 2000 + +#define AB8500_FILTER_CONTROL(xname, xcount, xmin, xmax) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = filter_control_info, \ + .get = filter_control_get, .put = filter_control_put, \ + .private_value = (unsigned long)&(struct filter_control) \ + {.count = xcount, .min = xmin, .max = xmax} } + +struct filter_control { + long min, max; + unsigned int count; + long value[128]; +}; + +/* Sidetone states */ +static const char * const enum_sid_state[] = { + "Unconfigured", + "Apply FIR", + "FIR is configured", +}; +enum sid_state { + SID_UNCONFIGURED = 0, + SID_APPLY_FIR = 1, + SID_FIR_CONFIGURED = 2, +}; + +static const char * const enum_anc_state[] = { + "Unconfigured", + "Apply FIR and IIR", + "FIR and IIR are configured", + "Apply FIR", + "FIR is configured", + "Apply IIR", + "IIR is configured" +}; +enum anc_state { + ANC_UNCONFIGURED = 0, + ANC_APPLY_FIR_IIR = 1, + ANC_FIR_IIR_CONFIGURED = 2, + ANC_APPLY_FIR = 3, + ANC_FIR_CONFIGURED = 4, + ANC_APPLY_IIR = 5, + ANC_IIR_CONFIGURED = 6 +}; + +/* Analog microphones */ +enum amic_idx { + AMIC_IDX_1A, + AMIC_IDX_1B, + AMIC_IDX_2 +}; + +struct ab8500_codec_drvdata_dbg { + struct regulator *vaud; + struct regulator *vamic1; + struct regulator *vamic2; + struct regulator *vdmic; +}; + +/* Private data for AB8500 device-driver */ +struct ab8500_codec_drvdata { + /* Sidetone */ + long *sid_fir_values; + enum sid_state sid_status; + + /* ANC */ + struct mutex anc_lock; + long *anc_fir_values; + long *anc_iir_values; + enum anc_state anc_status; +}; + +static inline const char *amic_micbias_str(enum amic_micbias micbias) +{ + switch (micbias) { + case AMIC_MICBIAS_VAMIC1: + return "VAMIC1"; + case AMIC_MICBIAS_VAMIC2: + return "VAMIC2"; + default: + return "Unknown"; + } +} + +static inline const char *amic_type_str(enum amic_type type) +{ + switch (type) { + case AMIC_TYPE_DIFFERENTIAL: + return "DIFFERENTIAL"; + case AMIC_TYPE_SINGLE_ENDED: + return "SINGLE ENDED"; + default: + return "Unknown"; + } +} + +/* + * Read'n'write functions + */ + +/* Read a register from the audio-bank of AB8500 */ +static unsigned int ab8500_codec_read_reg(struct snd_soc_codec *codec, + unsigned int reg) +{ + int status; + unsigned int value = 0; + + u8 value8; + status = abx500_get_register_interruptible(codec->dev, AB8500_AUDIO, + reg, &value8); + if (status < 0) { + dev_err(codec->dev, + "%s: ERROR: Register (0x%02x:0x%02x) read failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + } else { + dev_dbg(codec->dev, + "%s: Read 0x%02x from register 0x%02x:0x%02x\n", + __func__, value8, (u8)AB8500_AUDIO, (u8)reg); + value = (unsigned int)value8; + } + + return value; +} + +/* Write to a register in the audio-bank of AB8500 */ +static int ab8500_codec_write_reg(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + int status; + + status = abx500_set_register_interruptible(codec->dev, AB8500_AUDIO, + reg, value); + if (status < 0) + dev_err(codec->dev, + "%s: ERROR: Register (%02x:%02x) write failed (%d).\n", + __func__, (u8)AB8500_AUDIO, (u8)reg, status); + else + dev_dbg(codec->dev, + "%s: Wrote 0x%02x into register %02x:%02x\n", + __func__, (u8)value, (u8)AB8500_AUDIO, (u8)reg); + + return status; +} + +/* + * Controls - DAPM + */ + +/* Earpiece */ + +/* Earpiece source selector */ +static const char * const enum_ear_lineout_source[] = {"Headset Left", + "Speaker Left"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ear_lineout_source, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DA3TOEAR, enum_ear_lineout_source); +static const struct snd_kcontrol_new dapm_ear_lineout_source = + SOC_DAPM_ENUM("Earpiece or LineOut Mono Source", + dapm_enum_ear_lineout_source); + +/* LineOut */ + +/* LineOut source selector */ +static const char * const enum_lineout_source[] = {"Mono Path", "Stereo Path"}; +static SOC_ENUM_DOUBLE_DECL(dapm_enum_lineout_source, AB8500_ANACONF5, + AB8500_ANACONF5_HSLDACTOLOL, + AB8500_ANACONF5_HSRDACTOLOR, enum_lineout_source); +static const struct snd_kcontrol_new dapm_lineout_source[] = { + SOC_DAPM_ENUM("LineOut Source", dapm_enum_lineout_source), +}; + +/* Handsfree */ + +/* Speaker Left - ANC selector */ +static const char * const enum_HFx_sel[] = {"Audio Path", "ANC"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFl_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFLSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFl_select[] = { + SOC_DAPM_ENUM("Speaker Left Source", dapm_enum_HFl_sel), +}; + +/* Speaker Right - ANC selector */ +static SOC_ENUM_SINGLE_DECL(dapm_enum_HFr_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_HFRSEL, enum_HFx_sel); +static const struct snd_kcontrol_new dapm_HFr_select[] = { + SOC_DAPM_ENUM("Speaker Right Source", dapm_enum_HFr_sel), +}; + +/* Mic 1 */ + +/* Mic 1 - Mic 1a or 1b selector */ +static const char * const enum_mic1ab_sel[] = {"Mic 1b", "Mic 1a"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic1ab_sel, AB8500_ANACONF3, + AB8500_ANACONF3_MIC1SEL, enum_mic1ab_sel); +static const struct snd_kcontrol_new dapm_mic1ab_mux[] = { + SOC_DAPM_ENUM("Mic 1a or 1b Select", dapm_enum_mic1ab_sel), +}; + +/* Mic 1 - AD3 - Mic 1 or DMic 3 selector */ +static const char * const enum_ad3_sel[] = {"Mic 1", "DMic 3"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad3_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD3SEL, enum_ad3_sel); +static const struct snd_kcontrol_new dapm_ad3_select[] = { + SOC_DAPM_ENUM("AD3 Source Select", dapm_enum_ad3_sel), +}; + +/* Mic 1 - AD6 - Mic 1 or DMic 6 selector */ +static const char * const enum_ad6_sel[] = {"Mic 1", "DMic 6"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad6_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD6SEL, enum_ad6_sel); +static const struct snd_kcontrol_new dapm_ad6_select[] = { + SOC_DAPM_ENUM("AD6 Source Select", dapm_enum_ad6_sel), +}; + +/* Mic 2 */ + +/* Mic 2 - AD5 - Mic 2 or DMic 5 selector */ +static const char * const enum_ad5_sel[] = {"Mic 2", "DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad5_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD5SEL, enum_ad5_sel); +static const struct snd_kcontrol_new dapm_ad5_select[] = { + SOC_DAPM_ENUM("AD5 Source Select", dapm_enum_ad5_sel), +}; + +/* LineIn */ + +/* LineIn left - AD1 - LineIn Left or DMic 1 selector */ +static const char * const enum_ad1_sel[] = {"LineIn Left", "DMic 1"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad1_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD1SEL, enum_ad1_sel); +static const struct snd_kcontrol_new dapm_ad1_select[] = { + SOC_DAPM_ENUM("AD1 Source Select", dapm_enum_ad1_sel), +}; + +/* LineIn right - Mic 2 or LineIn Right selector */ +static const char * const enum_mic2lr_sel[] = {"Mic 2", "LineIn Right"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_mic2lr_sel, AB8500_ANACONF3, + AB8500_ANACONF3_LINRSEL, enum_mic2lr_sel); +static const struct snd_kcontrol_new dapm_mic2lr_select[] = { + SOC_DAPM_ENUM("Mic 2 or LINR Select", dapm_enum_mic2lr_sel), +}; + +/* LineIn right - AD2 - LineIn Right or DMic2 selector */ +static const char * const enum_ad2_sel[] = {"LineIn Right", "DMic 2"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_ad2_sel, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_AD2SEL, enum_ad2_sel); +static const struct snd_kcontrol_new dapm_ad2_select[] = { + SOC_DAPM_ENUM("AD2 Source Select", dapm_enum_ad2_sel), +}; + + +/* ANC */ + +static const char * const enum_anc_in_sel[] = {"Mic 1 / DMic 6", + "Mic 2 / DMic 5"}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_anc_in_sel, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_ANCINSEL, enum_anc_in_sel); +static const struct snd_kcontrol_new dapm_anc_in_select[] = { + SOC_DAPM_ENUM("ANC Source", dapm_enum_anc_in_sel), +}; + +/* ANC - Enable/Disable */ +static const struct snd_kcontrol_new dapm_anc_enable[] = { + SOC_DAPM_SINGLE("Switch", AB8500_ANCCONF1, + AB8500_ANCCONF1_ENANC, 0, 0), +}; + +/* ANC to Earpiece - Mute */ +static const struct snd_kcontrol_new dapm_anc_ear_mute[] = { + SOC_DAPM_SINGLE("Switch", AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_ANCSEL, 1, 0), +}; + + + +/* Sidetone left */ + +/* Sidetone left - Input selector */ +static const char * const enum_stfir1_in_sel[] = { + "LineIn Left", "LineIn Right", "Mic 1", "Headset Left" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir1_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID1SEL, enum_stfir1_in_sel); +static const struct snd_kcontrol_new dapm_stfir1_in_select[] = { + SOC_DAPM_ENUM("Sidetone Left Source", dapm_enum_stfir1_in_sel), +}; + +/* Sidetone right path */ + +/* Sidetone right - Input selector */ +static const char * const enum_stfir2_in_sel[] = { + "LineIn Right", "Mic 1", "DMic 4", "Headset Right" +}; +static SOC_ENUM_SINGLE_DECL(dapm_enum_stfir2_in_sel, AB8500_DIGMULTCONF2, + AB8500_DIGMULTCONF2_FIRSID2SEL, enum_stfir2_in_sel); +static const struct snd_kcontrol_new dapm_stfir2_in_select[] = { + SOC_DAPM_ENUM("Sidetone Right Source", dapm_enum_stfir2_in_sel), +}; + +/* Vibra */ + +static const char * const enum_pwm2vibx[] = {"Audio Path", "PWM Generator"}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib1, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB1, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib1[] = { + SOC_DAPM_ENUM("Vibra 1 Controller", dapm_enum_pwm2vib1), +}; + +static SOC_ENUM_SINGLE_DECL(dapm_enum_pwm2vib2, AB8500_PWMGENCONF1, + AB8500_PWMGENCONF1_PWMTOVIB2, enum_pwm2vibx); + +static const struct snd_kcontrol_new dapm_pwm2vib2[] = { + SOC_DAPM_ENUM("Vibra 2 Controller", dapm_enum_pwm2vib2), +}; + +/* + * DAPM-widgets + */ + +static const struct snd_soc_dapm_widget ab8500_dapm_widgets[] = { + + /* Clocks */ + SND_SOC_DAPM_CLOCK_SUPPLY("audioclk"), + + /* Regulators */ + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AUD", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC1", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-AMIC2", 0), + SND_SOC_DAPM_REGULATOR_SUPPLY("V-DMIC", 0), + + /* Power */ + SND_SOC_DAPM_SUPPLY("Audio Power", + AB8500_POWERUP, AB8500_POWERUP_POWERUP, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("Audio Analog Power", + AB8500_POWERUP, AB8500_POWERUP_ENANA, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* Main supply node */ + SND_SOC_DAPM_SUPPLY("Main Supply", SND_SOC_NOPM, 0, 0, + NULL, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* DA/AD */ + + SND_SOC_DAPM_INPUT("ADC Input"), + SND_SOC_DAPM_ADC("ADC", "ab8500_0c", SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_DAC("DAC", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("DAC Output"), + + SND_SOC_DAPM_AIF_IN("DA_IN1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN5", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("DA_IN6", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT1", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT2", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT3", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT4", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT57", NULL, 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AD_OUT68", NULL, 0, SND_SOC_NOPM, 0, 0), + + /* Headset path */ + + SND_SOC_DAPM_SUPPLY("Charge Pump", AB8500_ANACONF5, + AB8500_ANACONF5_ENCPHS, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DA1 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA1, 0), + SND_SOC_DAPM_DAC("DA2 Enable", "ab8500_0p", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA2, 0), + + SND_SOC_DAPM_PGA("HSL Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Digital Volume", SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_DAC("HSL DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC", "ab8500_0p", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHSR, 0), + SND_SOC_DAPM_MIXER("HSL DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR DAC Mute", AB8500_MUTECONF, + AB8500_MUTECONF_MUTDACHSR, 1, + NULL, 0), + SND_SOC_DAPM_DAC("HSL DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSL, 0), + SND_SOC_DAPM_DAC("HSR DAC Driver", "ab8500_0p", + AB8500_ANACONF3, AB8500_ANACONF3_ENDRVHSR, 0), + + SND_SOC_DAPM_MIXER("HSL Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTHSR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("HSL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HSR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHSR, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_PGA("HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Headset Left"), + SND_SOC_DAPM_OUTPUT("Headset Right"), + + /* LineOut path */ + + SND_SOC_DAPM_MUX("LineOut Source", + SND_SOC_NOPM, 0, 0, dapm_lineout_source), + + SND_SOC_DAPM_MIXER("LOL Disable HFL", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Disable HFR", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 1, + NULL, 0), + + SND_SOC_DAPM_MIXER("LOL Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LOR Enable", + AB8500_ANACONF5, AB8500_ANACONF5_ENLOR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("LineOut Left"), + SND_SOC_DAPM_OUTPUT("LineOut Right"), + + /* Earpiece path */ + + SND_SOC_DAPM_MUX("Earpiece or LineOut Mono Source", + SND_SOC_NOPM, 0, 0, &dapm_ear_lineout_source), + SND_SOC_DAPM_MIXER("EAR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACEAR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Mute", + AB8500_MUTECONF, AB8500_MUTECONF_MUTEAR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("EAR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENEAR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Earpiece"), + + /* Handsfree path */ + + SND_SOC_DAPM_MIXER("DA3 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA4, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Speaker Left Source", + SND_SOC_NOPM, 0, 0, dapm_HFl_select), + SND_SOC_DAPM_MUX("Speaker Right Source", + SND_SOC_NOPM, 0, 0, dapm_HFr_select), + SND_SOC_DAPM_MIXER("HFL DAC", AB8500_DAPATHCONF, + AB8500_DAPATHCONF_ENDACHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACHFR, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA4 or ANC path to HfR", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFREN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA3 or ANC path to HfL", + AB8500_DIGMULTCONF2, AB8500_DIGMULTCONF2_DATOHFLEN, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFL Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("HFR Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENHFR, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Speaker Left"), + SND_SOC_DAPM_OUTPUT("Speaker Right"), + + /* Vibrator path */ + + SND_SOC_DAPM_INPUT("PWMGEN1"), + SND_SOC_DAPM_INPUT("PWMGEN2"), + + SND_SOC_DAPM_MIXER("DA5 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DA6 Channel Volume", + AB8500_DAPATHENA, AB8500_DAPATHENA_ENDA6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB1 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 DAC", + AB8500_DAPATHCONF, AB8500_DAPATHCONF_ENDACVIB2, 0, + NULL, 0), + SND_SOC_DAPM_MUX("Vibra 1 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib1), + SND_SOC_DAPM_MUX("Vibra 2 Controller", + SND_SOC_NOPM, 0, 0, dapm_pwm2vib2), + SND_SOC_DAPM_MIXER("VIB1 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("VIB2 Enable", + AB8500_ANACONF4, AB8500_ANACONF4_ENVIB2, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("Vibra 1"), + SND_SOC_DAPM_OUTPUT("Vibra 2"), + + /* Mic 1 */ + + SND_SOC_DAPM_INPUT("Mic 1"), + + SND_SOC_DAPM_MUX("Mic 1a or 1b Select", + SND_SOC_NOPM, 0, 0, dapm_mic1ab_mux), + SND_SOC_DAPM_MIXER("MIC1 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC1, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1A V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1B V-AMICx Enable", + AB8500_ANACONF2, AB8500_ANACONF2_ENMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC1 ADC", + AB8500_ANACONF3, AB8500_ANACONF3_ENADCMIC, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD3 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad3_select), + SND_SOC_DAPM_MIXER("AD3 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD3 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, 0, + NULL, 0), + + /* Mic 2 */ + + SND_SOC_DAPM_INPUT("Mic 2"), + + SND_SOC_DAPM_MIXER("MIC2 Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTMIC2, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("MIC2 V-AMICx Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENMIC2, 0, + NULL, 0), + + /* LineIn */ + + SND_SOC_DAPM_INPUT("LineIn Left"), + SND_SOC_DAPM_INPUT("LineIn Right"), + + SND_SOC_DAPM_MIXER("LINL Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINL, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Mute", + AB8500_ANACONF2, AB8500_ANACONF2_MUTLINR, 1, + NULL, 0), + SND_SOC_DAPM_MIXER("LINL Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR Enable", AB8500_ANACONF2, + AB8500_ANACONF2_ENLINR, 0, + NULL, 0), + + /* LineIn Bypass path */ + SND_SOC_DAPM_MIXER("LINL to HSL Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR to HSR Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + /* LineIn, Mic 2 */ + SND_SOC_DAPM_MUX("Mic 2 or LINR Select", + SND_SOC_NOPM, 0, 0, dapm_mic2lr_select), + SND_SOC_DAPM_MIXER("LINL ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINL, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("LINR ADC", AB8500_ANACONF3, + AB8500_ANACONF3_ENADCLINR, 0, + NULL, 0), + SND_SOC_DAPM_MUX("AD1 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad1_select), + SND_SOC_DAPM_MUX("AD2 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad2_select), + SND_SOC_DAPM_MIXER("AD1 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD2 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + + SND_SOC_DAPM_MIXER("AD12 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD12, 0, + NULL, 0), + + /* HD Capture path */ + + SND_SOC_DAPM_MUX("AD5 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad5_select), + SND_SOC_DAPM_MUX("AD6 Source Select", + SND_SOC_NOPM, 0, 0, dapm_ad6_select), + SND_SOC_DAPM_MIXER("AD5 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD6 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD57 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD68 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD5768, 0, + NULL, 0), + + /* Digital Microphone path */ + + SND_SOC_DAPM_INPUT("DMic 1"), + SND_SOC_DAPM_INPUT("DMic 2"), + SND_SOC_DAPM_INPUT("DMic 3"), + SND_SOC_DAPM_INPUT("DMic 4"), + SND_SOC_DAPM_INPUT("DMic 5"), + SND_SOC_DAPM_INPUT("DMic 6"), + + SND_SOC_DAPM_MIXER("DMIC1", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC1, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC2", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC2, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC3", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC3, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC4", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC4, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC5", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC5, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("DMIC6", + AB8500_DIGMICCONF, AB8500_DIGMICCONF_ENDMIC6, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Channel Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("AD4 Enable", + AB8500_ADPATHENA, AB8500_ADPATHENA_ENAD34, + 0, NULL, 0), + + /* Acoustical Noise Cancellation path */ + + SND_SOC_DAPM_INPUT("ANC Configure Input"), + SND_SOC_DAPM_OUTPUT("ANC Configure Output"), + + SND_SOC_DAPM_MUX("ANC Source", + SND_SOC_NOPM, 0, 0, + dapm_anc_in_select), + SND_SOC_DAPM_SWITCH("ANC", + SND_SOC_NOPM, 0, 0, + dapm_anc_enable), + SND_SOC_DAPM_SWITCH("ANC to Earpiece", + SND_SOC_NOPM, 0, 0, + dapm_anc_ear_mute), + + /* Sidetone Filter path */ + + SND_SOC_DAPM_MUX("Sidetone Left Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir1_in_select), + SND_SOC_DAPM_MUX("Sidetone Right Source", + SND_SOC_NOPM, 0, 0, + dapm_stfir2_in_select), + SND_SOC_DAPM_MIXER("STFIR1 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Control", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR1 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), + SND_SOC_DAPM_MIXER("STFIR2 Volume", + SND_SOC_NOPM, 0, 0, + NULL, 0), +}; + +/* + * DAPM-routes + */ +static const struct snd_soc_dapm_route ab8500_dapm_routes[] = { + /* Power AB8500 audio-block when AD/DA is active */ + {"Main Supply", NULL, "V-AUD"}, + {"Main Supply", NULL, "audioclk"}, + {"Main Supply", NULL, "Audio Power"}, + {"Main Supply", NULL, "Audio Analog Power"}, + + {"DAC", NULL, "ab8500_0p"}, + {"DAC", NULL, "Main Supply"}, + {"ADC", NULL, "ab8500_0c"}, + {"ADC", NULL, "Main Supply"}, + + /* ANC Configure */ + {"ANC Configure Input", NULL, "Main Supply"}, + {"ANC Configure Output", NULL, "ANC Configure Input"}, + + /* AD/DA */ + {"ADC", NULL, "ADC Input"}, + {"DAC Output", NULL, "DAC"}, + + /* Powerup charge pump if DA1/2 is in use */ + + {"DA_IN1", NULL, "ab8500_0p"}, + {"DA_IN1", NULL, "Charge Pump"}, + {"DA_IN2", NULL, "ab8500_0p"}, + {"DA_IN2", NULL, "Charge Pump"}, + + /* Headset path */ + + {"DA1 Enable", NULL, "DA_IN1"}, + {"DA2 Enable", NULL, "DA_IN2"}, + + {"HSL Digital Volume", NULL, "DA1 Enable"}, + {"HSR Digital Volume", NULL, "DA2 Enable"}, + + {"HSL DAC", NULL, "HSL Digital Volume"}, + {"HSR DAC", NULL, "HSR Digital Volume"}, + + {"HSL DAC Mute", NULL, "HSL DAC"}, + {"HSR DAC Mute", NULL, "HSR DAC"}, + + {"HSL DAC Driver", NULL, "HSL DAC Mute"}, + {"HSR DAC Driver", NULL, "HSR DAC Mute"}, + + {"HSL Mute", NULL, "HSL DAC Driver"}, + {"HSR Mute", NULL, "HSR DAC Driver"}, + + {"HSL Enable", NULL, "HSL Mute"}, + {"HSR Enable", NULL, "HSR Mute"}, + + {"HSL Volume", NULL, "HSL Enable"}, + {"HSR Volume", NULL, "HSR Enable"}, + + {"Headset Left", NULL, "HSL Volume"}, + {"Headset Right", NULL, "HSR Volume"}, + + /* HF or LineOut path */ + + {"DA_IN3", NULL, "ab8500_0p"}, + {"DA3 Channel Volume", NULL, "DA_IN3"}, + {"DA_IN4", NULL, "ab8500_0p"}, + {"DA4 Channel Volume", NULL, "DA_IN4"}, + + {"Speaker Left Source", "Audio Path", "DA3 Channel Volume"}, + {"Speaker Right Source", "Audio Path", "DA4 Channel Volume"}, + + {"DA3 or ANC path to HfL", NULL, "Speaker Left Source"}, + {"DA4 or ANC path to HfR", NULL, "Speaker Right Source"}, + + /* HF path */ + + {"HFL DAC", NULL, "DA3 or ANC path to HfL"}, + {"HFR DAC", NULL, "DA4 or ANC path to HfR"}, + + {"HFL Enable", NULL, "HFL DAC"}, + {"HFR Enable", NULL, "HFR DAC"}, + + {"Speaker Left", NULL, "HFL Enable"}, + {"Speaker Right", NULL, "HFR Enable"}, + + /* Earpiece path */ + + {"Earpiece or LineOut Mono Source", "Headset Left", + "HSL Digital Volume"}, + {"Earpiece or LineOut Mono Source", "Speaker Left", + "DA3 or ANC path to HfL"}, + + {"EAR DAC", NULL, "Earpiece or LineOut Mono Source"}, + + {"EAR Mute", NULL, "EAR DAC"}, + + {"EAR Enable", NULL, "EAR Mute"}, + + {"Earpiece", NULL, "EAR Enable"}, + + /* LineOut path stereo */ + + {"LineOut Source", "Stereo Path", "HSL DAC Driver"}, + {"LineOut Source", "Stereo Path", "HSR DAC Driver"}, + + /* LineOut path mono */ + + {"LineOut Source", "Mono Path", "EAR DAC"}, + + /* LineOut path */ + + {"LOL Disable HFL", NULL, "LineOut Source"}, + {"LOR Disable HFR", NULL, "LineOut Source"}, + + {"LOL Enable", NULL, "LOL Disable HFL"}, + {"LOR Enable", NULL, "LOR Disable HFR"}, + + {"LineOut Left", NULL, "LOL Enable"}, + {"LineOut Right", NULL, "LOR Enable"}, + + /* Vibrator path */ + + {"DA_IN5", NULL, "ab8500_0p"}, + {"DA5 Channel Volume", NULL, "DA_IN5"}, + {"DA_IN6", NULL, "ab8500_0p"}, + {"DA6 Channel Volume", NULL, "DA_IN6"}, + + {"VIB1 DAC", NULL, "DA5 Channel Volume"}, + {"VIB2 DAC", NULL, "DA6 Channel Volume"}, + + {"Vibra 1 Controller", "Audio Path", "VIB1 DAC"}, + {"Vibra 2 Controller", "Audio Path", "VIB2 DAC"}, + {"Vibra 1 Controller", "PWM Generator", "PWMGEN1"}, + {"Vibra 2 Controller", "PWM Generator", "PWMGEN2"}, + + {"VIB1 Enable", NULL, "Vibra 1 Controller"}, + {"VIB2 Enable", NULL, "Vibra 2 Controller"}, + + {"Vibra 1", NULL, "VIB1 Enable"}, + {"Vibra 2", NULL, "VIB2 Enable"}, + + + /* Mic 2 */ + + {"MIC2 V-AMICx Enable", NULL, "Mic 2"}, + + /* LineIn */ + {"LINL Mute", NULL, "LineIn Left"}, + {"LINR Mute", NULL, "LineIn Right"}, + + {"LINL Enable", NULL, "LINL Mute"}, + {"LINR Enable", NULL, "LINR Mute"}, + + /* LineIn, Mic 2 */ + {"Mic 2 or LINR Select", "LineIn Right", "LINR Enable"}, + {"Mic 2 or LINR Select", "Mic 2", "MIC2 V-AMICx Enable"}, + + {"LINL ADC", NULL, "LINL Enable"}, + {"LINR ADC", NULL, "Mic 2 or LINR Select"}, + + {"AD1 Source Select", "LineIn Left", "LINL ADC"}, + {"AD2 Source Select", "LineIn Right", "LINR ADC"}, + + {"AD1 Channel Volume", NULL, "AD1 Source Select"}, + {"AD2 Channel Volume", NULL, "AD2 Source Select"}, + + {"AD12 Enable", NULL, "AD1 Channel Volume"}, + {"AD12 Enable", NULL, "AD2 Channel Volume"}, + + {"AD_OUT1", NULL, "ab8500_0c"}, + {"AD_OUT1", NULL, "AD12 Enable"}, + {"AD_OUT2", NULL, "ab8500_0c"}, + {"AD_OUT2", NULL, "AD12 Enable"}, + + /* Mic 1 */ + + {"MIC1 Mute", NULL, "Mic 1"}, + + {"MIC1A V-AMICx Enable", NULL, "MIC1 Mute"}, + {"MIC1B V-AMICx Enable", NULL, "MIC1 Mute"}, + + {"Mic 1a or 1b Select", "Mic 1a", "MIC1A V-AMICx Enable"}, + {"Mic 1a or 1b Select", "Mic 1b", "MIC1B V-AMICx Enable"}, + + {"MIC1 ADC", NULL, "Mic 1a or 1b Select"}, + + {"AD3 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD3 Channel Volume", NULL, "AD3 Source Select"}, + + {"AD3 Enable", NULL, "AD3 Channel Volume"}, + + {"AD_OUT3", NULL, "ab8500_0c"}, + {"AD_OUT3", NULL, "AD3 Enable"}, + + /* HD Capture path */ + + {"AD5 Source Select", "Mic 2", "LINR ADC"}, + {"AD6 Source Select", "Mic 1", "MIC1 ADC"}, + + {"AD5 Channel Volume", NULL, "AD5 Source Select"}, + {"AD6 Channel Volume", NULL, "AD6 Source Select"}, + + {"AD57 Enable", NULL, "AD5 Channel Volume"}, + {"AD68 Enable", NULL, "AD6 Channel Volume"}, + + {"AD_OUT57", NULL, "ab8500_0c"}, + {"AD_OUT57", NULL, "AD57 Enable"}, + {"AD_OUT68", NULL, "ab8500_0c"}, + {"AD_OUT68", NULL, "AD68 Enable"}, + + /* Digital Microphone path */ + + {"DMic 1", NULL, "V-DMIC"}, + {"DMic 2", NULL, "V-DMIC"}, + {"DMic 3", NULL, "V-DMIC"}, + {"DMic 4", NULL, "V-DMIC"}, + {"DMic 5", NULL, "V-DMIC"}, + {"DMic 6", NULL, "V-DMIC"}, + + {"AD1 Source Select", NULL, "DMic 1"}, + {"AD2 Source Select", NULL, "DMic 2"}, + {"AD3 Source Select", NULL, "DMic 3"}, + {"AD5 Source Select", NULL, "DMic 5"}, + {"AD6 Source Select", NULL, "DMic 6"}, + + {"AD4 Channel Volume", NULL, "DMic 4"}, + {"AD4 Enable", NULL, "AD4 Channel Volume"}, + + {"AD_OUT4", NULL, "ab8500_0c"}, + {"AD_OUT4", NULL, "AD4 Enable"}, + + /* LineIn Bypass path */ + + {"LINL to HSL Volume", NULL, "LINL Enable"}, + {"LINR to HSR Volume", NULL, "LINR Enable"}, + + {"HSL DAC Driver", NULL, "LINL to HSL Volume"}, + {"HSR DAC Driver", NULL, "LINR to HSR Volume"}, + + /* ANC path (Acoustic Noise Cancellation) */ + + {"ANC Source", "Mic 2 / DMic 5", "AD5 Channel Volume"}, + {"ANC Source", "Mic 1 / DMic 6", "AD6 Channel Volume"}, + + {"ANC", "Switch", "ANC Source"}, + + {"Speaker Left Source", "ANC", "ANC"}, + {"Speaker Right Source", "ANC", "ANC"}, + {"ANC to Earpiece", "Switch", "ANC"}, + + {"HSL Digital Volume", NULL, "ANC to Earpiece"}, + + /* Sidetone Filter path */ + + {"Sidetone Left Source", "LineIn Left", "AD12 Enable"}, + {"Sidetone Left Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Left Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Left Source", "Headset Left", "DA_IN1"}, + {"Sidetone Right Source", "LineIn Right", "AD12 Enable"}, + {"Sidetone Right Source", "Mic 1", "AD3 Enable"}, + {"Sidetone Right Source", "DMic 4", "AD4 Enable"}, + {"Sidetone Right Source", "Headset Right", "DA_IN2"}, + + {"STFIR1 Control", NULL, "Sidetone Left Source"}, + {"STFIR2 Control", NULL, "Sidetone Right Source"}, + + {"STFIR1 Volume", NULL, "STFIR1 Control"}, + {"STFIR2 Volume", NULL, "STFIR2 Control"}, + + {"DA1 Enable", NULL, "STFIR1 Volume"}, + {"DA2 Enable", NULL, "STFIR2 Volume"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1a_vamicx[] = { + {"MIC1A V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1A V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic1b_vamicx[] = { + {"MIC1B V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC1B V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +static const struct snd_soc_dapm_route ab8500_dapm_routes_mic2_vamicx[] = { + {"MIC2 V-AMICx Enable", NULL, "V-AMIC1"}, + {"MIC2 V-AMICx Enable", NULL, "V-AMIC2"}, +}; + +/* ANC FIR-coefficients configuration sequence */ +static void anc_fir(struct snd_soc_codec *codec, + unsigned int bnk, unsigned int par, unsigned int val) +{ + if (par == 0 && bnk == 0) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), + BIT(AB8500_ANCCONF1_ANCFIRUPDATE)); + + snd_soc_write(codec, AB8500_ANCCONF5, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF6, val & 0xff); + + if (par == AB8500_ANC_FIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCFIRUPDATE), 0); +} + +/* ANC IIR-coefficients configuration sequence */ +static void anc_iir(struct snd_soc_codec *codec, unsigned int bnk, + unsigned int par, unsigned int val) +{ + if (par == 0) { + if (bnk == 0) { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), + BIT(AB8500_ANCCONF1_ANCIIRINIT)); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRINIT), 0); + usleep_range(AB8500_ANC_SM_DELAY, AB8500_ANC_SM_DELAY); + } else { + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), + BIT(AB8500_ANCCONF1_ANCIIRUPDATE)); + } + } else if (par > 3) { + snd_soc_write(codec, AB8500_ANCCONF7, 0); + snd_soc_write(codec, AB8500_ANCCONF8, val >> 16 & 0xff); + } + + snd_soc_write(codec, AB8500_ANCCONF7, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_ANCCONF8, val & 0xff); + + if (par == AB8500_ANC_IIR_COEFFS - 1 && bnk == 1) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ANCIIRUPDATE), 0); +} + +/* ANC IIR-/FIR-coefficients configuration sequence */ +static void anc_configure(struct snd_soc_codec *codec, + bool apply_fir, bool apply_iir) +{ + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int bnk, par, val; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + if (apply_fir) + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), 0); + + snd_soc_update_bits(codec, AB8500_ANCCONF1, + BIT(AB8500_ANCCONF1_ENANC), BIT(AB8500_ANCCONF1_ENANC)); + + if (apply_fir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_FIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_fir_values[par]); + anc_fir(codec, bnk, par, val); + } + + if (apply_iir) + for (bnk = 0; bnk < AB8500_NR_OF_ANC_COEFF_BANKS; bnk++) + for (par = 0; par < AB8500_ANC_IIR_COEFFS; par++) { + val = snd_soc_read(codec, + drvdata->anc_iir_values[par]); + anc_iir(codec, bnk, par, val); + } + + dev_dbg(codec->dev, "%s: Exit.\n", __func__); +} + +/* + * Control-events + */ + +static int sid_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->sid_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* Write sidetone FIR-coefficients configuration sequence */ +static int sid_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + unsigned int param, sidconf, val; + int status = 1; + + dev_dbg(codec->dev, "%s: Enter\n", __func__); + + if (ucontrol->value.integer.value[0] != SID_APPLY_FIR) { + dev_err(codec->dev, + "%s: ERROR: This control supports '%s' only!\n", + __func__, enum_sid_state[SID_APPLY_FIR]); + return -EIO; + } + + mutex_lock(&codec->mutex); + + sidconf = snd_soc_read(codec, AB8500_SIDFIRCONF); + if (((sidconf & BIT(AB8500_SIDFIRCONF_FIRSIDBUSY)) != 0)) { + if ((sidconf & BIT(AB8500_SIDFIRCONF_ENFIRSIDS)) == 0) { + dev_err(codec->dev, "%s: Sidetone busy while off!\n", + __func__); + status = -EPERM; + } else { + status = -EBUSY; + } + goto out; + } + + snd_soc_write(codec, AB8500_SIDFIRADR, 0); + + for (param = 0; param < AB8500_SID_FIR_COEFFS; param++) { + val = snd_soc_read(codec, drvdata->sid_fir_values[param]); + snd_soc_write(codec, AB8500_SIDFIRCOEF1, val >> 8 & 0xff); + snd_soc_write(codec, AB8500_SIDFIRCOEF2, val & 0xff); + } + + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), + BIT(AB8500_SIDFIRADR_FIRSIDSET)); + snd_soc_update_bits(codec, AB8500_SIDFIRADR, + BIT(AB8500_SIDFIRADR_FIRSIDSET), 0); + + drvdata->sid_status = SID_FIR_CONFIGURED; + +out: + mutex_unlock(&codec->mutex); + + dev_dbg(codec->dev, "%s: Exit\n", __func__); + + return status; +} + +static int anc_status_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + + mutex_lock(&codec->mutex); + ucontrol->value.integer.value[0] = drvdata->anc_status; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int anc_status_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); + struct device *dev = codec->dev; + bool apply_fir, apply_iir; + int req, status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + mutex_lock(&drvdata->anc_lock); + + req = ucontrol->value.integer.value[0]; + if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && + req != ANC_APPLY_IIR) { + dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", + __func__, enum_anc_state[req]); + return -EINVAL; + } + apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; + apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR; + + status = snd_soc_dapm_force_enable_pin(&codec->dapm, + "ANC Configure Input"); + if (status < 0) { + dev_err(dev, + "%s: ERROR: Failed to enable power (status = %d)!\n", + __func__, status); + goto cleanup; + } + snd_soc_dapm_sync(&codec->dapm); + + mutex_lock(&codec->mutex); + anc_configure(codec, apply_fir, apply_iir); + mutex_unlock(&codec->mutex); + + if (apply_fir) { + if (drvdata->anc_status == ANC_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_CONFIGURED; + } + if (apply_iir) { + if (drvdata->anc_status == ANC_FIR_CONFIGURED) + drvdata->anc_status = ANC_FIR_IIR_CONFIGURED; + else if (drvdata->anc_status != ANC_FIR_IIR_CONFIGURED) + drvdata->anc_status = ANC_IIR_CONFIGURED; + } + + status = snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + snd_soc_dapm_sync(&codec->dapm); + +cleanup: + mutex_unlock(&drvdata->anc_lock); + + if (status < 0) + dev_err(dev, "%s: Unable to configure ANC! (status = %d)\n", + __func__, status); + + dev_dbg(dev, "%s: Exit.\n", __func__); + + return (status < 0) ? status : 1; +} + +static int filter_control_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = fc->count; + uinfo->value.integer.min = fc->min; + uinfo->value.integer.max = fc->max; + + return 0; +} + +static int filter_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + ucontrol->value.integer.value[i] = fc->value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +static int filter_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct filter_control *fc = + (struct filter_control *)kcontrol->private_value; + unsigned int i; + + mutex_lock(&codec->mutex); + for (i = 0; i < fc->count; i++) + fc->value[i] = ucontrol->value.integer.value[i]; + mutex_unlock(&codec->mutex); + + return 0; +} + +/* + * Controls - Non-DAPM ASoC + */ + +static DECLARE_TLV_DB_SCALE(adx_dig_gain_tlv, -3200, 100, 1); +/* -32dB = Mute */ + +static DECLARE_TLV_DB_SCALE(dax_dig_gain_tlv, -6300, 100, 1); +/* -63dB = Mute */ + +static DECLARE_TLV_DB_SCALE(hs_ear_dig_gain_tlv, -100, 100, 1); +/* -1dB = Mute */ + +static const unsigned int hs_gain_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0, 3, TLV_DB_SCALE_ITEM(-3200, 400, 0), + 4, 15, TLV_DB_SCALE_ITEM(-1800, 200, 0), +}; + +static DECLARE_TLV_DB_SCALE(mic_gain_tlv, 0, 100, 0); + +static DECLARE_TLV_DB_SCALE(lin_gain_tlv, -1000, 200, 0); + +static DECLARE_TLV_DB_SCALE(lin2hs_gain_tlv, -3800, 200, 1); +/* -38dB = Mute */ + +static const char * const enum_hsfadspeed[] = {"2ms", "0.5ms", "10.6ms", + "5ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsfadspeed, + AB8500_DIGMICCONF, AB8500_DIGMICCONF_HSFADSPEED, enum_hsfadspeed); + +static const char * const enum_envdetthre[] = { + "250mV", "300mV", "350mV", "400mV", + "450mV", "500mV", "550mV", "600mV", + "650mV", "700mV", "750mV", "800mV", + "850mV", "900mV", "950mV", "1.00V" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdeththre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETHTHRE, enum_envdetthre); +static SOC_ENUM_SINGLE_DECL(soc_enum_envdetlthre, + AB8500_ENVCPCONF, AB8500_ENVCPCONF_ENVDETLTHRE, enum_envdetthre); +static const char * const enum_envdettime[] = { + "26.6us", "53.2us", "106us", "213us", + "426us", "851us", "1.70ms", "3.40ms", + "6.81ms", "13.6ms", "27.2ms", "54.5ms", + "109ms", "218ms", "436ms", "872ms" }; +static SOC_ENUM_SINGLE_DECL(soc_enum_envdettime, + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETTIME, enum_envdettime); + +static const char * const enum_sinc31[] = {"Sinc 3", "Sinc 1"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_hsesinc, AB8500_HSLEARDIGGAIN, + AB8500_HSLEARDIGGAIN_HSSINC1, enum_sinc31); + +static const char * const enum_fadespeed[] = {"1ms", "4ms", "8ms", "16ms"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_fadespeed, AB8500_HSRDIGGAIN, + AB8500_HSRDIGGAIN_FADESPEED, enum_fadespeed); + +/* Earpiece */ + +static const char * const enum_lowpow[] = {"Normal", "Low Power"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_eardaclowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDACLOWPOW, enum_lowpow); +static SOC_ENUM_SINGLE_DECL(soc_enum_eardrvlowpow, AB8500_ANACONF1, + AB8500_ANACONF1_EARDRVLOWPOW, enum_lowpow); + +static const char * const enum_av_mode[] = {"Audio", "Voice"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad12voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1VOICE, AB8500_ADFILTCONF_AD2VOICE, enum_av_mode); +static SOC_ENUM_DOUBLE_DECL(soc_enum_ad34voice, AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3VOICE, AB8500_ADFILTCONF_AD4VOICE, enum_av_mode); + +/* DA */ + +static SOC_ENUM_SINGLE_DECL(soc_enum_da12voice, + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DA12VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da34voice, + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DA34VOICE, + enum_av_mode); +static SOC_ENUM_SINGLE_DECL(soc_enum_da56voice, + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DA56VOICE, + enum_av_mode); + +static const char * const enum_da2hslr[] = {"Sidetone", "Audio Path"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_da2hslr, AB8500_DIGMULTCONF1, + AB8500_DIGMULTCONF1_DATOHSLEN, + AB8500_DIGMULTCONF1_DATOHSREN, enum_da2hslr); + +static const char * const enum_sinc53[] = {"Sinc 5", "Sinc 3"}; +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic12sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC1SINC3, + AB8500_DMICFILTCONF_DMIC2SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic34sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC3SINC3, + AB8500_DMICFILTCONF_DMIC4SINC3, enum_sinc53); +static SOC_ENUM_DOUBLE_DECL(soc_enum_dmic56sinc, AB8500_DMICFILTCONF, + AB8500_DMICFILTCONF_DMIC5SINC3, + AB8500_DMICFILTCONF_DMIC6SINC3, enum_sinc53); + +/* Digital interface - DA from slot mapping */ +static const char * const enum_da_from_slot_map[] = {"SLOT0", + "SLOT1", + "SLOT2", + "SLOT3", + "SLOT4", + "SLOT5", + "SLOT6", + "SLOT7", + "SLOT8", + "SLOT9", + "SLOT10", + "SLOT11", + "SLOT12", + "SLOT13", + "SLOT14", + "SLOT15", + "SLOT16", + "SLOT17", + "SLOT18", + "SLOT19", + "SLOT20", + "SLOT21", + "SLOT22", + "SLOT23", + "SLOT24", + "SLOT25", + "SLOT26", + "SLOT27", + "SLOT28", + "SLOT29", + "SLOT30", + "SLOT31"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_da1slotmap, + AB8500_DASLOTCONF1, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da2slotmap, + AB8500_DASLOTCONF2, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da3slotmap, + AB8500_DASLOTCONF3, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da4slotmap, + AB8500_DASLOTCONF4, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da5slotmap, + AB8500_DASLOTCONF5, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da6slotmap, + AB8500_DASLOTCONF6, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da7slotmap, + AB8500_DASLOTCONF7, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_da8slotmap, + AB8500_DASLOTCONF8, AB8500_DASLOTCONFX_SLTODAX_SHIFT, + enum_da_from_slot_map); + +/* Digital interface - AD to slot mapping */ +static const char * const enum_ad_to_slot_map[] = {"AD_OUT1", + "AD_OUT2", + "AD_OUT3", + "AD_OUT4", + "AD_OUT5", + "AD_OUT6", + "AD_OUT7", + "AD_OUT8", + "zeroes", + "tristate"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot0map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot1map, + AB8500_ADSLOTSEL1, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot2map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot3map, + AB8500_ADSLOTSEL2, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot4map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot5map, + AB8500_ADSLOTSEL3, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot6map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot7map, + AB8500_ADSLOTSEL4, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot8map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot9map, + AB8500_ADSLOTSEL5, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot10map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot11map, + AB8500_ADSLOTSEL6, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot12map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot13map, + AB8500_ADSLOTSEL7, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot14map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot15map, + AB8500_ADSLOTSEL8, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot16map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot17map, + AB8500_ADSLOTSEL9, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot18map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot19map, + AB8500_ADSLOTSEL10, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot20map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot21map, + AB8500_ADSLOTSEL11, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot22map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot23map, + AB8500_ADSLOTSEL12, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot24map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot25map, + AB8500_ADSLOTSEL13, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot26map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot27map, + AB8500_ADSLOTSEL14, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot28map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot29map, + AB8500_ADSLOTSEL15, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot30map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_EVEN_SHIFT, + enum_ad_to_slot_map); +static SOC_ENUM_SINGLE_DECL(soc_enum_adslot31map, + AB8500_ADSLOTSEL16, AB8500_ADSLOTSELX_ODD_SHIFT, + enum_ad_to_slot_map); + +/* Digital interface - Burst mode */ +static const char * const enum_mask[] = {"Unmasked", "Masked"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomask, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOMASK, + enum_mask); +static const char * const enum_bitclk0[] = {"19_2_MHz", "38_4_MHz"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifo19m2, + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFO19M2, + enum_bitclk0); +static const char * const enum_slavemaster[] = {"Slave", "Master"}; +static SOC_ENUM_SINGLE_DECL(soc_enum_bfifomast, + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOMAST_SHIFT, + enum_slavemaster); + +/* Sidetone */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_sidstate, enum_sid_state); + +/* ANC */ +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_ancstate, enum_anc_state); + +static struct snd_kcontrol_new ab8500_ctrls[] = { + /* Charge pump */ + SOC_ENUM("Charge Pump High Threshold For Low Voltage", + soc_enum_envdeththre), + SOC_ENUM("Charge Pump Low Threshold For Low Voltage", + soc_enum_envdetlthre), + SOC_SINGLE("Charge Pump Envelope Detection Switch", + AB8500_SIGENVCONF, AB8500_SIGENVCONF_ENVDETCPEN, + 1, 0), + SOC_ENUM("Charge Pump Envelope Detection Decay Time", + soc_enum_envdettime), + + /* Headset */ + SOC_ENUM("Headset Mode", soc_enum_da12voice), + SOC_SINGLE("Headset High Pass Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSHPEN, + 1, 0), + SOC_SINGLE("Headset Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_HSLOWPOW, + 1, 0), + SOC_SINGLE("Headset DAC Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW1, + 1, 0), + SOC_SINGLE("Headset DAC Drv Low Power Switch", + AB8500_ANACONF1, AB8500_ANACONF1_DACLOWPOW0, + 1, 0), + SOC_ENUM("Headset Fade Speed", soc_enum_hsfadspeed), + SOC_ENUM("Headset Source", soc_enum_da2hslr), + SOC_ENUM("Headset Filter", soc_enum_hsesinc), + SOC_DOUBLE_R_TLV("Headset Master Volume", + AB8500_DADIGGAIN1, AB8500_DADIGGAIN2, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + SOC_DOUBLE_R_TLV("Headset Digital Volume", + AB8500_HSLEARDIGGAIN, AB8500_HSRDIGGAIN, + 0, AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX, 1, hs_ear_dig_gain_tlv), + SOC_DOUBLE_TLV("Headset Volume", + AB8500_ANAGAIN3, + AB8500_ANAGAIN3_HSLGAIN, AB8500_ANAGAIN3_HSRGAIN, + AB8500_ANAGAIN3_HSXGAIN_MAX, 1, hs_gain_tlv), + + /* Earpiece */ + SOC_ENUM("Earpiece DAC Mode", + soc_enum_eardaclowpow), + SOC_ENUM("Earpiece DAC Drv Mode", + soc_enum_eardrvlowpow), + + /* HandsFree */ + SOC_ENUM("HF Mode", soc_enum_da34voice), + SOC_SINGLE("HF and Headset Swap Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_SWAPDA12_34, + 1, 0), + SOC_DOUBLE("HF Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_HFLSWAPEN, AB8500_CLASSDCONF1_HFRSWAPEN, + 1, 0), + SOC_DOUBLE("HF FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP0, AB8500_CLASSDCONF2_FIRBYP1, + 1, 0), + SOC_DOUBLE("HF High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN0, AB8500_CLASSDCONF2_HIGHVOLEN1, + 1, 0), + SOC_SINGLE("HF L and R Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLHF, + 1, 0), + SOC_DOUBLE_R_TLV("HF Master Volume", + AB8500_DADIGGAIN3, AB8500_DADIGGAIN4, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Vibra */ + SOC_DOUBLE("Vibra High Volume Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_HIGHVOLEN2, AB8500_CLASSDCONF2_HIGHVOLEN3, + 1, 0), + SOC_DOUBLE("Vibra Low EMI Mode Switch", + AB8500_CLASSDCONF1, + AB8500_CLASSDCONF1_VIB1SWAPEN, AB8500_CLASSDCONF1_VIB2SWAPEN, + 1, 0), + SOC_DOUBLE("Vibra FIR Bypass Switch", + AB8500_CLASSDCONF2, + AB8500_CLASSDCONF2_FIRBYP2, AB8500_CLASSDCONF2_FIRBYP3, + 1, 0), + SOC_ENUM("Vibra Mode", soc_enum_da56voice), + SOC_DOUBLE_R("Vibra PWM Duty Cycle N", + AB8500_PWMGENCONF3, AB8500_PWMGENCONF5, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_DOUBLE_R("Vibra PWM Duty Cycle P", + AB8500_PWMGENCONF2, AB8500_PWMGENCONF4, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC, + AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX, 0), + SOC_SINGLE("Vibra 1 and 2 Bridge Switch", + AB8500_CLASSDCONF1, AB8500_CLASSDCONF1_PARLVIB, + 1, 0), + SOC_DOUBLE_R_TLV("Vibra Master Volume", + AB8500_DADIGGAIN5, AB8500_DADIGGAIN6, + 0, AB8500_DADIGGAINX_DAXGAIN_MAX, 1, dax_dig_gain_tlv), + + /* HandsFree, Vibra */ + SOC_SINGLE("ClassD High Pass Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHHPGAIN, + AB8500_CLASSDCONF3_DITHHPGAIN_MAX, 0), + SOC_SINGLE("ClassD White Volume", + AB8500_CLASSDCONF3, AB8500_CLASSDCONF3_DITHWGAIN, + AB8500_CLASSDCONF3_DITHWGAIN_MAX, 0), + + /* Mic 1, Mic 2, LineIn */ + SOC_DOUBLE_R_TLV("Mic Master Volume", + AB8500_ADDIGGAIN3, AB8500_ADDIGGAIN4, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Mic 1 */ + SOC_SINGLE_TLV("Mic 1", + AB8500_ANAGAIN1, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 1 Low Power Switch", + AB8500_ANAGAIN1, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* Mic 2 */ + SOC_DOUBLE("Mic High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD3NH, AB8500_ADFILTCONF_AD4NH, + 1, 1), + SOC_ENUM("Mic Mode", soc_enum_ad34voice), + SOC_ENUM("Mic Filter", soc_enum_dmic34sinc), + SOC_SINGLE_TLV("Mic 2", + AB8500_ANAGAIN2, + AB8500_ANAGAINX_MICXGAIN, + AB8500_ANAGAINX_MICXGAIN_MAX, 0, mic_gain_tlv), + SOC_SINGLE("Mic 2 Low Power Switch", + AB8500_ANAGAIN2, AB8500_ANAGAINX_LOWPOWMICX, + 1, 0), + + /* LineIn */ + SOC_DOUBLE("LineIn High Pass Switch", + AB8500_ADFILTCONF, + AB8500_ADFILTCONF_AD1NH, AB8500_ADFILTCONF_AD2NH, + 1, 1), + SOC_ENUM("LineIn Filter", soc_enum_dmic12sinc), + SOC_ENUM("LineIn Mode", soc_enum_ad12voice), + SOC_DOUBLE_R_TLV("LineIn Master Volume", + AB8500_ADDIGGAIN1, AB8500_ADDIGGAIN2, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + SOC_DOUBLE_TLV("LineIn", + AB8500_ANAGAIN4, + AB8500_ANAGAIN4_LINLGAIN, AB8500_ANAGAIN4_LINRGAIN, + AB8500_ANAGAIN4_LINXGAIN_MAX, 0, lin_gain_tlv), + SOC_DOUBLE_R_TLV("LineIn to Headset Volume", + AB8500_DIGLINHSLGAIN, AB8500_DIGLINHSRGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN, + AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX, + 1, lin2hs_gain_tlv), + + /* DMic */ + SOC_ENUM("DMic Filter", soc_enum_dmic56sinc), + SOC_DOUBLE_R_TLV("DMic Master Volume", + AB8500_ADDIGGAIN5, AB8500_ADDIGGAIN6, + 0, AB8500_ADDIGGAINX_ADXGAIN_MAX, 1, adx_dig_gain_tlv), + + /* Digital gains */ + SOC_ENUM("Digital Gain Fade Speed", soc_enum_fadespeed), + + /* Analog loopback */ + SOC_DOUBLE_R_TLV("Analog Loopback Volume", + AB8500_ADDIGLOOPGAIN1, AB8500_ADDIGLOOPGAIN2, + 0, AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX, 1, dax_dig_gain_tlv), + + /* Digital interface - DA from slot mapping */ + SOC_ENUM("Digital Interface DA 1 From Slot Map", soc_enum_da1slotmap), + SOC_ENUM("Digital Interface DA 2 From Slot Map", soc_enum_da2slotmap), + SOC_ENUM("Digital Interface DA 3 From Slot Map", soc_enum_da3slotmap), + SOC_ENUM("Digital Interface DA 4 From Slot Map", soc_enum_da4slotmap), + SOC_ENUM("Digital Interface DA 5 From Slot Map", soc_enum_da5slotmap), + SOC_ENUM("Digital Interface DA 6 From Slot Map", soc_enum_da6slotmap), + SOC_ENUM("Digital Interface DA 7 From Slot Map", soc_enum_da7slotmap), + SOC_ENUM("Digital Interface DA 8 From Slot Map", soc_enum_da8slotmap), + + /* Digital interface - AD to slot mapping */ + SOC_ENUM("Digital Interface AD To Slot 0 Map", soc_enum_adslot0map), + SOC_ENUM("Digital Interface AD To Slot 1 Map", soc_enum_adslot1map), + SOC_ENUM("Digital Interface AD To Slot 2 Map", soc_enum_adslot2map), + SOC_ENUM("Digital Interface AD To Slot 3 Map", soc_enum_adslot3map), + SOC_ENUM("Digital Interface AD To Slot 4 Map", soc_enum_adslot4map), + SOC_ENUM("Digital Interface AD To Slot 5 Map", soc_enum_adslot5map), + SOC_ENUM("Digital Interface AD To Slot 6 Map", soc_enum_adslot6map), + SOC_ENUM("Digital Interface AD To Slot 7 Map", soc_enum_adslot7map), + SOC_ENUM("Digital Interface AD To Slot 8 Map", soc_enum_adslot8map), + SOC_ENUM("Digital Interface AD To Slot 9 Map", soc_enum_adslot9map), + SOC_ENUM("Digital Interface AD To Slot 10 Map", soc_enum_adslot10map), + SOC_ENUM("Digital Interface AD To Slot 11 Map", soc_enum_adslot11map), + SOC_ENUM("Digital Interface AD To Slot 12 Map", soc_enum_adslot12map), + SOC_ENUM("Digital Interface AD To Slot 13 Map", soc_enum_adslot13map), + SOC_ENUM("Digital Interface AD To Slot 14 Map", soc_enum_adslot14map), + SOC_ENUM("Digital Interface AD To Slot 15 Map", soc_enum_adslot15map), + SOC_ENUM("Digital Interface AD To Slot 16 Map", soc_enum_adslot16map), + SOC_ENUM("Digital Interface AD To Slot 17 Map", soc_enum_adslot17map), + SOC_ENUM("Digital Interface AD To Slot 18 Map", soc_enum_adslot18map), + SOC_ENUM("Digital Interface AD To Slot 19 Map", soc_enum_adslot19map), + SOC_ENUM("Digital Interface AD To Slot 20 Map", soc_enum_adslot20map), + SOC_ENUM("Digital Interface AD To Slot 21 Map", soc_enum_adslot21map), + SOC_ENUM("Digital Interface AD To Slot 22 Map", soc_enum_adslot22map), + SOC_ENUM("Digital Interface AD To Slot 23 Map", soc_enum_adslot23map), + SOC_ENUM("Digital Interface AD To Slot 24 Map", soc_enum_adslot24map), + SOC_ENUM("Digital Interface AD To Slot 25 Map", soc_enum_adslot25map), + SOC_ENUM("Digital Interface AD To Slot 26 Map", soc_enum_adslot26map), + SOC_ENUM("Digital Interface AD To Slot 27 Map", soc_enum_adslot27map), + SOC_ENUM("Digital Interface AD To Slot 28 Map", soc_enum_adslot28map), + SOC_ENUM("Digital Interface AD To Slot 29 Map", soc_enum_adslot29map), + SOC_ENUM("Digital Interface AD To Slot 30 Map", soc_enum_adslot30map), + SOC_ENUM("Digital Interface AD To Slot 31 Map", soc_enum_adslot31map), + + /* Digital interface - Loopback */ + SOC_SINGLE("Digital Interface AD 1 Loopback Switch", + AB8500_DASLOTCONF1, AB8500_DASLOTCONF1_DAI7TOADO1, + 1, 0), + SOC_SINGLE("Digital Interface AD 2 Loopback Switch", + AB8500_DASLOTCONF2, AB8500_DASLOTCONF2_DAI8TOADO2, + 1, 0), + SOC_SINGLE("Digital Interface AD 3 Loopback Switch", + AB8500_DASLOTCONF3, AB8500_DASLOTCONF3_DAI7TOADO3, + 1, 0), + SOC_SINGLE("Digital Interface AD 4 Loopback Switch", + AB8500_DASLOTCONF4, AB8500_DASLOTCONF4_DAI8TOADO4, + 1, 0), + SOC_SINGLE("Digital Interface AD 5 Loopback Switch", + AB8500_DASLOTCONF5, AB8500_DASLOTCONF5_DAI7TOADO5, + 1, 0), + SOC_SINGLE("Digital Interface AD 6 Loopback Switch", + AB8500_DASLOTCONF6, AB8500_DASLOTCONF6_DAI8TOADO6, + 1, 0), + SOC_SINGLE("Digital Interface AD 7 Loopback Switch", + AB8500_DASLOTCONF7, AB8500_DASLOTCONF7_DAI8TOADO7, + 1, 0), + SOC_SINGLE("Digital Interface AD 8 Loopback Switch", + AB8500_DASLOTCONF8, AB8500_DASLOTCONF8_DAI7TOADO8, + 1, 0), + + /* Digital interface - Burst FIFO */ + SOC_SINGLE("Digital Interface 0 FIFO Enable Switch", + AB8500_DIGIFCONF3, AB8500_DIGIFCONF3_IF0BFIFOEN, + 1, 0), + SOC_ENUM("Burst FIFO Mask", soc_enum_bfifomask), + SOC_ENUM("Burst FIFO Bit-clock Frequency", soc_enum_bfifo19m2), + SOC_SINGLE("Burst FIFO Threshold", + AB8500_FIFOCONF1, AB8500_FIFOCONF1_BFIFOINT_SHIFT, + AB8500_FIFOCONF1_BFIFOINT_MAX, 0), + SOC_SINGLE("Burst FIFO Length", + AB8500_FIFOCONF2, AB8500_FIFOCONF2_BFIFOTX_SHIFT, + AB8500_FIFOCONF2_BFIFOTX_MAX, 0), + SOC_SINGLE("Burst FIFO EOS Extra Slots", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFOEXSL_SHIFT, + AB8500_FIFOCONF3_BFIFOEXSL_MAX, 0), + SOC_SINGLE("Burst FIFO FS Extra Bit-clocks", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_PREBITCLK0_SHIFT, + AB8500_FIFOCONF3_PREBITCLK0_MAX, 0), + SOC_ENUM("Burst FIFO Interface Mode", soc_enum_bfifomast), + + SOC_SINGLE("Burst FIFO Interface Switch", + AB8500_FIFOCONF3, AB8500_FIFOCONF3_BFIFORUN_SHIFT, + 1, 0), + SOC_SINGLE("Burst FIFO Switch Frame Number", + AB8500_FIFOCONF4, AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT, + AB8500_FIFOCONF4_BFIFOFRAMSW_MAX, 0), + SOC_SINGLE("Burst FIFO Wake Up Delay", + AB8500_FIFOCONF5, AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT, + AB8500_FIFOCONF5_BFIFOWAKEUP_MAX, 0), + SOC_SINGLE("Burst FIFO Samples In FIFO", + AB8500_FIFOCONF6, AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT, + AB8500_FIFOCONF6_BFIFOSAMPLE_MAX, 0), + + /* ANC */ + SOC_ENUM_EXT("ANC Status", soc_enum_ancstate, + anc_status_control_get, anc_status_control_put), + SOC_SINGLE_XR_SX("ANC Warp Delay Shift", + AB8500_ANCCONF2, 1, AB8500_ANCCONF2_SHIFT, + AB8500_ANCCONF2_MIN, AB8500_ANCCONF2_MAX, 0), + SOC_SINGLE_XR_SX("ANC FIR Output Shift", + AB8500_ANCCONF3, 1, AB8500_ANCCONF3_SHIFT, + AB8500_ANCCONF3_MIN, AB8500_ANCCONF3_MAX, 0), + SOC_SINGLE_XR_SX("ANC IIR Output Shift", + AB8500_ANCCONF4, 1, AB8500_ANCCONF4_SHIFT, + AB8500_ANCCONF4_MIN, AB8500_ANCCONF4_MAX, 0), + SOC_SINGLE_XR_SX("ANC Warp Delay", + AB8500_ANCCONF9, 2, AB8500_ANC_WARP_DELAY_SHIFT, + AB8500_ANC_WARP_DELAY_MIN, AB8500_ANC_WARP_DELAY_MAX, 0), + + /* Sidetone */ + SOC_ENUM_EXT("Sidetone Status", soc_enum_sidstate, + sid_status_control_get, sid_status_control_put), + SOC_SINGLE_STROBE("Sidetone Reset", + AB8500_SIDFIRADR, AB8500_SIDFIRADR_FIRSIDSET, 0), +}; + +static struct snd_kcontrol_new ab8500_filter_controls[] = { + AB8500_FILTER_CONTROL("ANC FIR Coefficients", AB8500_ANC_FIR_COEFFS, + AB8500_ANC_FIR_COEFF_MIN, AB8500_ANC_FIR_COEFF_MAX), + AB8500_FILTER_CONTROL("ANC IIR Coefficients", AB8500_ANC_IIR_COEFFS, + AB8500_ANC_IIR_COEFF_MIN, AB8500_ANC_IIR_COEFF_MAX), + AB8500_FILTER_CONTROL("Sidetone FIR Coefficients", + AB8500_SID_FIR_COEFFS, AB8500_SID_FIR_COEFF_MIN, + AB8500_SID_FIR_COEFF_MAX) +}; +enum ab8500_filter { + AB8500_FILTER_ANC_FIR = 0, + AB8500_FILTER_ANC_IIR = 1, + AB8500_FILTER_SID_FIR = 2, +}; + +/* + * Extended interface for codec-driver + */ + +static int ab8500_audio_init_audioblock(struct snd_soc_codec *codec) +{ + int status; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Reset audio-registers and disable 32kHz-clock output 2 */ + status = ab8500_sysctrl_write(AB8500_STW4500CTRL3, + AB8500_STW4500CTRL3_CLK32KOUT2DIS | + AB8500_STW4500CTRL3_RESETAUDN, + AB8500_STW4500CTRL3_RESETAUDN); + if (status < 0) + return status; + + return 0; +} + +static int ab8500_audio_setup_mics(struct snd_soc_codec *codec, + struct amic_settings *amics) +{ + u8 value8; + unsigned int value; + int status; + const struct snd_soc_dapm_route *route; + + dev_dbg(codec->dev, "%s: Enter.\n", __func__); + + /* Set DMic-clocks to outputs */ + status = abx500_get_register_interruptible(codec->dev, (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + &value8); + if (status < 0) + return status; + value = value8 | GPIO27_DIR_OUTPUT | GPIO29_DIR_OUTPUT | + GPIO31_DIR_OUTPUT; + status = abx500_set_register_interruptible(codec->dev, + (u8)AB8500_MISC, + (u8)AB8500_GPIO_DIR4_REG, + value); + if (status < 0) + return status; + + /* Attach regulators to AMic DAPM-paths */ + dev_dbg(codec->dev, "%s: Mic 1a regulator: %s\n", __func__, + amic_micbias_str(amics->mic1a_micbias)); + route = &ab8500_dapm_routes_mic1a_vamicx[amics->mic1a_micbias]; + status = snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 1b regulator: %s\n", __func__, + amic_micbias_str(amics->mic1b_micbias)); + route = &ab8500_dapm_routes_mic1b_vamicx[amics->mic1b_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + dev_dbg(codec->dev, "%s: Mic 2 regulator: %s\n", __func__, + amic_micbias_str(amics->mic2_micbias)); + route = &ab8500_dapm_routes_mic2_vamicx[amics->mic2_micbias]; + status |= snd_soc_dapm_add_routes(&codec->dapm, route, 1); + if (status < 0) { + dev_err(codec->dev, + "%s: Failed to add AMic-regulator DAPM-routes (%d).\n", + __func__, status); + return status; + } + + /* Set AMic-configuration */ + dev_dbg(codec->dev, "%s: Mic 1 mic-type: %s\n", __func__, + amic_type_str(amics->mic1_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN1, AB8500_ANAGAINX_ENSEMICX, + amics->mic1_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + dev_dbg(codec->dev, "%s: Mic 2 mic-type: %s\n", __func__, + amic_type_str(amics->mic2_type)); + snd_soc_update_bits(codec, AB8500_ANAGAIN2, AB8500_ANAGAINX_ENSEMICX, + amics->mic2_type == AMIC_TYPE_DIFFERENTIAL ? + 0 : AB8500_ANAGAINX_ENSEMICX); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_setup_mics); + +static int ab8500_audio_set_ear_cmv(struct snd_soc_codec *codec, + enum ear_cm_voltage ear_cmv) +{ + char *cmv_str; + + switch (ear_cmv) { + case EAR_CMV_0_95V: + cmv_str = "0.95V"; + break; + case EAR_CMV_1_10V: + cmv_str = "1.10V"; + break; + case EAR_CMV_1_27V: + cmv_str = "1.27V"; + break; + case EAR_CMV_1_58V: + cmv_str = "1.58V"; + break; + default: + dev_err(codec->dev, + "%s: Unknown earpiece CM-voltage (%d)!\n", + __func__, (int)ear_cmv); + return -EINVAL; + } + dev_dbg(codec->dev, "%s: Earpiece CM-voltage: %s\n", __func__, + cmv_str); + snd_soc_update_bits(codec, AB8500_ANACONF1, AB8500_ANACONF1_EARSELCM, + ear_cmv); + + return 0; +} +EXPORT_SYMBOL_GPL(ab8500_audio_set_ear_cmv); + +static int ab8500_audio_set_bit_delay(struct snd_soc_dai *dai, + unsigned int delay) +{ + unsigned int mask, val; + struct snd_soc_codec *codec = dai->codec; + + mask = BIT(AB8500_DIGIFCONF2_IF0DEL); + val = 0; + + switch (delay) { + case 0: + break; + case 1: + val |= BIT(AB8500_DIGIFCONF2_IF0DEL); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported bit-delay (0x%x)!\n", + __func__, delay); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 Bit-delay: %d bits.\n", + __func__, delay); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +/* Gates clocking according format mask */ +static int ab8500_codec_set_dai_clock_gate(struct snd_soc_codec *codec, + unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + + mask = BIT(AB8500_DIGIFCONF1_ENMASTGEN) | + BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + + val = BIT(AB8500_DIGIFCONF1_ENMASTGEN); + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: /* continuous clock */ + dev_dbg(codec->dev, "%s: IF0 Clock is continuous.\n", + __func__); + val |= BIT(AB8500_DIGIFCONF1_ENFSBITCLK0); + break; + case SND_SOC_DAIFMT_GATED: /* clock is gated */ + dev_dbg(codec->dev, "%s: IF0 Clock is gated.\n", + __func__); + break; + default: + dev_err(codec->dev, + "%s: ERROR: Unsupported clock mask (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_CLOCK_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + unsigned int mask; + unsigned int val; + struct snd_soc_codec *codec = dai->codec; + int status; + + dev_dbg(codec->dev, "%s: Enter (fmt = 0x%x)\n", __func__, fmt); + + mask = BIT(AB8500_DIGIFCONF3_IF1DATOIF0AD) | + BIT(AB8500_DIGIFCONF3_IF1CLKTOIF0CLK) | + BIT(AB8500_DIGIFCONF3_IF0BFIFOEN) | + BIT(AB8500_DIGIFCONF3_IF0MASTER); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: /* codec clk & FRM master */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 master.\n", __func__); + val |= BIT(AB8500_DIGIFCONF3_IF0MASTER); + break; + case SND_SOC_DAIFMT_CBS_CFS: /* codec clk & FRM slave */ + dev_dbg(dai->codec->dev, + "%s: IF0 Master-mode: AB8500 slave.\n", __func__); + break; + case SND_SOC_DAIFMT_CBS_CFM: /* codec clk slave & FRM master */ + case SND_SOC_DAIFMT_CBM_CFS: /* codec clk master & frame slave */ + dev_err(dai->codec->dev, + "%s: ERROR: The device is either a master or a slave.\n", + __func__); + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupporter master mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + break; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF3, mask, val); + + /* Set clock gating */ + status = ab8500_codec_set_dai_clock_gate(codec, fmt); + if (status) { + dev_err(dai->codec->dev, + "%s: ERRROR: Failed to set clock gate (%d).\n", + __func__, status); + return status; + } + + /* Setting data transfer format */ + + mask = BIT(AB8500_DIGIFCONF2_IF0FORMAT0) | + BIT(AB8500_DIGIFCONF2_IF0FORMAT1) | + BIT(AB8500_DIGIFCONF2_FSYNC0P) | + BIT(AB8500_DIGIFCONF2_BITCLK0P); + val = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: /* I2S mode */ + dev_dbg(dai->codec->dev, "%s: IF0 Protocol: I2S\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT1); + ab8500_audio_set_bit_delay(dai, 0); + break; + + case SND_SOC_DAIFMT_DSP_A: /* L data MSB after FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP A (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 1); + break; + + case SND_SOC_DAIFMT_DSP_B: /* L data MSB during FRM LRC */ + dev_dbg(dai->codec->dev, + "%s: IF0 Protocol: DSP B (TDM)\n", __func__); + val |= BIT(AB8500_DIGIFCONF2_IF0FORMAT0); + ab8500_audio_set_bit_delay(dai, 0); + break; + + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported format (0x%x)!\n", + __func__, fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: /* normal bit clock + frame */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, normal frame\n", + __func__); + break; + case SND_SOC_DAIFMT_NB_IF: /* normal BCLK + inv FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Normal bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + break; + case SND_SOC_DAIFMT_IB_NF: /* invert BCLK + nor FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, normal frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + case SND_SOC_DAIFMT_IB_IF: /* invert BCLK + FRM */ + dev_dbg(dai->codec->dev, + "%s: IF0: Inverted bit clock, inverted frame\n", + __func__); + val |= BIT(AB8500_DIGIFCONF2_FSYNC0P); + val |= BIT(AB8500_DIGIFCONF2_BITCLK0P); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported INV mask 0x%x\n", + __func__, fmt & SND_SOC_DAIFMT_INV_MASK); + return -EINVAL; + } + + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + return 0; +} + +static int ab8500_codec_set_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int val, mask, slots_active; + + mask = BIT(AB8500_DIGIFCONF2_IF0WL0) | + BIT(AB8500_DIGIFCONF2_IF0WL1); + val = 0; + + switch (slot_width) { + case 16: + break; + case 20: + val |= BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + case 24: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1); + break; + case 32: + val |= BIT(AB8500_DIGIFCONF2_IF0WL1) | + BIT(AB8500_DIGIFCONF2_IF0WL0); + break; + default: + dev_err(dai->codec->dev, "%s: Unsupported slot-width 0x%x\n", + __func__, slot_width); + return -EINVAL; + } + + dev_dbg(dai->codec->dev, "%s: IF0 slot-width: %d bits.\n", + __func__, slot_width); + snd_soc_update_bits(codec, AB8500_DIGIFCONF2, mask, val); + + /* Setup TDM clocking according to slot count */ + dev_dbg(dai->codec->dev, "%s: Slots, total: %d\n", __func__, slots); + mask = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + switch (slots) { + case 2: + val = AB8500_MASK_NONE; + break; + case 4: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0); + break; + case 8: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + case 16: + val = BIT(AB8500_DIGIFCONF1_IF0BITCLKOS0) | + BIT(AB8500_DIGIFCONF1_IF0BITCLKOS1); + break; + default: + dev_err(dai->codec->dev, + "%s: ERROR: Unsupported number of slots (%d)!\n", + __func__, slots); + return -EINVAL; + } + snd_soc_update_bits(codec, AB8500_DIGIFCONF1, mask, val); + + /* Setup TDM DA according to active tx slots */ + mask = AB8500_DASLOTCONFX_SLTODAX_MASK; + slots_active = hweight32(tx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, TX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* Slot 9 -> DA_IN1 & DA_IN3 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + break; + case 2: + /* Slot 9 -> DA_IN1 & DA_IN3, Slot 11 -> DA_IN2 & DA_IN4 */ + snd_soc_update_bits(codec, AB8500_DASLOTCONF1, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF3, mask, 9); + snd_soc_update_bits(codec, AB8500_DASLOTCONF2, mask, 11); + snd_soc_update_bits(codec, AB8500_DASLOTCONF4, mask, 11); + + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode DA-from-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active TX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + /* Setup TDM AD according to active RX-slots */ + slots_active = hweight32(rx_mask); + dev_dbg(dai->codec->dev, "%s: Slots, active, RX: %d\n", __func__, + slots_active); + switch (slots_active) { + case 0: + break; + case 1: + /* AD_OUT3 -> slot 0 & 1 */ + snd_soc_update_bits(codec, AB8500_ADSLOTSEL1, AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD); + break; + case 2: + /* AD_OUT3 -> slot 0, AD_OUT2 -> slot 1 */ + snd_soc_update_bits(codec, + AB8500_ADSLOTSEL1, + AB8500_MASK_ALL, + AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN | + AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD); + break; + case 8: + dev_dbg(dai->codec->dev, + "%s: In 8-channel mode AD-to-slot mapping is set manually.", + __func__); + break; + default: + dev_err(dai->codec->dev, + "%s: Unsupported number of active RX-slots (%d)!\n", + __func__, slots_active); + return -EINVAL; + } + + return 0; +} + +struct snd_soc_dai_driver ab8500_codec_dai[] = { + { + .name = "ab8500-codec-dai.0", + .id = 0, + .playback = { + .stream_name = "ab8500_0p", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + }, + { + .name = "ab8500-codec-dai.1", + .id = 1, + .capture = { + .stream_name = "ab8500_0c", + .channels_min = 1, + .channels_max = 8, + .rates = AB8500_SUPPORTED_RATE, + .formats = AB8500_SUPPORTED_FMT, + }, + .ops = (struct snd_soc_dai_ops[]) { + { + .set_tdm_slot = ab8500_codec_set_dai_tdm_slot, + .set_fmt = ab8500_codec_set_dai_fmt, + } + }, + .symmetric_rates = 1 + } +}; + +static int ab8500_codec_probe(struct snd_soc_codec *codec) +{ + struct device *dev = codec->dev; + struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(dev); + struct ab8500_platform_data *pdata; + struct filter_control *fc; + int status; + + dev_dbg(dev, "%s: Enter.\n", __func__); + + /* Setup AB8500 according to board-settings */ + pdata = (struct ab8500_platform_data *)dev_get_platdata(dev->parent); + status = ab8500_audio_setup_mics(codec, &pdata->codec->amics); + if (status < 0) { + pr_err("%s: Failed to setup mics (%d)!\n", __func__, status); + return status; + } + status = ab8500_audio_set_ear_cmv(codec, pdata->codec->ear_cmv); + if (status < 0) { + pr_err("%s: Failed to set earpiece CM-voltage (%d)!\n", + __func__, status); + return status; + } + + status = ab8500_audio_init_audioblock(codec); + if (status < 0) { + dev_err(dev, "%s: failed to init audio-block (%d)!\n", + __func__, status); + return status; + } + + /* Override HW-defaults */ + ab8500_codec_write_reg(codec, + AB8500_ANACONF5, + BIT(AB8500_ANACONF5_HSAUTOEN)); + ab8500_codec_write_reg(codec, + AB8500_SHORTCIRCONF, + BIT(AB8500_SHORTCIRCONF_HSZCDDIS)); + + /* Add filter controls */ + status = snd_soc_add_codec_controls(codec, ab8500_filter_controls, + ARRAY_SIZE(ab8500_filter_controls)); + if (status < 0) { + dev_err(dev, + "%s: failed to add ab8500 filter controls (%d).\n", + __func__, status); + return status; + } + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_FIR].private_value; + drvdata->anc_fir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_ANC_IIR].private_value; + drvdata->anc_iir_values = (long *)fc->value; + fc = (struct filter_control *) + &ab8500_filter_controls[AB8500_FILTER_SID_FIR].private_value; + drvdata->sid_fir_values = (long *)fc->value; + + (void)snd_soc_dapm_disable_pin(&codec->dapm, "ANC Configure Input"); + + mutex_init(&drvdata->anc_lock); + + return status; +} + +static struct snd_soc_codec_driver ab8500_codec_driver = { + .probe = ab8500_codec_probe, + .read = ab8500_codec_read_reg, + .write = ab8500_codec_write_reg, + .reg_word_size = sizeof(u8), + .controls = ab8500_ctrls, + .num_controls = ARRAY_SIZE(ab8500_ctrls), + .dapm_widgets = ab8500_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ab8500_dapm_widgets), + .dapm_routes = ab8500_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(ab8500_dapm_routes), +}; + +static int __devinit ab8500_codec_driver_probe(struct platform_device *pdev) +{ + int status; + struct ab8500_codec_drvdata *drvdata; + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(&pdev->dev, sizeof(struct ab8500_codec_drvdata), + GFP_KERNEL); + drvdata->sid_status = SID_UNCONFIGURED; + drvdata->anc_status = ANC_UNCONFIGURED; + dev_set_drvdata(&pdev->dev, drvdata); + + dev_dbg(&pdev->dev, "%s: Register codec.\n", __func__); + status = snd_soc_register_codec(&pdev->dev, &ab8500_codec_driver, + ab8500_codec_dai, + ARRAY_SIZE(ab8500_codec_dai)); + if (status < 0) + dev_err(&pdev->dev, + "%s: Error: Failed to register codec (%d).\n", + __func__, status); + + return status; +} + +static int __devexit ab8500_codec_driver_remove(struct platform_device *pdev) +{ + dev_info(&pdev->dev, "%s Enter.\n", __func__); + + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver ab8500_codec_platform_driver = { + .driver = { + .name = "ab8500-codec", + .owner = THIS_MODULE, + }, + .probe = ab8500_codec_driver_probe, + .remove = __devexit_p(ab8500_codec_driver_remove), + .suspend = NULL, + .resume = NULL, +}; +module_platform_driver(ab8500_codec_platform_driver); + +MODULE_LICENSE("GPLv2"); diff --git a/sound/soc/codecs/ab8500-codec.h b/sound/soc/codecs/ab8500-codec.h new file mode 100644 index 0000000..114f69a --- /dev/null +++ b/sound/soc/codecs/ab8500-codec.h @@ -0,0 +1,590 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja , + * Kristoffer Karlsson , + * Roger Nilsson , + * for ST-Ericsson. + * + * Based on the early work done by: + * Mikko J. Lehto , + * Mikko Sarmanne , + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CODEC_REGISTERS_H +#define AB8500_CODEC_REGISTERS_H + +#define AB8500_SUPPORTED_RATE (SNDRV_PCM_RATE_48000) +#define AB8500_SUPPORTED_FMT (SNDRV_PCM_FMTBIT_S16_LE) + +/* AB8500 audio bank (0x0d) register definitions */ + +#define AB8500_POWERUP 0x00 +#define AB8500_AUDSWRESET 0x01 +#define AB8500_ADPATHENA 0x02 +#define AB8500_DAPATHENA 0x03 +#define AB8500_ANACONF1 0x04 +#define AB8500_ANACONF2 0x05 +#define AB8500_DIGMICCONF 0x06 +#define AB8500_ANACONF3 0x07 +#define AB8500_ANACONF4 0x08 +#define AB8500_DAPATHCONF 0x09 +#define AB8500_MUTECONF 0x0A +#define AB8500_SHORTCIRCONF 0x0B +#define AB8500_ANACONF5 0x0C +#define AB8500_ENVCPCONF 0x0D +#define AB8500_SIGENVCONF 0x0E +#define AB8500_PWMGENCONF1 0x0F +#define AB8500_PWMGENCONF2 0x10 +#define AB8500_PWMGENCONF3 0x11 +#define AB8500_PWMGENCONF4 0x12 +#define AB8500_PWMGENCONF5 0x13 +#define AB8500_ANAGAIN1 0x14 +#define AB8500_ANAGAIN2 0x15 +#define AB8500_ANAGAIN3 0x16 +#define AB8500_ANAGAIN4 0x17 +#define AB8500_DIGLINHSLGAIN 0x18 +#define AB8500_DIGLINHSRGAIN 0x19 +#define AB8500_ADFILTCONF 0x1A +#define AB8500_DIGIFCONF1 0x1B +#define AB8500_DIGIFCONF2 0x1C +#define AB8500_DIGIFCONF3 0x1D +#define AB8500_DIGIFCONF4 0x1E +#define AB8500_ADSLOTSEL1 0x1F +#define AB8500_ADSLOTSEL2 0x20 +#define AB8500_ADSLOTSEL3 0x21 +#define AB8500_ADSLOTSEL4 0x22 +#define AB8500_ADSLOTSEL5 0x23 +#define AB8500_ADSLOTSEL6 0x24 +#define AB8500_ADSLOTSEL7 0x25 +#define AB8500_ADSLOTSEL8 0x26 +#define AB8500_ADSLOTSEL9 0x27 +#define AB8500_ADSLOTSEL10 0x28 +#define AB8500_ADSLOTSEL11 0x29 +#define AB8500_ADSLOTSEL12 0x2A +#define AB8500_ADSLOTSEL13 0x2B +#define AB8500_ADSLOTSEL14 0x2C +#define AB8500_ADSLOTSEL15 0x2D +#define AB8500_ADSLOTSEL16 0x2E +#define AB8500_ADSLOTHIZCTRL1 0x2F +#define AB8500_ADSLOTHIZCTRL2 0x30 +#define AB8500_ADSLOTHIZCTRL3 0x31 +#define AB8500_ADSLOTHIZCTRL4 0x32 +#define AB8500_DASLOTCONF1 0x33 +#define AB8500_DASLOTCONF2 0x34 +#define AB8500_DASLOTCONF3 0x35 +#define AB8500_DASLOTCONF4 0x36 +#define AB8500_DASLOTCONF5 0x37 +#define AB8500_DASLOTCONF6 0x38 +#define AB8500_DASLOTCONF7 0x39 +#define AB8500_DASLOTCONF8 0x3A +#define AB8500_CLASSDCONF1 0x3B +#define AB8500_CLASSDCONF2 0x3C +#define AB8500_CLASSDCONF3 0x3D +#define AB8500_DMICFILTCONF 0x3E +#define AB8500_DIGMULTCONF1 0x3F +#define AB8500_DIGMULTCONF2 0x40 +#define AB8500_ADDIGGAIN1 0x41 +#define AB8500_ADDIGGAIN2 0x42 +#define AB8500_ADDIGGAIN3 0x43 +#define AB8500_ADDIGGAIN4 0x44 +#define AB8500_ADDIGGAIN5 0x45 +#define AB8500_ADDIGGAIN6 0x46 +#define AB8500_DADIGGAIN1 0x47 +#define AB8500_DADIGGAIN2 0x48 +#define AB8500_DADIGGAIN3 0x49 +#define AB8500_DADIGGAIN4 0x4A +#define AB8500_DADIGGAIN5 0x4B +#define AB8500_DADIGGAIN6 0x4C +#define AB8500_ADDIGLOOPGAIN1 0x4D +#define AB8500_ADDIGLOOPGAIN2 0x4E +#define AB8500_HSLEARDIGGAIN 0x4F +#define AB8500_HSRDIGGAIN 0x50 +#define AB8500_SIDFIRGAIN1 0x51 +#define AB8500_SIDFIRGAIN2 0x52 +#define AB8500_ANCCONF1 0x53 +#define AB8500_ANCCONF2 0x54 +#define AB8500_ANCCONF3 0x55 +#define AB8500_ANCCONF4 0x56 +#define AB8500_ANCCONF5 0x57 +#define AB8500_ANCCONF6 0x58 +#define AB8500_ANCCONF7 0x59 +#define AB8500_ANCCONF8 0x5A +#define AB8500_ANCCONF9 0x5B +#define AB8500_ANCCONF10 0x5C +#define AB8500_ANCCONF11 0x5D +#define AB8500_ANCCONF12 0x5E +#define AB8500_ANCCONF13 0x5F +#define AB8500_ANCCONF14 0x60 +#define AB8500_SIDFIRADR 0x61 +#define AB8500_SIDFIRCOEF1 0x62 +#define AB8500_SIDFIRCOEF2 0x63 +#define AB8500_SIDFIRCONF 0x64 +#define AB8500_AUDINTMASK1 0x65 +#define AB8500_AUDINTSOURCE1 0x66 +#define AB8500_AUDINTMASK2 0x67 +#define AB8500_AUDINTSOURCE2 0x68 +#define AB8500_FIFOCONF1 0x69 +#define AB8500_FIFOCONF2 0x6A +#define AB8500_FIFOCONF3 0x6B +#define AB8500_FIFOCONF4 0x6C +#define AB8500_FIFOCONF5 0x6D +#define AB8500_FIFOCONF6 0x6E +#define AB8500_AUDREV 0x6F + +#define AB8500_FIRST_REG AB8500_POWERUP +#define AB8500_LAST_REG AB8500_AUDREV +#define AB8500_CACHEREGNUM (AB8500_LAST_REG + 1) + +#define AB8500_MASK_ALL 0xFF +#define AB8500_MASK_NONE 0x00 + +/* AB8500_POWERUP */ +#define AB8500_POWERUP_POWERUP 7 +#define AB8500_POWERUP_ENANA 3 + +/* AB8500_AUDSWRESET */ +#define AB8500_AUDSWRESET_SWRESET 7 + +/* AB8500_ADPATHENA */ +#define AB8500_ADPATHENA_ENAD12 7 +#define AB8500_ADPATHENA_ENAD34 5 +#define AB8500_ADPATHENA_ENAD5768 3 + +/* AB8500_DAPATHENA */ +#define AB8500_DAPATHENA_ENDA1 7 +#define AB8500_DAPATHENA_ENDA2 6 +#define AB8500_DAPATHENA_ENDA3 5 +#define AB8500_DAPATHENA_ENDA4 4 +#define AB8500_DAPATHENA_ENDA5 3 +#define AB8500_DAPATHENA_ENDA6 2 + +/* AB8500_ANACONF1 */ +#define AB8500_ANACONF1_HSLOWPOW 7 +#define AB8500_ANACONF1_DACLOWPOW1 6 +#define AB8500_ANACONF1_DACLOWPOW0 5 +#define AB8500_ANACONF1_EARDACLOWPOW 4 +#define AB8500_ANACONF1_EARSELCM 2 +#define AB8500_ANACONF1_HSHPEN 1 +#define AB8500_ANACONF1_EARDRVLOWPOW 0 + +/* AB8500_ANACONF2 */ +#define AB8500_ANACONF2_ENMIC1 7 +#define AB8500_ANACONF2_ENMIC2 6 +#define AB8500_ANACONF2_ENLINL 5 +#define AB8500_ANACONF2_ENLINR 4 +#define AB8500_ANACONF2_MUTMIC1 3 +#define AB8500_ANACONF2_MUTMIC2 2 +#define AB8500_ANACONF2_MUTLINL 1 +#define AB8500_ANACONF2_MUTLINR 0 + +/* AB8500_DIGMICCONF */ +#define AB8500_DIGMICCONF_ENDMIC1 7 +#define AB8500_DIGMICCONF_ENDMIC2 6 +#define AB8500_DIGMICCONF_ENDMIC3 5 +#define AB8500_DIGMICCONF_ENDMIC4 4 +#define AB8500_DIGMICCONF_ENDMIC5 3 +#define AB8500_DIGMICCONF_ENDMIC6 2 +#define AB8500_DIGMICCONF_HSFADSPEED 0 + +/* AB8500_ANACONF3 */ +#define AB8500_ANACONF3_MIC1SEL 7 +#define AB8500_ANACONF3_LINRSEL 6 +#define AB8500_ANACONF3_ENDRVHSL 5 +#define AB8500_ANACONF3_ENDRVHSR 4 +#define AB8500_ANACONF3_ENADCMIC 2 +#define AB8500_ANACONF3_ENADCLINL 1 +#define AB8500_ANACONF3_ENADCLINR 0 + +/* AB8500_ANACONF4 */ +#define AB8500_ANACONF4_DISPDVSS 7 +#define AB8500_ANACONF4_ENEAR 6 +#define AB8500_ANACONF4_ENHSL 5 +#define AB8500_ANACONF4_ENHSR 4 +#define AB8500_ANACONF4_ENHFL 3 +#define AB8500_ANACONF4_ENHFR 2 +#define AB8500_ANACONF4_ENVIB1 1 +#define AB8500_ANACONF4_ENVIB2 0 + +/* AB8500_DAPATHCONF */ +#define AB8500_DAPATHCONF_ENDACEAR 6 +#define AB8500_DAPATHCONF_ENDACHSL 5 +#define AB8500_DAPATHCONF_ENDACHSR 4 +#define AB8500_DAPATHCONF_ENDACHFL 3 +#define AB8500_DAPATHCONF_ENDACHFR 2 +#define AB8500_DAPATHCONF_ENDACVIB1 1 +#define AB8500_DAPATHCONF_ENDACVIB2 0 + +/* AB8500_MUTECONF */ +#define AB8500_MUTECONF_MUTEAR 6 +#define AB8500_MUTECONF_MUTHSL 5 +#define AB8500_MUTECONF_MUTHSR 4 +#define AB8500_MUTECONF_MUTDACEAR 2 +#define AB8500_MUTECONF_MUTDACHSL 1 +#define AB8500_MUTECONF_MUTDACHSR 0 + +/* AB8500_SHORTCIRCONF */ +#define AB8500_SHORTCIRCONF_ENSHORTPWD 7 +#define AB8500_SHORTCIRCONF_EARSHORTDIS 6 +#define AB8500_SHORTCIRCONF_HSSHORTDIS 5 +#define AB8500_SHORTCIRCONF_HSPULLDEN 4 +#define AB8500_SHORTCIRCONF_HSOSCEN 2 +#define AB8500_SHORTCIRCONF_HSFADDIS 1 +#define AB8500_SHORTCIRCONF_HSZCDDIS 0 +/* Zero cross should be disabled */ + +/* AB8500_ANACONF5 */ +#define AB8500_ANACONF5_ENCPHS 7 +#define AB8500_ANACONF5_HSLDACTOLOL 5 +#define AB8500_ANACONF5_HSRDACTOLOR 4 +#define AB8500_ANACONF5_ENLOL 3 +#define AB8500_ANACONF5_ENLOR 2 +#define AB8500_ANACONF5_HSAUTOEN 0 + +/* AB8500_ENVCPCONF */ +#define AB8500_ENVCPCONF_ENVDETHTHRE 4 +#define AB8500_ENVCPCONF_ENVDETLTHRE 0 +#define AB8500_ENVCPCONF_ENVDETHTHRE_MAX 0x0F +#define AB8500_ENVCPCONF_ENVDETLTHRE_MAX 0x0F + +/* AB8500_SIGENVCONF */ +#define AB8500_SIGENVCONF_CPLVEN 5 +#define AB8500_SIGENVCONF_ENVDETCPEN 4 +#define AB8500_SIGENVCONF_ENVDETTIME 0 +#define AB8500_SIGENVCONF_ENVDETTIME_MAX 0x0F + +/* AB8500_PWMGENCONF1 */ +#define AB8500_PWMGENCONF1_PWMTOVIB1 7 +#define AB8500_PWMGENCONF1_PWMTOVIB2 6 +#define AB8500_PWMGENCONF1_PWM1CTRL 5 +#define AB8500_PWMGENCONF1_PWM2CTRL 4 +#define AB8500_PWMGENCONF1_PWM1NCTRL 3 +#define AB8500_PWMGENCONF1_PWM1PCTRL 2 +#define AB8500_PWMGENCONF1_PWM2NCTRL 1 +#define AB8500_PWMGENCONF1_PWM2PCTRL 0 + +/* AB8500_PWMGENCONF2 */ +/* AB8500_PWMGENCONF3 */ +/* AB8500_PWMGENCONF4 */ +/* AB8500_PWMGENCONF5 */ +#define AB8500_PWMGENCONFX_PWMVIBXPOL 7 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC 0 +#define AB8500_PWMGENCONFX_PWMVIBXDUTCYC_MAX 0x64 + +/* AB8500_ANAGAIN1 */ +/* AB8500_ANAGAIN2 */ +#define AB8500_ANAGAINX_ENSEMICX 7 +#define AB8500_ANAGAINX_LOWPOWMICX 6 +#define AB8500_ANAGAINX_MICXGAIN 0 +#define AB8500_ANAGAINX_MICXGAIN_MAX 0x1F + +/* AB8500_ANAGAIN3 */ +#define AB8500_ANAGAIN3_HSLGAIN 4 +#define AB8500_ANAGAIN3_HSRGAIN 0 +#define AB8500_ANAGAIN3_HSXGAIN_MAX 0x0F + +/* AB8500_ANAGAIN4 */ +#define AB8500_ANAGAIN4_LINLGAIN 4 +#define AB8500_ANAGAIN4_LINRGAIN 0 +#define AB8500_ANAGAIN4_LINXGAIN_MAX 0x0F + +/* AB8500_DIGLINHSLGAIN */ +/* AB8500_DIGLINHSRGAIN */ +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN 0 +#define AB8500_DIGLINHSXGAIN_LINTOHSXGAIN_MAX 0x13 + +/* AB8500_ADFILTCONF */ +#define AB8500_ADFILTCONF_AD1NH 7 +#define AB8500_ADFILTCONF_AD2NH 6 +#define AB8500_ADFILTCONF_AD3NH 5 +#define AB8500_ADFILTCONF_AD4NH 4 +#define AB8500_ADFILTCONF_AD1VOICE 3 +#define AB8500_ADFILTCONF_AD2VOICE 2 +#define AB8500_ADFILTCONF_AD3VOICE 1 +#define AB8500_ADFILTCONF_AD4VOICE 0 + +/* AB8500_DIGIFCONF1 */ +#define AB8500_DIGIFCONF1_ENMASTGEN 7 +#define AB8500_DIGIFCONF1_IF1BITCLKOS1 6 +#define AB8500_DIGIFCONF1_IF1BITCLKOS0 5 +#define AB8500_DIGIFCONF1_ENFSBITCLK1 4 +#define AB8500_DIGIFCONF1_IF0BITCLKOS1 2 +#define AB8500_DIGIFCONF1_IF0BITCLKOS0 1 +#define AB8500_DIGIFCONF1_ENFSBITCLK0 0 + +/* AB8500_DIGIFCONF2 */ +#define AB8500_DIGIFCONF2_FSYNC0P 6 +#define AB8500_DIGIFCONF2_BITCLK0P 5 +#define AB8500_DIGIFCONF2_IF0DEL 4 +#define AB8500_DIGIFCONF2_IF0FORMAT1 3 +#define AB8500_DIGIFCONF2_IF0FORMAT0 2 +#define AB8500_DIGIFCONF2_IF0WL1 1 +#define AB8500_DIGIFCONF2_IF0WL0 0 + +/* AB8500_DIGIFCONF3 */ +#define AB8500_DIGIFCONF3_IF0DATOIF1AD 7 +#define AB8500_DIGIFCONF3_IF0CLKTOIF1CLK 6 +#define AB8500_DIGIFCONF3_IF1MASTER 5 +#define AB8500_DIGIFCONF3_IF1DATOIF0AD 3 +#define AB8500_DIGIFCONF3_IF1CLKTOIF0CLK 2 +#define AB8500_DIGIFCONF3_IF0MASTER 1 +#define AB8500_DIGIFCONF3_IF0BFIFOEN 0 + +/* AB8500_DIGIFCONF4 */ +#define AB8500_DIGIFCONF4_FSYNC1P 6 +#define AB8500_DIGIFCONF4_BITCLK1P 5 +#define AB8500_DIGIFCONF4_IF1DEL 4 +#define AB8500_DIGIFCONF4_IF1FORMAT1 3 +#define AB8500_DIGIFCONF4_IF1FORMAT0 2 +#define AB8500_DIGIFCONF4_IF1WL1 1 +#define AB8500_DIGIFCONF4_IF1WL0 0 + +/* AB8500_ADSLOTSELX */ +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_ODD 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_ODD 0x01 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_ODD 0x02 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_ODD 0x03 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_ODD 0x04 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_ODD 0x05 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_ODD 0x06 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_ODD 0x07 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_ODD 0x08 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_ODD 0x0F +#define AB8500_ADSLOTSELX_AD_OUT1_TO_SLOT_EVEN 0x00 +#define AB8500_ADSLOTSELX_AD_OUT2_TO_SLOT_EVEN 0x10 +#define AB8500_ADSLOTSELX_AD_OUT3_TO_SLOT_EVEN 0x20 +#define AB8500_ADSLOTSELX_AD_OUT4_TO_SLOT_EVEN 0x30 +#define AB8500_ADSLOTSELX_AD_OUT5_TO_SLOT_EVEN 0x40 +#define AB8500_ADSLOTSELX_AD_OUT6_TO_SLOT_EVEN 0x50 +#define AB8500_ADSLOTSELX_AD_OUT7_TO_SLOT_EVEN 0x60 +#define AB8500_ADSLOTSELX_AD_OUT8_TO_SLOT_EVEN 0x70 +#define AB8500_ADSLOTSELX_ZEROES_TO_SLOT_EVEN 0x80 +#define AB8500_ADSLOTSELX_TRISTATE_TO_SLOT_EVEN 0xF0 +#define AB8500_ADSLOTSELX_EVEN_SHIFT 0 +#define AB8500_ADSLOTSELX_ODD_SHIFT 4 + +/* AB8500_ADSLOTHIZCTRL1 */ +/* AB8500_ADSLOTHIZCTRL2 */ +/* AB8500_ADSLOTHIZCTRL3 */ +/* AB8500_ADSLOTHIZCTRL4 */ +/* AB8500_DASLOTCONF1 */ +#define AB8500_DASLOTCONF1_DA12VOICE 7 +#define AB8500_DASLOTCONF1_SWAPDA12_34 6 +#define AB8500_DASLOTCONF1_DAI7TOADO1 5 + +/* AB8500_DASLOTCONF2 */ +#define AB8500_DASLOTCONF2_DAI8TOADO2 5 + +/* AB8500_DASLOTCONF3 */ +#define AB8500_DASLOTCONF3_DA34VOICE 7 +#define AB8500_DASLOTCONF3_DAI7TOADO3 5 + +/* AB8500_DASLOTCONF4 */ +#define AB8500_DASLOTCONF4_DAI8TOADO4 5 + +/* AB8500_DASLOTCONF5 */ +#define AB8500_DASLOTCONF5_DA56VOICE 7 +#define AB8500_DASLOTCONF5_DAI7TOADO5 5 + +/* AB8500_DASLOTCONF6 */ +#define AB8500_DASLOTCONF6_DAI8TOADO6 5 + +/* AB8500_DASLOTCONF7 */ +#define AB8500_DASLOTCONF7_DAI8TOADO7 5 + +/* AB8500_DASLOTCONF8 */ +#define AB8500_DASLOTCONF8_DAI7TOADO8 5 + +#define AB8500_DASLOTCONFX_SLTODAX_SHIFT 0 +#define AB8500_DASLOTCONFX_SLTODAX_MASK 0x1F + +/* AB8500_CLASSDCONF1 */ +#define AB8500_CLASSDCONF1_PARLHF 7 +#define AB8500_CLASSDCONF1_PARLVIB 6 +#define AB8500_CLASSDCONF1_VIB1SWAPEN 3 +#define AB8500_CLASSDCONF1_VIB2SWAPEN 2 +#define AB8500_CLASSDCONF1_HFLSWAPEN 1 +#define AB8500_CLASSDCONF1_HFRSWAPEN 0 + +/* AB8500_CLASSDCONF2 */ +#define AB8500_CLASSDCONF2_FIRBYP3 7 +#define AB8500_CLASSDCONF2_FIRBYP2 6 +#define AB8500_CLASSDCONF2_FIRBYP1 5 +#define AB8500_CLASSDCONF2_FIRBYP0 4 +#define AB8500_CLASSDCONF2_HIGHVOLEN3 3 +#define AB8500_CLASSDCONF2_HIGHVOLEN2 2 +#define AB8500_CLASSDCONF2_HIGHVOLEN1 1 +#define AB8500_CLASSDCONF2_HIGHVOLEN0 0 + +/* AB8500_CLASSDCONF3 */ +#define AB8500_CLASSDCONF3_DITHHPGAIN 4 +#define AB8500_CLASSDCONF3_DITHHPGAIN_MAX 0x0A +#define AB8500_CLASSDCONF3_DITHWGAIN 0 +#define AB8500_CLASSDCONF3_DITHWGAIN_MAX 0x0A + +/* AB8500_DMICFILTCONF */ +#define AB8500_DMICFILTCONF_ANCINSEL 7 +#define AB8500_DMICFILTCONF_DA3TOEAR 6 +#define AB8500_DMICFILTCONF_DMIC1SINC3 5 +#define AB8500_DMICFILTCONF_DMIC2SINC3 4 +#define AB8500_DMICFILTCONF_DMIC3SINC3 3 +#define AB8500_DMICFILTCONF_DMIC4SINC3 2 +#define AB8500_DMICFILTCONF_DMIC5SINC3 1 +#define AB8500_DMICFILTCONF_DMIC6SINC3 0 + +/* AB8500_DIGMULTCONF1 */ +#define AB8500_DIGMULTCONF1_DATOHSLEN 7 +#define AB8500_DIGMULTCONF1_DATOHSREN 6 +#define AB8500_DIGMULTCONF1_AD1SEL 5 +#define AB8500_DIGMULTCONF1_AD2SEL 4 +#define AB8500_DIGMULTCONF1_AD3SEL 3 +#define AB8500_DIGMULTCONF1_AD5SEL 2 +#define AB8500_DIGMULTCONF1_AD6SEL 1 +#define AB8500_DIGMULTCONF1_ANCSEL 0 + +/* AB8500_DIGMULTCONF2 */ +#define AB8500_DIGMULTCONF2_DATOHFREN 7 +#define AB8500_DIGMULTCONF2_DATOHFLEN 6 +#define AB8500_DIGMULTCONF2_HFRSEL 5 +#define AB8500_DIGMULTCONF2_HFLSEL 4 +#define AB8500_DIGMULTCONF2_FIRSID1SEL 2 +#define AB8500_DIGMULTCONF2_FIRSID2SEL 0 + +/* AB8500_ADDIGGAIN1 */ +/* AB8500_ADDIGGAIN2 */ +/* AB8500_ADDIGGAIN3 */ +/* AB8500_ADDIGGAIN4 */ +/* AB8500_ADDIGGAIN5 */ +/* AB8500_ADDIGGAIN6 */ +#define AB8500_ADDIGGAINX_FADEDISADX 6 +#define AB8500_ADDIGGAINX_ADXGAIN_MAX 0x3F + +/* AB8500_DADIGGAIN1 */ +/* AB8500_DADIGGAIN2 */ +/* AB8500_DADIGGAIN3 */ +/* AB8500_DADIGGAIN4 */ +/* AB8500_DADIGGAIN5 */ +/* AB8500_DADIGGAIN6 */ +#define AB8500_DADIGGAINX_FADEDISDAX 6 +#define AB8500_DADIGGAINX_DAXGAIN_MAX 0x3F + +/* AB8500_ADDIGLOOPGAIN1 */ +/* AB8500_ADDIGLOOPGAIN2 */ +#define AB8500_ADDIGLOOPGAINX_FADEDISADXL 6 +#define AB8500_ADDIGLOOPGAINX_ADXLBGAIN_MAX 0x3F + +/* AB8500_HSLEARDIGGAIN */ +#define AB8500_HSLEARDIGGAIN_HSSINC1 7 +#define AB8500_HSLEARDIGGAIN_FADEDISHSL 4 +#define AB8500_HSLEARDIGGAIN_HSLDGAIN_MAX 0x09 + +/* AB8500_HSRDIGGAIN */ +#define AB8500_HSRDIGGAIN_FADESPEED 6 +#define AB8500_HSRDIGGAIN_FADEDISHSR 4 +#define AB8500_HSRDIGGAIN_HSRDGAIN_MAX 0x09 + +/* AB8500_SIDFIRGAIN1 */ +/* AB8500_SIDFIRGAIN2 */ +#define AB8500_SIDFIRGAINX_FIRSIDXGAIN_MAX 0x1F + +/* AB8500_ANCCONF1 */ +#define AB8500_ANCCONF1_ANCIIRUPDATE 3 +#define AB8500_ANCCONF1_ENANC 2 +#define AB8500_ANCCONF1_ANCIIRINIT 1 +#define AB8500_ANCCONF1_ANCFIRUPDATE 0 + +/* AB8500_ANCCONF2 */ +#define AB8500_ANCCONF2_SHIFT 5 +#define AB8500_ANCCONF2_MIN -0x10 +#define AB8500_ANCCONF2_MAX 0xF + +/* AB8500_ANCCONF3 */ +#define AB8500_ANCCONF3_SHIFT 5 +#define AB8500_ANCCONF3_MIN -0x10 +#define AB8500_ANCCONF3_MAX 0xF + +/* AB8500_ANCCONF4 */ +#define AB8500_ANCCONF4_SHIFT 5 +#define AB8500_ANCCONF4_MIN -0x10 +#define AB8500_ANCCONF4_MAX 0xF + +/* AB8500_ANC_FIR_COEFFS */ +#define AB8500_ANC_FIR_COEFF_MIN -0x8000 +#define AB8500_ANC_FIR_COEFF_MAX 0x7FFF +#define AB8500_ANC_FIR_COEFFS 15 + +/* AB8500_ANC_IIR_COEFFS */ +#define AB8500_ANC_IIR_COEFF_MIN -0x800000 +#define AB8500_ANC_IIR_COEFF_MAX 0x7FFFFF +#define AB8500_ANC_IIR_COEFFS 24 +/* AB8500_ANC_WARP_DELAY */ +#define AB8500_ANC_WARP_DELAY_SHIFT 16 +#define AB8500_ANC_WARP_DELAY_MIN 0x0000 +#define AB8500_ANC_WARP_DELAY_MAX 0xFFFF + +/* AB8500_ANCCONF11 */ +/* AB8500_ANCCONF12 */ +/* AB8500_ANCCONF13 */ +/* AB8500_ANCCONF14 */ + +/* AB8500_SIDFIRADR */ +#define AB8500_SIDFIRADR_FIRSIDSET 7 +#define AB8500_SIDFIRADR_ADDRESS_SHIFT 0 +#define AB8500_SIDFIRADR_ADDRESS_MAX 0x7F + +/* AB8500_SIDFIRCOEF1 */ +/* AB8500_SIDFIRCOEF2 */ +#define AB8500_SID_FIR_COEFF_MIN 0 +#define AB8500_SID_FIR_COEFF_MAX 0xFFFF +#define AB8500_SID_FIR_COEFFS 128 + +/* AB8500_SIDFIRCONF */ +#define AB8500_SIDFIRCONF_ENFIRSIDS 2 +#define AB8500_SIDFIRCONF_FIRSIDSTOIF1 1 +#define AB8500_SIDFIRCONF_FIRSIDBUSY 0 + +/* AB8500_AUDINTMASK1 */ +/* AB8500_AUDINTSOURCE1 */ +/* AB8500_AUDINTMASK2 */ +/* AB8500_AUDINTSOURCE2 */ + +/* AB8500_FIFOCONF1 */ +#define AB8500_FIFOCONF1_BFIFOMASK 0x80 +#define AB8500_FIFOCONF1_BFIFO19M2 0x40 +#define AB8500_FIFOCONF1_BFIFOINT_SHIFT 0 +#define AB8500_FIFOCONF1_BFIFOINT_MAX 0x3F + +/* AB8500_FIFOCONF2 */ +#define AB8500_FIFOCONF2_BFIFOTX_SHIFT 0 +#define AB8500_FIFOCONF2_BFIFOTX_MAX 0xFF + +/* AB8500_FIFOCONF3 */ +#define AB8500_FIFOCONF3_BFIFOEXSL_SHIFT 5 +#define AB8500_FIFOCONF3_BFIFOEXSL_MAX 0x5 +#define AB8500_FIFOCONF3_PREBITCLK0_SHIFT 2 +#define AB8500_FIFOCONF3_PREBITCLK0_MAX 0x7 +#define AB8500_FIFOCONF3_BFIFOMAST_SHIFT 1 +#define AB8500_FIFOCONF3_BFIFORUN_SHIFT 0 + +/* AB8500_FIFOCONF4 */ +#define AB8500_FIFOCONF4_BFIFOFRAMSW_SHIFT 0 +#define AB8500_FIFOCONF4_BFIFOFRAMSW_MAX 0xFF + +/* AB8500_FIFOCONF5 */ +#define AB8500_FIFOCONF5_BFIFOWAKEUP_SHIFT 0 +#define AB8500_FIFOCONF5_BFIFOWAKEUP_MAX 0xFF + +/* AB8500_FIFOCONF6 */ +#define AB8500_FIFOCONF6_BFIFOSAMPLE_SHIFT 0 +#define AB8500_FIFOCONF6_BFIFOSAMPLE_MAX 0xFF + +/* AB8500_AUDREV */ + +#endif -- cgit v0.10.2 From 66c2b7377a7cf22c48ebba7fdff5340ab492b7bc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:09:46 +0800 Subject: ASoC: wm8996: Remove write sequencer registers from the defaults table They aren't marked as readable and the feature is never used so they'll never get referenced. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index efc4e9d..f24989f 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -296,184 +296,6 @@ static struct reg_default wm8996_reg[] = { { WM8996_RIGHT_PDM_SPEAKER, 0x1 }, { WM8996_PDM_SPEAKER_MUTE_SEQUENCE, 0x69 }, { WM8996_PDM_SPEAKER_VOLUME, 0x66 }, - { WM8996_WRITE_SEQUENCER_0, 0x1 }, - { WM8996_WRITE_SEQUENCER_1, 0x1 }, - { WM8996_WRITE_SEQUENCER_3, 0x6 }, - { WM8996_WRITE_SEQUENCER_4, 0x40 }, - { WM8996_WRITE_SEQUENCER_5, 0x1 }, - { WM8996_WRITE_SEQUENCER_6, 0xf }, - { WM8996_WRITE_SEQUENCER_7, 0x6 }, - { WM8996_WRITE_SEQUENCER_8, 0x1 }, - { WM8996_WRITE_SEQUENCER_9, 0x3 }, - { WM8996_WRITE_SEQUENCER_10, 0x104 }, - { WM8996_WRITE_SEQUENCER_12, 0x60 }, - { WM8996_WRITE_SEQUENCER_13, 0x11 }, - { WM8996_WRITE_SEQUENCER_14, 0x401 }, - { WM8996_WRITE_SEQUENCER_16, 0x50 }, - { WM8996_WRITE_SEQUENCER_17, 0x3 }, - { WM8996_WRITE_SEQUENCER_18, 0x100 }, - { WM8996_WRITE_SEQUENCER_20, 0x51 }, - { WM8996_WRITE_SEQUENCER_21, 0x3 }, - { WM8996_WRITE_SEQUENCER_22, 0x104 }, - { WM8996_WRITE_SEQUENCER_23, 0xa }, - { WM8996_WRITE_SEQUENCER_24, 0x60 }, - { WM8996_WRITE_SEQUENCER_25, 0x3b }, - { WM8996_WRITE_SEQUENCER_26, 0x502 }, - { WM8996_WRITE_SEQUENCER_27, 0x100 }, - { WM8996_WRITE_SEQUENCER_28, 0x2fff }, - { WM8996_WRITE_SEQUENCER_32, 0x2fff }, - { WM8996_WRITE_SEQUENCER_36, 0x2fff }, - { WM8996_WRITE_SEQUENCER_40, 0x2fff }, - { WM8996_WRITE_SEQUENCER_44, 0x2fff }, - { WM8996_WRITE_SEQUENCER_48, 0x2fff }, - { WM8996_WRITE_SEQUENCER_52, 0x2fff }, - { WM8996_WRITE_SEQUENCER_56, 0x2fff }, - { WM8996_WRITE_SEQUENCER_60, 0x2fff }, - { WM8996_WRITE_SEQUENCER_64, 0x1 }, - { WM8996_WRITE_SEQUENCER_65, 0x1 }, - { WM8996_WRITE_SEQUENCER_67, 0x6 }, - { WM8996_WRITE_SEQUENCER_68, 0x40 }, - { WM8996_WRITE_SEQUENCER_69, 0x1 }, - { WM8996_WRITE_SEQUENCER_70, 0xf }, - { WM8996_WRITE_SEQUENCER_71, 0x6 }, - { WM8996_WRITE_SEQUENCER_72, 0x1 }, - { WM8996_WRITE_SEQUENCER_73, 0x3 }, - { WM8996_WRITE_SEQUENCER_74, 0x104 }, - { WM8996_WRITE_SEQUENCER_76, 0x60 }, - { WM8996_WRITE_SEQUENCER_77, 0x11 }, - { WM8996_WRITE_SEQUENCER_78, 0x401 }, - { WM8996_WRITE_SEQUENCER_80, 0x50 }, - { WM8996_WRITE_SEQUENCER_81, 0x3 }, - { WM8996_WRITE_SEQUENCER_82, 0x100 }, - { WM8996_WRITE_SEQUENCER_84, 0x60 }, - { WM8996_WRITE_SEQUENCER_85, 0x3b }, - { WM8996_WRITE_SEQUENCER_86, 0x502 }, - { WM8996_WRITE_SEQUENCER_87, 0x100 }, - { WM8996_WRITE_SEQUENCER_88, 0x2fff }, - { WM8996_WRITE_SEQUENCER_92, 0x2fff }, - { WM8996_WRITE_SEQUENCER_96, 0x2fff }, - { WM8996_WRITE_SEQUENCER_100, 0x2fff }, - { WM8996_WRITE_SEQUENCER_104, 0x2fff }, - { WM8996_WRITE_SEQUENCER_108, 0x2fff }, - { WM8996_WRITE_SEQUENCER_112, 0x2fff }, - { WM8996_WRITE_SEQUENCER_116, 0x2fff }, - { WM8996_WRITE_SEQUENCER_120, 0x2fff }, - { WM8996_WRITE_SEQUENCER_124, 0x2fff }, - { WM8996_WRITE_SEQUENCER_128, 0x1 }, - { WM8996_WRITE_SEQUENCER_129, 0x1 }, - { WM8996_WRITE_SEQUENCER_131, 0x6 }, - { WM8996_WRITE_SEQUENCER_132, 0x40 }, - { WM8996_WRITE_SEQUENCER_133, 0x1 }, - 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{ WM8996_WRITE_SEQUENCER_388, 0x61 }, - { WM8996_WRITE_SEQUENCER_390, 0x601 }, - { WM8996_WRITE_SEQUENCER_392, 0x50 }, - { WM8996_WRITE_SEQUENCER_394, 0x300 }, - { WM8996_WRITE_SEQUENCER_396, 0x1 }, - { WM8996_WRITE_SEQUENCER_398, 0x304 }, - { WM8996_WRITE_SEQUENCER_400, 0x40 }, - { WM8996_WRITE_SEQUENCER_402, 0xf }, - { WM8996_WRITE_SEQUENCER_404, 0x1 }, - { WM8996_WRITE_SEQUENCER_407, 0x100 }, }; static const DECLARE_TLV_DB_SCALE(inpga_tlv, 0, 100, 0); -- cgit v0.10.2 From af691fb62c626fe374955ab306092b09f672e27d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:20:48 +0800 Subject: ASoC: wm8996: Convert to devm_regmap_init_i2c() Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index f24989f..a6b5cff 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3000,7 +3000,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, msleep(5); } - wm8996->regmap = regmap_init_i2c(i2c, &wm8996_regmap); + wm8996->regmap = devm_regmap_init_i2c(i2c, &wm8996_regmap); if (IS_ERR(wm8996->regmap)) { ret = PTR_ERR(wm8996->regmap); dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); @@ -3049,7 +3049,6 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, err_gpiolib: wm8996_free_gpio(wm8996); err_regmap: - regmap_exit(wm8996->regmap); err_enable: if (wm8996->pdata.ldo_ena > 0) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); @@ -3068,7 +3067,6 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); - regmap_exit(wm8996->regmap); if (wm8996->pdata.ldo_ena > 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); -- cgit v0.10.2 From 48e278746070b5fc62ec3da2e65f7cd511f6bbf4 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Mon, 11 Jun 2012 13:15:27 +0100 Subject: ASoC: codecs: Add DA732x codec driver This patch adds support for Dialog DA732x audio codecs. Signed-off-by: Michal Hajduk Signed-off-by: Adam Thomson Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f63776d4..43f5240 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI select SND_SOC_CX20442 select SND_SOC_DA7210 if I2C + select SND_SOC_DA732X if I2C select SND_SOC_DFBMCS320 select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC @@ -224,6 +225,9 @@ config SND_SOC_L3 config SND_SOC_DA7210 tristate +config SND_SOC_DA732X + tristate + config SND_SOC_DFBMCS320 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fc93b4b..3d30654 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o +snd-soc-da732x-objs := da732x.o snd-soc-dfbmcs320-objs := dfbmcs320.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o @@ -135,6 +136,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o +obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c new file mode 100644 index 0000000..04af369 --- /dev/null +++ b/sound/soc/codecs/da732x.c @@ -0,0 +1,1627 @@ +/* + * da732x.c --- Dialog DA732X ALSA SoC Audio Driver + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "da732x.h" +#include "da732x_reg.h" + + +struct da732x_priv { + struct regmap *regmap; + struct snd_soc_codec *codec; + + unsigned int sysclk; + bool pll_en; +}; + +/* + * da732x register cache - default settings + */ +static struct reg_default da732x_reg_cache[] = { + { DA732X_REG_REF1 , 0x02 }, + { DA732X_REG_BIAS_EN , 0x80 }, + { DA732X_REG_BIAS1 , 0x00 }, + { DA732X_REG_BIAS2 , 0x00 }, + { DA732X_REG_BIAS3 , 0x00 }, + { DA732X_REG_BIAS4 , 0x00 }, + { DA732X_REG_MICBIAS2 , 0x00 }, + { DA732X_REG_MICBIAS1 , 0x00 }, + { DA732X_REG_MICDET , 0x00 }, + { DA732X_REG_MIC1_PRE , 0x01 }, + { DA732X_REG_MIC1 , 0x40 }, + { DA732X_REG_MIC2_PRE , 0x01 }, + { DA732X_REG_MIC2 , 0x40 }, + { DA732X_REG_AUX1L , 0x75 }, + { DA732X_REG_AUX1R , 0x75 }, + { DA732X_REG_MIC3_PRE , 0x01 }, + { DA732X_REG_MIC3 , 0x40 }, + { DA732X_REG_INP_PINBIAS , 0x00 }, + { DA732X_REG_INP_ZC_EN , 0x00 }, + { DA732X_REG_INP_MUX , 0x50 }, + { DA732X_REG_HP_DET , 0x00 }, + { DA732X_REG_HPL_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPL_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPL_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPL , 0x40 }, + { DA732X_REG_HPL_VOL , 0x0F }, + { DA732X_REG_HPR_DAC_OFFSET , 0x00 }, + { DA732X_REG_HPR_DAC_OFF_CNTL , 0x00 }, + { DA732X_REG_HPR_OUT_OFFSET , 0x00 }, + { DA732X_REG_HPR , 0x40 }, + { DA732X_REG_HPR_VOL , 0x0F }, + { DA732X_REG_LIN2 , 0x4F }, + { DA732X_REG_LIN3 , 0x4F }, + { DA732X_REG_LIN4 , 0x4F }, + { DA732X_REG_OUT_ZC_EN , 0x00 }, + { DA732X_REG_HP_LIN1_GNDSEL , 0x00 }, + { DA732X_REG_CP_HP1 , 0x0C }, + { DA732X_REG_CP_HP2 , 0x03 }, + { DA732X_REG_CP_CTRL1 , 0x00 }, + { DA732X_REG_CP_CTRL2 , 0x99 }, + { DA732X_REG_CP_CTRL3 , 0x25 }, + { DA732X_REG_CP_LEVEL_MASK , 0x3F }, + { DA732X_REG_CP_DET , 0x00 }, + { DA732X_REG_CP_STATUS , 0x00 }, + { DA732X_REG_CP_THRESH1 , 0x00 }, + { DA732X_REG_CP_THRESH2 , 0x00 }, + { DA732X_REG_CP_THRESH3 , 0x00 }, + { DA732X_REG_CP_THRESH4 , 0x00 }, + { DA732X_REG_CP_THRESH5 , 0x00 }, + { DA732X_REG_CP_THRESH6 , 0x00 }, + { DA732X_REG_CP_THRESH7 , 0x00 }, + { DA732X_REG_CP_THRESH8 , 0x00 }, + { DA732X_REG_PLL_DIV_LO , 0x00 }, + { DA732X_REG_PLL_DIV_MID , 0x00 }, + { DA732X_REG_PLL_DIV_HI , 0x00 }, + { DA732X_REG_PLL_CTRL , 0x02 }, + { DA732X_REG_CLK_CTRL , 0xaa }, + { DA732X_REG_CLK_DSP , 0x07 }, + { DA732X_REG_CLK_EN1 , 0x00 }, + { DA732X_REG_CLK_EN2 , 0x00 }, + { DA732X_REG_CLK_EN3 , 0x00 }, + { DA732X_REG_CLK_EN4 , 0x00 }, + { DA732X_REG_CLK_EN5 , 0x00 }, + { DA732X_REG_AIF_MCLK , 0x00 }, + { DA732X_REG_AIFA1 , 0x02 }, + { DA732X_REG_AIFA2 , 0x00 }, + { DA732X_REG_AIFA3 , 0x08 }, + { DA732X_REG_AIFB1 , 0x02 }, + { DA732X_REG_AIFB2 , 0x00 }, + { DA732X_REG_AIFB3 , 0x08 }, + { DA732X_REG_PC_CTRL , 0xC0 }, + { DA732X_REG_DATA_ROUTE , 0x00 }, + { DA732X_REG_DSP_CTRL , 0x00 }, + { DA732X_REG_CIF_CTRL2 , 0x00 }, + { DA732X_REG_HANDSHAKE , 0x00 }, + { DA732X_REG_SPARE1_OUT , 0x00 }, + { DA732X_REG_SPARE2_OUT , 0x00 }, + { DA732X_REG_SPARE1_IN , 0x00 }, + { DA732X_REG_ADC1_PD , 0x00 }, + { DA732X_REG_ADC1_HPF , 0x00 }, + { DA732X_REG_ADC1_SEL , 0x00 }, + { DA732X_REG_ADC1_EQ12 , 0x00 }, + { DA732X_REG_ADC1_EQ34 , 0x00 }, + { DA732X_REG_ADC1_EQ5 , 0x00 }, + { DA732X_REG_ADC2_PD , 0x00 }, + { DA732X_REG_ADC2_HPF , 0x00 }, + { DA732X_REG_ADC2_SEL , 0x00 }, + { DA732X_REG_ADC2_EQ12 , 0x00 }, + { DA732X_REG_ADC2_EQ34 , 0x00 }, + { DA732X_REG_ADC2_EQ5 , 0x00 }, + { DA732X_REG_DAC1_HPF , 0x00 }, + { DA732X_REG_DAC1_L_VOL , 0x00 }, + { DA732X_REG_DAC1_R_VOL , 0x00 }, + { DA732X_REG_DAC1_SEL , 0x00 }, + { DA732X_REG_DAC1_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC1_EQ12 , 0x00 }, + { DA732X_REG_DAC1_EQ34 , 0x00 }, + { DA732X_REG_DAC1_EQ5 , 0x00 }, + { DA732X_REG_DAC2_HPF , 0x00 }, + { DA732X_REG_DAC2_L_VOL , 0x00 }, + { DA732X_REG_DAC2_R_VOL , 0x00 }, + { DA732X_REG_DAC2_SEL , 0x00 }, + { DA732X_REG_DAC2_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC2_EQ12 , 0x00 }, + { DA732X_REG_DAC2_EQ34 , 0x00 }, + { DA732X_REG_DAC2_EQ5 , 0x00 }, + { DA732X_REG_DAC3_HPF , 0x00 }, + { DA732X_REG_DAC3_VOL , 0x00 }, + { DA732X_REG_DAC3_SEL , 0x00 }, + { DA732X_REG_DAC3_SOFTMUTE , 0x00 }, + { DA732X_REG_DAC3_EQ12 , 0x00 }, + { DA732X_REG_DAC3_EQ34 , 0x00 }, + { DA732X_REG_DAC3_EQ5 , 0x00 }, + { DA732X_REG_BIQ_BYP , 0x00 }, + { DA732X_REG_DMA_CMD , 0x00 }, + { DA732X_REG_DMA_ADDR0 , 0x00 }, + { DA732X_REG_DMA_ADDR1 , 0x00 }, + { DA732X_REG_DMA_DATA0 , 0x00 }, + { DA732X_REG_DMA_DATA1 , 0x00 }, + { DA732X_REG_DMA_DATA2 , 0x00 }, + { DA732X_REG_DMA_DATA3 , 0x00 }, + { DA732X_REG_UNLOCK , 0x00 }, +}; + +static inline int da732x_get_input_div(struct snd_soc_codec *codec, int sysclk) +{ + int val; + int ret; + + if (sysclk < DA732X_MCLK_10MHZ) { + val = DA732X_MCLK_RET_0_10MHZ; + ret = DA732X_MCLK_VAL_0_10MHZ; + } else if ((sysclk >= DA732X_MCLK_10MHZ) && + (sysclk < DA732X_MCLK_20MHZ)) { + val = DA732X_MCLK_RET_10_20MHZ; + ret = DA732X_MCLK_VAL_10_20MHZ; + } else if ((sysclk >= DA732X_MCLK_20MHZ) && + (sysclk < DA732X_MCLK_40MHZ)) { + val = DA732X_MCLK_RET_20_40MHZ; + ret = DA732X_MCLK_VAL_20_40MHZ; + } else if ((sysclk >= DA732X_MCLK_40MHZ) && + (sysclk <= DA732X_MCLK_54MHZ)) { + val = DA732X_MCLK_RET_40_54MHZ; + ret = DA732X_MCLK_VAL_40_54MHZ; + } else { + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_PLL_CTRL, val); + + return ret; +} + +static void da732x_set_charge_pump(struct snd_soc_codec *codec, int state) +{ + switch (state) { + case DA732X_ENABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_EN); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_EN | + DA732X_HP_CP_REG | DA732X_HP_CP_PULSESKIP); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA732X_CP_EN | + DA732X_CP_CTRL_CPVDD1); + snd_soc_write(codec, DA732X_REG_CP_CTRL2, + DA732X_CP_MANAGE_MAGNITUDE | DA732X_CP_BOOST); + snd_soc_write(codec, DA732X_REG_CP_CTRL3, DA732X_CP_1MHZ); + break; + case DA732X_DISABLE_CP: + snd_soc_write(codec, DA732X_REG_CLK_EN2, DA732X_CP_CLK_DIS); + snd_soc_write(codec, DA732X_REG_CP_HP2, DA732X_HP_CP_DIS); + snd_soc_write(codec, DA732X_REG_CP_CTRL1, DA723X_CP_DIS); + break; + default: + pr_err(KERN_ERR "Wrong charge pump state\n"); + break; + } +} + +static const DECLARE_TLV_DB_SCALE(mic_boost_tlv, DA732X_MIC_PRE_VOL_DB_MIN, + DA732X_MIC_PRE_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, DA732X_MIC_VOL_DB_MIN, + DA732X_MIC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(aux_pga_tlv, DA732X_AUX_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(hp_pga_tlv, DA732X_HP_VOL_DB_MIN, + DA732X_AUX_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin2_pga_tlv, DA732X_LIN2_VOL_DB_MIN, + DA732X_LIN2_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin3_pga_tlv, DA732X_LIN3_VOL_DB_MIN, + DA732X_LIN3_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(lin4_pga_tlv, DA732X_LIN4_VOL_DB_MIN, + DA732X_LIN4_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(adc_pga_tlv, DA732X_ADC_VOL_DB_MIN, + DA732X_ADC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(dac_pga_tlv, DA732X_DAC_VOL_DB_MIN, + DA732X_DAC_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_band_pga_tlv, DA732X_EQ_BAND_VOL_DB_MIN, + DA732X_EQ_BAND_VOL_DB_INC, 0); + +static const DECLARE_TLV_DB_SCALE(eq_overall_tlv, DA732X_EQ_OVERALL_VOL_DB_MIN, + DA732X_EQ_OVERALL_VOL_DB_INC, 0); + +/* High Pass Filter */ +static const char *da732x_hpf_mode[] = { + "Disable", "Music", "Voice", +}; + +static const char *da732x_hpf_music[] = { + "1.8Hz", "3.75Hz", "7.5Hz", "15Hz", +}; + +static const char *da732x_hpf_voice[] = { + "2.5Hz", "25Hz", "50Hz", "100Hz", + "150Hz", "200Hz", "300Hz", "400Hz" +}; + +static const struct soc_enum da732x_dac1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac3_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc1_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_adc2_hpf_mode_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, + DA732X_HPF_MODE_MAX, da732x_hpf_mode) +}; + +static const struct soc_enum da732x_dac1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac3_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc1_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_adc2_hp_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, + DA732X_HPF_MUSIC_MAX, da732x_hpf_music) +}; + +static const struct soc_enum da732x_dac1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_dac3_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc1_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + +static const struct soc_enum da732x_adc2_voice_filter_enum[] = { + SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, + DA732X_HPF_VOICE_MAX, da732x_hpf_voice) +}; + + +static int da732x_hpf_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + unsigned int sel = ucontrol->value.integer.value[0]; + unsigned int bits; + + switch (sel) { + case DA732X_HPF_DISABLED: + bits = DA732X_HPF_DIS; + break; + case DA732X_HPF_VOICE: + bits = DA732X_HPF_VOICE_EN; + break; + case DA732X_HPF_MUSIC: + bits = DA732X_HPF_MUSIC_EN; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg, DA732X_HPF_MASK, bits); + + return 0; +} + +static int da732x_hpf_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; + unsigned int reg = enum_ctrl->reg; + int val; + + val = snd_soc_read(codec, reg) & DA732X_HPF_MASK; + + switch (val) { + case DA732X_HPF_VOICE_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_VOICE; + break; + case DA732X_HPF_MUSIC_EN: + ucontrol->value.integer.value[0] = DA732X_HPF_MUSIC; + break; + default: + ucontrol->value.integer.value[0] = DA732X_HPF_DISABLED; + break; + } + + return 0; +} + +static const struct snd_kcontrol_new da732x_snd_controls[] = { + /* Input PGAs */ + SOC_SINGLE_RANGE_TLV("MIC1 Boost Volume", DA732X_REG_MIC1_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC2 Boost Volume", DA732X_REG_MIC2_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + SOC_SINGLE_RANGE_TLV("MIC3 Boost Volume", DA732X_REG_MIC3_PRE, + DA732X_MICBOOST_SHIFT, DA732X_MICBOOST_MIN, + DA732X_MICBOOST_MAX, 0, mic_boost_tlv), + + /* MICs */ + SOC_SINGLE("MIC1 Switch", DA732X_REG_MIC1, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC1 Volume", DA732X_REG_MIC1, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC2 Switch", DA732X_REG_MIC2, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC2 Volume", DA732X_REG_MIC2, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + SOC_SINGLE("MIC3 Switch", DA732X_REG_MIC3, DA732X_MIC_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_RANGE_TLV("MIC3 Volume", DA732X_REG_MIC3, + DA732X_MIC_VOL_SHIFT, DA732X_MIC_VOL_VAL_MIN, + DA732X_MIC_VOL_VAL_MAX, 0, mic_pga_tlv), + + /* AUXs */ + SOC_SINGLE("AUX1L Switch", DA732X_REG_AUX1L, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1L Volume", DA732X_REG_AUX1L, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + SOC_SINGLE("AUX1R Switch", DA732X_REG_AUX1R, DA732X_AUX_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("AUX1R Volume", DA732X_REG_AUX1R, + DA732X_AUX_VOL_SHIFT, DA732X_AUX_VOL_VAL_MAX, + DA732X_NO_INVERT, aux_pga_tlv), + + /* ADCs */ + SOC_DOUBLE_TLV("ADC1 Volume", DA732X_REG_ADC1_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + SOC_DOUBLE_TLV("ADC2 Volume", DA732X_REG_ADC2_SEL, + DA732X_ADCL_VOL_SHIFT, DA732X_ADCR_VOL_SHIFT, + DA732X_ADC_VOL_VAL_MAX, DA732X_INVERT, adc_pga_tlv), + + /* DACs */ + SOC_DOUBLE("Digital Playback DAC12 Switch", DA732X_REG_DAC1_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_DACR_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Digital Playback DAC12 Volume", DA732X_REG_DAC1_L_VOL, + DA732X_REG_DAC1_R_VOL, DA732X_DAC_VOL_SHIFT, + DA732X_DAC_VOL_VAL_MAX, DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC3 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC3 Volume", DA732X_REG_DAC2_L_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC4 Switch", DA732X_REG_DAC2_SEL, + DA732X_DACR_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC4 Volume", DA732X_REG_DAC2_R_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + SOC_SINGLE("Digital Playback DAC5 Switch", DA732X_REG_DAC3_SEL, + DA732X_DACL_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Digital Playback DAC5 Volume", DA732X_REG_DAC3_VOL, + DA732X_DAC_VOL_SHIFT, DA732X_DAC_VOL_VAL_MAX, + DA732X_INVERT, dac_pga_tlv), + + /* High Pass Filters */ + SOC_ENUM_EXT("DAC1 High Pass Filter Mode", + da732x_dac1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC1 High Pass Filter", da732x_dac1_hp_filter_enum), + SOC_ENUM("DAC1 Voice Filter", da732x_dac1_voice_filter_enum), + + SOC_ENUM_EXT("DAC2 High Pass Filter Mode", + da732x_dac2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC2 High Pass Filter", da732x_dac2_hp_filter_enum), + SOC_ENUM("DAC2 Voice Filter", da732x_dac2_voice_filter_enum), + + SOC_ENUM_EXT("DAC3 High Pass Filter Mode", + da732x_dac3_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("DAC3 High Pass Filter", da732x_dac3_hp_filter_enum), + SOC_ENUM("DAC3 Filter Mode", da732x_dac3_voice_filter_enum), + + SOC_ENUM_EXT("ADC1 High Pass Filter Mode", + da732x_adc1_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC1 High Pass Filter", da732x_adc1_hp_filter_enum), + SOC_ENUM("ADC1 Voice Filter", da732x_adc1_voice_filter_enum), + + SOC_ENUM_EXT("ADC2 High Pass Filter Mode", + da732x_adc2_hpf_mode_enum, da732x_hpf_get, da732x_hpf_set), + SOC_ENUM("ADC2 High Pass Filter", da732x_adc2_hp_filter_enum), + SOC_ENUM("ADC2 Voice Filter", da732x_adc2_voice_filter_enum), + + /* Equalizers */ + SOC_SINGLE("ADC1 EQ Switch", DA732X_REG_ADC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC1 EQ Band 1 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 2 Volume", DA732X_REG_ADC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 3 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 4 Volume", DA732X_REG_ADC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Band 5 Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC1 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("ADC2 EQ Switch", DA732X_REG_ADC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("ADC2 EQ Band 1 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 2 Volume", DA732X_REG_ADC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Band 3 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 4 Volume", DA732X_REG_ADC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ACD2 EQ Band 5 Volume", DA732X_REG_ADC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("ADC2 EQ Overall Volume", DA732X_REG_ADC1_EQ5, + DA732X_EQ_OVERALL_SHIFT, DA732X_EQ_OVERALL_VOL_VAL_MAX, + DA732X_INVERT, eq_overall_tlv), + + SOC_SINGLE("DAC1 EQ Switch", DA732X_REG_DAC1_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC1 EQ Band 1 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 2 Volume", DA732X_REG_DAC1_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 3 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 4 Volume", DA732X_REG_DAC1_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC1 EQ Band 5 Volume", DA732X_REG_DAC1_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC2 EQ Switch", DA732X_REG_DAC2_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC2 EQ Band 1 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 2 Volume", DA732X_REG_DAC2_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 3 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 4 Volume", DA732X_REG_DAC2_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC2 EQ Band 5 Volume", DA732X_REG_DAC2_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + SOC_SINGLE("DAC3 EQ Switch", DA732X_REG_DAC3_EQ5, + DA732X_EQ_EN_SHIFT, DA732X_EQ_EN_MAX, DA732X_NO_INVERT), + SOC_SINGLE_TLV("DAC3 EQ Band 1 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND1_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 2 Volume", DA732X_REG_DAC3_EQ12, + DA732X_EQ_BAND2_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 3 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND3_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 4 Volume", DA732X_REG_DAC3_EQ34, + DA732X_EQ_BAND4_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + SOC_SINGLE_TLV("DAC3 EQ Band 5 Volume", DA732X_REG_DAC3_EQ5, + DA732X_EQ_BAND5_SHIFT, DA732X_EQ_VOL_VAL_MAX, + DA732X_INVERT, eq_band_pga_tlv), + + /* Lineout 2 Reciever*/ + SOC_SINGLE("Lineout 2 Switch", DA732X_REG_LIN2, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 2 Volume", DA732X_REG_LIN2, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin2_pga_tlv), + + /* Lineout 3 SPEAKER*/ + SOC_SINGLE("Lineout 3 Switch", DA732X_REG_LIN3, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 3 Volume", DA732X_REG_LIN3, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin3_pga_tlv), + + /* Lineout 4 */ + SOC_SINGLE("Lineout 4 Switch", DA732X_REG_LIN4, DA732X_LOUT_MUTE_SHIFT, + DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_SINGLE_TLV("Lineout 4 Volume", DA732X_REG_LIN4, + DA732X_LOUT_VOL_SHIFT, DA732X_LOUT_VOL_VAL_MAX, + DA732X_NO_INVERT, lin4_pga_tlv), + + /* Headphones */ + SOC_DOUBLE_R("Headphone Switch", DA732X_REG_HPR, DA732X_REG_HPL, + DA732X_HP_MUTE_SHIFT, DA732X_SWITCH_MAX, DA732X_INVERT), + SOC_DOUBLE_R_TLV("Headphone Volume", DA732X_REG_HPL_VOL, + DA732X_REG_HPR_VOL, DA732X_HP_VOL_SHIFT, + DA732X_HP_VOL_VAL_MAX, DA732X_NO_INVERT, hp_pga_tlv), +}; + +static int da732x_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, + DA732X_ADCA_BB_CLK_EN); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, + DA732X_ADCC_BB_CLK_EN); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_ACT); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_ON); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, DA732X_ADC_PD_MASK, + DA732X_ADC_OFF); + snd_soc_update_bits(codec, w->reg, DA732X_ADC_RST_MASK, + DA732X_ADC_SET_RST); + + switch (w->reg) { + case DA732X_REG_ADC1_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCA_BB_CLK_EN, 0); + break; + case DA732X_REG_ADC2_PD: + snd_soc_update_bits(codec, DA732X_REG_CLK_EN3, + DA732X_ADCC_BB_CLK_EN, 0); + break; + default: + return -EINVAL; + } + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int da732x_out_pga_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_EN); + break; + case SND_SOC_DAPM_POST_PMD: + snd_soc_update_bits(codec, w->reg, + (1 << w->shift) | DA732X_OUT_HIZ_EN, + (1 << w->shift) | DA732X_OUT_HIZ_DIS); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char *adcl_text[] = { + "AUX1L", "MIC1" +}; + +static const char *adcr_text[] = { + "AUX1R", "MIC2", "MIC3" +}; + +static const char *enable_text[] = { + "Disabled", + "Enabled" +}; + +/* ADC1LMUX */ +static const struct soc_enum adc1l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc1l_mux = + SOC_DAPM_ENUM("ADC Route", adc1l_enum); + +/* ADC1RMUX */ +static const struct soc_enum adc1r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); +static const struct snd_kcontrol_new adc1r_mux = + SOC_DAPM_ENUM("ADC Route", adc1r_enum); + +/* ADC2LMUX */ +static const struct soc_enum adc2l_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, + DA732X_ADCL_MUX_MAX, adcl_text); +static const struct snd_kcontrol_new adc2l_mux = + SOC_DAPM_ENUM("ADC Route", adc2l_enum); + +/* ADC2RMUX */ +static const struct soc_enum adc2r_enum = + SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, + DA732X_ADCR_MUX_MAX, adcr_text); + +static const struct snd_kcontrol_new adc2r_mux = + SOC_DAPM_ENUM("ADC Route", adc2r_enum); + +static const struct soc_enum da732x_hp_left_output = + SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpl_mux = + SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output); + +static const struct soc_enum da732x_hp_right_output = + SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new hpr_mux = + SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output); + +static const struct soc_enum da732x_speaker_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new spk_mux = + SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output); + +static const struct soc_enum da732x_lout4_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout4_mux = + SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output); + +static const struct soc_enum da732x_lout2_output = + SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, + DA732X_DAC_EN_MAX, enable_text); + +static const struct snd_kcontrol_new lout2_mux = + SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output); + +static const struct snd_soc_dapm_widget da732x_dapm_widgets[] = { + /* Supplies */ + SND_SOC_DAPM_SUPPLY("ADC1 Supply", DA732X_REG_ADC1_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("ADC2 Supply", DA732X_REG_ADC2_PD, 0, + DA732X_NO_INVERT, da732x_adc_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SUPPLY("DAC1 CLK", DA732X_REG_CLK_EN4, + DA732X_DACA_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC2 CLK", DA732X_REG_CLK_EN4, + DA732X_DACC_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC3 CLK", DA732X_REG_CLK_EN5, + DA732X_DACE_BB_CLK_SHIFT, DA732X_NO_INVERT, + NULL, 0), + + /* Micbias */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", DA732X_REG_MICBIAS1, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", DA732X_REG_MICBIAS2, + DA732X_MICBIAS_EN_SHIFT, + DA732X_NO_INVERT, NULL, 0), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("MIC2"), + SND_SOC_DAPM_INPUT("MIC3"), + SND_SOC_DAPM_INPUT("AUX1L"), + SND_SOC_DAPM_INPUT("AUX1R"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + SND_SOC_DAPM_OUTPUT("LOUTL"), + SND_SOC_DAPM_OUTPUT("LOUTR"), + SND_SOC_DAPM_OUTPUT("ClassD"), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1L", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC1R", NULL, DA732X_REG_ADC1_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2L", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_ADC("ADC2R", NULL, DA732X_REG_ADC2_SEL, + DA732X_ADCR_EN_SHIFT, DA732X_NO_INVERT), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC1L", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC1R", NULL, DA732X_REG_DAC1_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2L", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC2R", NULL, DA732X_REG_DAC2_SEL, + DA732X_DACR_EN_SHIFT, DA732X_NO_INVERT), + SND_SOC_DAPM_DAC("DAC3", NULL, DA732X_REG_DAC3_SEL, + DA732X_DACL_EN_SHIFT, DA732X_NO_INVERT), + + /* Input Pgas */ + SND_SOC_DAPM_PGA("MIC1 PGA", DA732X_REG_MIC1, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC2 PGA", DA732X_REG_MIC2, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("MIC3 PGA", DA732X_REG_MIC3, DA732X_MIC_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1L PGA", DA732X_REG_AUX1L, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("AUX1R PGA", DA732X_REG_AUX1R, DA732X_AUX_EN_SHIFT, + 0, NULL, 0), + + SND_SOC_DAPM_PGA_E("HP Left", DA732X_REG_HPL, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("HP Right", DA732X_REG_HPR, DA732X_HP_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN2", DA732X_REG_LIN2, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN3", DA732X_REG_LIN3, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_PGA_E("LIN4", DA732X_REG_LIN4, DA732X_LIN_OUT_EN_SHIFT, + 0, NULL, 0, da732x_out_pga_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* MUXs */ + SND_SOC_DAPM_MUX("ADC1 Left MUX", SND_SOC_NOPM, 0, 0, &adc1l_mux), + SND_SOC_DAPM_MUX("ADC1 Right MUX", SND_SOC_NOPM, 0, 0, &adc1r_mux), + SND_SOC_DAPM_MUX("ADC2 Left MUX", SND_SOC_NOPM, 0, 0, &adc2l_mux), + SND_SOC_DAPM_MUX("ADC2 Right MUX", SND_SOC_NOPM, 0, 0, &adc2r_mux), + + SND_SOC_DAPM_MUX("HP Left MUX", SND_SOC_NOPM, 0, 0, &hpl_mux), + SND_SOC_DAPM_MUX("HP Right MUX", SND_SOC_NOPM, 0, 0, &hpr_mux), + SND_SOC_DAPM_MUX("Speaker MUX", SND_SOC_NOPM, 0, 0, &spk_mux), + SND_SOC_DAPM_MUX("LOUT2 MUX", SND_SOC_NOPM, 0, 0, &lout2_mux), + SND_SOC_DAPM_MUX("LOUT4 MUX", SND_SOC_NOPM, 0, 0, &lout4_mux), + + /* AIF interfaces */ + SND_SOC_DAPM_AIF_OUT("AIFA Output", "AIFA Capture", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFA Input", "AIFA Playback", 0, DA732X_REG_AIFA3, + DA732X_AIF_EN_SHIFT, 0), + + SND_SOC_DAPM_AIF_OUT("AIFB Output", "AIFB Capture", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), + SND_SOC_DAPM_AIF_IN("AIFB Input", "AIFB Playback", 0, DA732X_REG_AIFB3, + DA732X_AIF_EN_SHIFT, 0), +}; + +static const struct snd_soc_dapm_route da732x_dapm_routes[] = { + /* Inputs */ + {"AUX1L PGA", "NULL", "AUX1L"}, + {"AUX1R PGA", "NULL", "AUX1R"}, + {"MIC1 PGA", NULL, "MIC1"}, + {"MIC2 PGA", "NULL", "MIC2"}, + {"MIC3 PGA", "NULL", "MIC3"}, + + /* Capture Path */ + {"ADC1 Left MUX", "MIC1", "MIC1 PGA"}, + {"ADC1 Left MUX", "AUX1L", "AUX1L PGA"}, + + {"ADC1 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC1 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC1 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC2 Left MUX", "AUX1L", "AUX1L PGA"}, + {"ADC2 Left MUX", "MIC1", "MIC1 PGA"}, + + {"ADC2 Right MUX", "AUX1R", "AUX1R PGA"}, + {"ADC2 Right MUX", "MIC2", "MIC2 PGA"}, + {"ADC2 Right MUX", "MIC3", "MIC3 PGA"}, + + {"ADC1L", NULL, "ADC1 Supply"}, + {"ADC1R", NULL, "ADC1 Supply"}, + {"ADC2L", NULL, "ADC2 Supply"}, + {"ADC2R", NULL, "ADC2 Supply"}, + + {"ADC1L", NULL, "ADC1 Left MUX"}, + {"ADC1R", NULL, "ADC1 Right MUX"}, + {"ADC2L", NULL, "ADC2 Left MUX"}, + {"ADC2R", NULL, "ADC2 Right MUX"}, + + {"AIFA Output", NULL, "ADC1L"}, + {"AIFA Output", NULL, "ADC1R"}, + {"AIFB Output", NULL, "ADC2L"}, + {"AIFB Output", NULL, "ADC2R"}, + + {"HP Left MUX", "Enabled", "AIFA Input"}, + {"HP Right MUX", "Enabled", "AIFA Input"}, + {"Speaker MUX", "Enabled", "AIFB Input"}, + {"LOUT2 MUX", "Enabled", "AIFB Input"}, + {"LOUT4 MUX", "Enabled", "AIFB Input"}, + + {"DAC1L", NULL, "DAC1 CLK"}, + {"DAC1R", NULL, "DAC1 CLK"}, + {"DAC2L", NULL, "DAC2 CLK"}, + {"DAC2R", NULL, "DAC2 CLK"}, + {"DAC3", NULL, "DAC3 CLK"}, + + {"DAC1L", NULL, "HP Left MUX"}, + {"DAC1R", NULL, "HP Right MUX"}, + {"DAC2L", NULL, "Speaker MUX"}, + {"DAC2R", NULL, "LOUT4 MUX"}, + {"DAC3", NULL, "LOUT2 MUX"}, + + /* Output Pgas */ + {"HP Left", NULL, "DAC1L"}, + {"HP Right", NULL, "DAC1R"}, + {"LIN3", NULL, "DAC2L"}, + {"LIN4", NULL, "DAC2R"}, + {"LIN2", NULL, "DAC3"}, + + /* Outputs */ + {"ClassD", NULL, "LIN3"}, + {"LOUTL", NULL, "LIN2"}, + {"LOUTR", NULL, "LIN4"}, + {"HPL", NULL, "HP Left"}, + {"HPR", NULL, "HP Right"}, +}; + +static int da732x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif = 0; + u32 reg_aif; + u32 fs; + + reg_aif = dai->driver->base; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + aif |= DA732X_AIF_WORD_16; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + aif |= DA732X_AIF_WORD_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + aif |= DA732X_AIF_WORD_24; + break; + case SNDRV_PCM_FORMAT_S32_LE: + aif |= DA732X_AIF_WORD_32; + break; + default: + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + fs = DA732X_SR_8KHZ; + break; + case 11025: + fs = DA732X_SR_11_025KHZ; + break; + case 12000: + fs = DA732X_SR_12KHZ; + break; + case 16000: + fs = DA732X_SR_16KHZ; + break; + case 22050: + fs = DA732X_SR_22_05KHZ; + break; + case 24000: + fs = DA732X_SR_24KHZ; + break; + case 32000: + fs = DA732X_SR_32KHZ; + break; + case 44100: + fs = DA732X_SR_44_1KHZ; + break; + case 48000: + fs = DA732X_SR_48KHZ; + break; + case 88100: + fs = DA732X_SR_88_1KHZ; + break; + case 96000: + fs = DA732X_SR_96KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, reg_aif, DA732X_AIF_WORD_MASK, aif); + snd_soc_update_bits(codec, DA732X_REG_CLK_CTRL, DA732X_SR1_MASK, fs); + + return 0; +} + +static int da732x_set_dai_fmt(struct snd_soc_dai *dai, u32 fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u32 aif_mclk, pc_count; + u32 reg_aif1, aif1; + u32 reg_aif3, aif3; + + switch (dai->id) { + case DA732X_DAI_ID1: + reg_aif1 = DA732X_REG_AIFA1; + reg_aif3 = DA732X_REG_AIFA3; + pc_count = DA732X_PC_PULSE_AIFA | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + case DA732X_DAI_ID2: + reg_aif1 = DA732X_REG_AIFB1; + reg_aif3 = DA732X_REG_AIFB3; + pc_count = DA732X_PC_PULSE_AIFB | DA732X_PC_RESYNC_NOT_AUT | + DA732X_PC_SAME; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + aif1 = DA732X_AIF_SLAVE; + aif_mclk = DA732X_AIFM_FRAME_64 | DA732X_AIFM_SRC_SEL_AIFA; + break; + case SND_SOC_DAIFMT_CBM_CFM: + aif1 = DA732X_AIF_CLK_FROM_SRC; + aif_mclk = DA732X_CLK_GENERATION_AIF_A; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + aif3 = DA732X_AIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_RIGHT_J: + aif3 = DA732X_AIF_RIGHT_J_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + aif3 = DA732X_AIF_LEFT_J_MODE; + break; + case SND_SOC_DAIFMT_DSP_B: + aif3 = DA732X_AIF_DSP_MODE; + break; + default: + return -EINVAL; + } + + /* Clock inversion */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_B: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + default: + return -EINVAL; + } + break; + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + aif3 |= DA732X_AIF_BCLK_INV | DA732X_AIF_WCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + aif3 |= DA732X_AIF_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + aif3 |= DA732X_AIF_WCLK_INV; + break; + default: + return -EINVAL; + } + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, DA732X_REG_AIF_MCLK, aif_mclk); + snd_soc_update_bits(codec, reg_aif1, DA732X_AIF1_CLK_MASK, aif1); + snd_soc_update_bits(codec, reg_aif3, DA732X_AIF_BCLK_INV | + DA732X_AIF_WCLK_INV | DA732X_AIF_MODE_MASK, aif3); + snd_soc_write(codec, DA732X_REG_PC_CTRL, pc_count); + + return 0; +} + + + +static int da732x_set_dai_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + int fref, indiv; + u8 div_lo, div_mid, div_hi; + u64 frac_div; + + /* Disable PLL */ + if (freq_out == 0) { + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + return 0; + } + + if (da732x->pll_en) + return -EBUSY; + + if (source == DA732X_SRCCLK_MCLK) { + /* Validate Sysclk rate */ + switch (da732x->sysclk) { + case 11290000: + case 12288000: + case 22580000: + case 24576000: + case 45160000: + case 49152000: + snd_soc_write(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_BYPASS); + return 0; + default: + dev_err(codec->dev, + "Cannot use PLL Bypass, invalid SYSCLK rate\n"); + return -EINVAL; + } + } + + indiv = da732x_get_input_div(codec, da732x->sysclk); + if (indiv < 0) + return indiv; + + fref = (da732x->sysclk / indiv); + div_hi = freq_out / fref; + frac_div = (u64)(freq_out % fref) * 8192ULL; + do_div(frac_div, fref); + div_mid = (frac_div >> DA732X_1BYTE_SHIFT) & DA732X_U8_MASK; + div_lo = (frac_div) & DA732X_U8_MASK; + + snd_soc_write(codec, DA732X_REG_PLL_DIV_LO, div_lo); + snd_soc_write(codec, DA732X_REG_PLL_DIV_MID, div_mid); + snd_soc_write(codec, DA732X_REG_PLL_DIV_HI, div_hi); + + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, DA732X_PLL_EN, + DA732X_PLL_EN); + + da732x->pll_en = true; + + return 0; +} + +static int da732x_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + da732x->sysclk = freq; + + return 0; +} + +#define DA732X_RATES SNDRV_PCM_RATE_8000_96000 + +#define DA732X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops da732x_dai1_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_ops da732x_dai2_ops = { + .hw_params = da732x_hw_params, + .set_fmt = da732x_set_dai_fmt, + .set_sysclk = da732x_set_dai_sysclk, +}; + +static struct snd_soc_dai_driver da732x_dai[] = { + { + .name = "DA732X_AIFA", + .id = DA732X_DAI_ID1, + .base = DA732X_REG_AIFA1, + .playback = { + .stream_name = "AIFA Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFA Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai1_ops, + }, + { + .name = "DA732X_AIFB", + .id = DA732X_DAI_ID2, + .base = DA732X_REG_AIFB1, + .playback = { + .stream_name = "AIFB Playback", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .capture = { + .stream_name = "AIFB Capture", + .channels_min = 1, + .channels_max = 2, + .rates = DA732X_RATES, + .formats = DA732X_FORMATS, + }, + .ops = &da732x_dai2_ops, + }, +}; + +static const struct regmap_config da732x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = DA732X_MAX_REG, + .reg_defaults = da732x_reg_cache, + .num_reg_defaults = ARRAY_SIZE(da732x_reg_cache), + .cache_type = REGCACHE_RBTREE, +}; + + +static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_DACS]; + u8 sign[DA732X_HP_DACS]; + u8 step = DA732X_DAC_OFFSET_STEP; + + /* Initialize DAC offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + DA732X_HP_DAC_OFFSET_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_CALIBRATION | + DA732X_HP_DAC_OFF_SCALE_STEPS); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check DAC offset sign */ + sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO); + + /* Binary search DAC offset values (both channels at once) */ + offset[DA732X_HPL_DAC] = sign[DA732X_HPL_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + offset[DA732X_HPR_DAC] = sign[DA732X_HPR_DAC] << DA732X_HP_DAC_COMPO_SHIFT; + + do { + offset[DA732X_HPL_DAC] |= step; + offset[DA732X_HPR_DAC] |= step; + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) + offset[DA732X_HPL_DAC] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) + offset[DA732X_HPR_DAC] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFFSET, + ~offset[DA732X_HPL_DAC] & DA732X_HP_DAC_OFF_MASK); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFFSET, + ~offset[DA732X_HPR_DAC] & DA732X_HP_DAC_OFF_MASK); + + /* End DAC calibration mode */ + snd_soc_write(codec, DA732X_REG_HPL_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); + snd_soc_write(codec, DA732X_REG_HPR_DAC_OFF_CNTL, + DA732X_HP_DAC_OFF_SCALE_STEPS); +} + +static void da732x_output_offset_adjust(struct snd_soc_codec *codec) +{ + u8 offset[DA732X_HP_AMPS]; + u8 sign[DA732X_HP_AMPS]; + u8 step = DA732X_OUTPUT_OFFSET_STEP; + + offset[DA732X_HPL_AMP] = DA732X_HP_OUT_TRIM_VAL; + offset[DA732X_HPR_AMP] = DA732X_HP_OUT_TRIM_VAL; + + /* Initialize output offset calibration circuits and registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, DA732X_HP_OUT_TRIM_VAL); + snd_soc_write(codec, DA732X_REG_HPL, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, + DA732X_HP_OUT_COMP | DA732X_HP_OUT_EN); + + /* Wait for voltage stabilization */ + msleep(DA732X_WAIT_FOR_STABILIZATION); + + /* Check output offset sign */ + sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO; + sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO; + + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | + (sign[DA732X_HPL_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_COMP | + (sign[DA732X_HPR_AMP] >> DA732X_HP_OUT_COMPO_SHIFT) | + DA732X_HP_OUT_EN); + + /* Binary search output offset values (both channels at once) */ + do { + offset[DA732X_HPL_AMP] |= step; + offset[DA732X_HPR_AMP] |= step; + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, + offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, + offset[DA732X_HPR_AMP]); + + msleep(DA732X_WAIT_FOR_STABILIZATION); + + if ((codec->hw_read(codec, DA732X_REG_HPL) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) + offset[DA732X_HPL_AMP] &= ~step; + if ((codec->hw_read(codec, DA732X_REG_HPR) & + DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) + offset[DA732X_HPR_AMP] &= ~step; + + step >>= 1; + } while (step); + + /* Write final DAC offsets to registers */ + snd_soc_write(codec, DA732X_REG_HPL_OUT_OFFSET, offset[DA732X_HPL_AMP]); + snd_soc_write(codec, DA732X_REG_HPR_OUT_OFFSET, offset[DA732X_HPR_AMP]); +} + +static void da732x_hp_dc_offset_cancellation(struct snd_soc_codec *codec) +{ + /* Make sure that we have Soft Mute enabled */ + snd_soc_write(codec, DA732X_REG_DAC1_SOFTMUTE, DA732X_SOFTMUTE_EN | + DA732X_GAIN_RAMPED | DA732X_16_SAMPLES); + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACL_EN | + DA732X_DACR_EN | DA732X_DACL_SDM | DA732X_DACR_SDM | + DA732X_DACL_MUTE | DA732X_DACR_MUTE); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_EN); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_OUT_EN | + DA732X_HP_OUT_MUTE | DA732X_HP_OUT_DAC_EN); + + da732x_dac_offset_adjust(codec); + da732x_output_offset_adjust(codec); + + snd_soc_write(codec, DA732X_REG_DAC1_SEL, DA732X_DACS_DIS); + snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_DIS); + snd_soc_write(codec, DA732X_REG_HPR, DA732X_HP_DIS); +} + +static int da732x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_100PC); + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* Init Codec */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_VMID_FASTCHG); + snd_soc_write(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_EN); + + mdelay(DA732X_STARTUP_DELAY); + + /* Disable Fast Charge and enable DAC ref voltage */ + snd_soc_write(codec, DA732X_REG_REF1, + DA732X_REFBUFX2_EN); + + /* Enable bypass DSP routing */ + snd_soc_write(codec, DA732X_REG_DATA_ROUTE, + DA732X_BYPASS_DSP); + + /* Enable Digital subsystem */ + snd_soc_write(codec, DA732X_REG_DSP_CTRL, + DA732X_DIGITAL_EN); + + snd_soc_write(codec, DA732X_REG_SPARE1_OUT, + DA732X_HP_DRIVER_EN | + DA732X_HP_GATE_LOW | + DA732X_HP_LOOP_GAIN_CTRL); + snd_soc_write(codec, DA732X_REG_HP_LIN1_GNDSEL, + DA732X_HP_OUT_GNDSEL); + + da732x_set_charge_pump(codec, DA732X_ENABLE_CP); + + snd_soc_write(codec, DA732X_REG_CLK_EN1, + DA732X_SYS3_CLK_EN | DA732X_PC_CLK_EN); + + /* Enable Zero Crossing */ + snd_soc_write(codec, DA732X_REG_INP_ZC_EN, + DA732X_MIC1_PRE_ZC_EN | + DA732X_MIC1_ZC_EN | + DA732X_MIC2_PRE_ZC_EN | + DA732X_MIC2_ZC_EN | + DA732X_AUXL_ZC_EN | + DA732X_AUXR_ZC_EN | + DA732X_MIC3_PRE_ZC_EN | + DA732X_MIC3_ZC_EN); + snd_soc_write(codec, DA732X_REG_OUT_ZC_EN, + DA732X_HPL_ZC_EN | DA732X_HPR_ZC_EN | + DA732X_LIN2_ZC_EN | DA732X_LIN3_ZC_EN | + DA732X_LIN4_ZC_EN); + + da732x_hp_dc_offset_cancellation(codec); + + regcache_cache_only(codec->control_data, false); + regcache_sync(codec->control_data); + } else { + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, + DA732X_BIAS_BOOST_MASK, + DA732X_BIAS_BOOST_50PC); + snd_soc_update_bits(codec, DA732X_REG_PLL_CTRL, + DA732X_PLL_EN, 0); + da732x->pll_en = false; + } + break; + case SND_SOC_BIAS_OFF: + regcache_cache_only(codec->control_data, true); + da732x_set_charge_pump(codec, DA732X_DISABLE_CP); + snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, + DA732X_BIAS_DIS); + da732x->pll_en = false; + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static int da732x_probe(struct snd_soc_codec *codec) +{ + struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + int ret = 0; + + da732x->codec = codec; + + dapm->idle_bias_off = false; + + codec->control_data = da732x->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec.\n"); + goto err; + } + + da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +err: + return ret; +} + +static int da732x_remove(struct snd_soc_codec *codec) +{ + + da732x_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +struct snd_soc_codec_driver soc_codec_dev_da732x = { + .probe = da732x_probe, + .remove = da732x_remove, + .set_bias_level = da732x_set_bias_level, + .controls = da732x_snd_controls, + .num_controls = ARRAY_SIZE(da732x_snd_controls), + .dapm_widgets = da732x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(da732x_dapm_widgets), + .dapm_routes = da732x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), + .set_pll = da732x_set_dai_pll, + .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), +}; + +static __devinit int da732x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct da732x_priv *da732x; + unsigned int reg; + int ret; + + da732x = devm_kzalloc(&i2c->dev, sizeof(struct da732x_priv), + GFP_KERNEL); + if (!da732x) + return -ENOMEM; + + i2c_set_clientdata(i2c, da732x); + + da732x->regmap = devm_regmap_init_i2c(i2c, &da732x_regmap); + if (IS_ERR(da732x->regmap)) { + ret = PTR_ERR(da732x->regmap); + dev_err(&i2c->dev, "Failed to initialize regmap\n"); + goto err; + } + + ret = regmap_read(da732x->regmap, DA732X_REG_ID, ®); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err; + } + + dev_info(&i2c->dev, "Revision: %d.%d\n", + (reg & DA732X_ID_MAJOR_MASK), (reg & DA732X_ID_MINOR_MASK)); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_da732x, + da732x_dai, ARRAY_SIZE(da732x_dai)); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register codec.\n"); + +err: + return ret; +} + +static __devexit int da732x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id da732x_i2c_id[] = { + { "da7320", 0}, + { } +}; +MODULE_DEVICE_TABLE(i2c, da732x_i2c_id); + +static struct i2c_driver da732x_i2c_driver = { + .driver = { + .name = "da7320", + .owner = THIS_MODULE, + }, + .probe = da732x_i2c_probe, + .remove = __devexit_p(da732x_i2c_remove), + .id_table = da732x_i2c_id, +}; + +module_i2c_driver(da732x_i2c_driver); + + +MODULE_DESCRIPTION("ASoC DA732X driver"); +MODULE_AUTHOR("Michal Hajduk "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h new file mode 100644 index 0000000..c8ce547 --- /dev/null +++ b/sound/soc/codecs/da732x.h @@ -0,0 +1,133 @@ +/* + * da732x.h -- Dialog DA732X ALSA SoC Audio Driver Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_H_ +#define __DA732X_H + +#include + +/* General */ +#define DA732X_U8_MASK 0xFF +#define DA732X_4BYTES 4 +#define DA732X_3BYTES 3 +#define DA732X_2BYTES 2 +#define DA732X_1BYTE 1 +#define DA732X_1BYTE_SHIFT 8 +#define DA732X_2BYTES_SHIFT 16 +#define DA732X_3BYTES_SHIFT 24 +#define DA732X_4BYTES_SHIFT 32 + +#define DA732X_DACS_DIS 0x0 +#define DA732X_HP_DIS 0x0 +#define DA732X_CLEAR_REG 0x0 + +/* Calibration */ +#define DA732X_DAC_OFFSET_STEP 0x20 +#define DA732X_OUTPUT_OFFSET_STEP 0x80 +#define DA732X_HP_OUT_TRIM_VAL 0x0 +#define DA732X_WAIT_FOR_STABILIZATION 1 +#define DA732X_HPL_DAC 0 +#define DA732X_HPR_DAC 1 +#define DA732X_HP_DACS 2 +#define DA732X_HPL_AMP 0 +#define DA732X_HPR_AMP 1 +#define DA732X_HP_AMPS 2 + +/* Clock settings */ +#define DA732X_STARTUP_DELAY 100 +#define DA732X_PLL_OUT_196608 196608000 +#define DA732X_PLL_OUT_180634 180633600 +#define DA732X_PLL_OUT_SRM 188620800 +#define DA732X_MCLK_10MHZ 10000000 +#define DA732X_MCLK_20MHZ 20000000 +#define DA732X_MCLK_40MHZ 40000000 +#define DA732X_MCLK_54MHZ 54000000 +#define DA732X_MCLK_RET_0_10MHZ 0 +#define DA732X_MCLK_VAL_0_10MHZ 1 +#define DA732X_MCLK_RET_10_20MHZ 1 +#define DA732X_MCLK_VAL_10_20MHZ 2 +#define DA732X_MCLK_RET_20_40MHZ 2 +#define DA732X_MCLK_VAL_20_40MHZ 4 +#define DA732X_MCLK_RET_40_54MHZ 3 +#define DA732X_MCLK_VAL_40_54MHZ 8 +#define DA732X_DAI_ID1 0 +#define DA732X_DAI_ID2 1 +#define DA732X_SRCCLK_PLL 0 +#define DA732X_SRCCLK_MCLK 1 + +#define DA732X_LIN_LP_VOL 0x4F +#define DA732X_LP_VOL 0x40 + +/* Kcontrols */ +#define DA732X_DAC_EN_MAX 2 +#define DA732X_ADCL_MUX_MAX 2 +#define DA732X_ADCR_MUX_MAX 3 +#define DA732X_HPF_MODE_MAX 3 +#define DA732X_HPF_MODE_SHIFT 4 +#define DA732X_HPF_MUSIC_SHIFT 0 +#define DA732X_HPF_MUSIC_MAX 4 +#define DA732X_HPF_VOICE_SHIFT 4 +#define DA732X_HPF_VOICE_MAX 8 +#define DA732X_EQ_EN_MAX 1 +#define DA732X_HPF_VOICE 1 +#define DA732X_HPF_MUSIC 2 +#define DA732X_HPF_DISABLED 0 +#define DA732X_NO_INVERT 0 +#define DA732X_INVERT 1 +#define DA732X_SWITCH_MAX 1 +#define DA732X_ENABLE_CP 1 +#define DA732X_DISABLE_CP 0 +#define DA732X_DISABLE_ALL_CLKS 0 +#define DA732X_RESET_ADCS 0 + +/* dB values */ +#define DA732X_MIC_VOL_DB_MIN 0 +#define DA732X_MIC_VOL_DB_INC 50 +#define DA732X_MIC_PRE_VOL_DB_MIN 0 +#define DA732X_MIC_PRE_VOL_DB_INC 600 +#define DA732X_AUX_VOL_DB_MIN -6000 +#define DA732X_AUX_VOL_DB_INC 150 +#define DA732X_HP_VOL_DB_MIN -2250 +#define DA732X_HP_VOL_DB_INC 150 +#define DA732X_LIN2_VOL_DB_MIN -1650 +#define DA732X_LIN2_VOL_DB_INC 150 +#define DA732X_LIN3_VOL_DB_MIN -1650 +#define DA732X_LIN3_VOL_DB_INC 150 +#define DA732X_LIN4_VOL_DB_MIN -2250 +#define DA732X_LIN4_VOL_DB_INC 150 +#define DA732X_EQ_BAND_VOL_DB_MIN -1050 +#define DA732X_EQ_BAND_VOL_DB_INC 150 +#define DA732X_DAC_VOL_DB_MIN -7725 +#define DA732X_DAC_VOL_DB_INC 75 +#define DA732X_ADC_VOL_DB_MIN 0 +#define DA732X_ADC_VOL_DB_INC -1 +#define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 +#define DA732X_EQ_OVERALL_VOL_DB_INC 600 + +#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ + {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} + +enum da732x_sysctl { + DA732X_SR_8KHZ = 0x1, + DA732X_SR_11_025KHZ = 0x2, + DA732X_SR_12KHZ = 0x3, + DA732X_SR_16KHZ = 0x5, + DA732X_SR_22_05KHZ = 0x6, + DA732X_SR_24KHZ = 0x7, + DA732X_SR_32KHZ = 0x9, + DA732X_SR_44_1KHZ = 0xA, + DA732X_SR_48KHZ = 0xB, + DA732X_SR_88_1KHZ = 0xE, + DA732X_SR_96KHZ = 0xF, +}; + +#endif /* __DA732X_H_ */ diff --git a/sound/soc/codecs/da732x_reg.h b/sound/soc/codecs/da732x_reg.h new file mode 100644 index 0000000..bdd03ca --- /dev/null +++ b/sound/soc/codecs/da732x_reg.h @@ -0,0 +1,654 @@ +/* + * da732x_reg.h --- Dialog DA732X ALSA SoC Audio Registers Header File + * + * Copyright (C) 2012 Dialog Semiconductor GmbH + * + * Author: Michal Hajduk + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __DA732X_REG_H_ +#define __DA732X_REG_H_ + +/* DA732X registers */ +#define DA732X_REG_STATUS_EXT 0x00 +#define DA732X_REG_STATUS 0x01 +#define DA732X_REG_REF1 0x02 +#define DA732X_REG_BIAS_EN 0x03 +#define DA732X_REG_BIAS1 0x04 +#define DA732X_REG_BIAS2 0x05 +#define DA732X_REG_BIAS3 0x06 +#define DA732X_REG_BIAS4 0x07 +#define DA732X_REG_MICBIAS2 0x0F +#define DA732X_REG_MICBIAS1 0x10 +#define DA732X_REG_MICDET 0x11 +#define DA732X_REG_MIC1_PRE 0x12 +#define DA732X_REG_MIC1 0x13 +#define DA732X_REG_MIC2_PRE 0x14 +#define DA732X_REG_MIC2 0x15 +#define DA732X_REG_AUX1L 0x16 +#define DA732X_REG_AUX1R 0x17 +#define DA732X_REG_MIC3_PRE 0x18 +#define DA732X_REG_MIC3 0x19 +#define DA732X_REG_INP_PINBIAS 0x1A +#define DA732X_REG_INP_ZC_EN 0x1B +#define DA732X_REG_INP_MUX 0x1D +#define DA732X_REG_HP_DET 0x20 +#define DA732X_REG_HPL_DAC_OFFSET 0x21 +#define DA732X_REG_HPL_DAC_OFF_CNTL 0x22 +#define DA732X_REG_HPL_OUT_OFFSET 0x23 +#define DA732X_REG_HPL 0x24 +#define DA732X_REG_HPL_VOL 0x25 +#define DA732X_REG_HPR_DAC_OFFSET 0x26 +#define DA732X_REG_HPR_DAC_OFF_CNTL 0x27 +#define DA732X_REG_HPR_OUT_OFFSET 0x28 +#define DA732X_REG_HPR 0x29 +#define DA732X_REG_HPR_VOL 0x2A +#define DA732X_REG_LIN2 0x2B +#define DA732X_REG_LIN3 0x2C +#define DA732X_REG_LIN4 0x2D +#define DA732X_REG_OUT_ZC_EN 0x2E +#define DA732X_REG_HP_LIN1_GNDSEL 0x37 +#define DA732X_REG_CP_HP1 0x3A +#define DA732X_REG_CP_HP2 0x3B +#define DA732X_REG_CP_CTRL1 0x40 +#define DA732X_REG_CP_CTRL2 0x41 +#define DA732X_REG_CP_CTRL3 0x42 +#define DA732X_REG_CP_LEVEL_MASK 0x43 +#define DA732X_REG_CP_DET 0x44 +#define DA732X_REG_CP_STATUS 0x45 +#define DA732X_REG_CP_THRESH1 0x46 +#define DA732X_REG_CP_THRESH2 0x47 +#define DA732X_REG_CP_THRESH3 0x48 +#define DA732X_REG_CP_THRESH4 0x49 +#define DA732X_REG_CP_THRESH5 0x4A +#define DA732X_REG_CP_THRESH6 0x4B +#define DA732X_REG_CP_THRESH7 0x4C +#define DA732X_REG_CP_THRESH8 0x4D +#define DA732X_REG_PLL_DIV_LO 0x50 +#define DA732X_REG_PLL_DIV_MID 0x51 +#define DA732X_REG_PLL_DIV_HI 0x52 +#define DA732X_REG_PLL_CTRL 0x53 +#define DA732X_REG_CLK_CTRL 0x54 +#define DA732X_REG_CLK_DSP 0x5A +#define DA732X_REG_CLK_EN1 0x5B +#define DA732X_REG_CLK_EN2 0x5C +#define DA732X_REG_CLK_EN3 0x5D +#define DA732X_REG_CLK_EN4 0x5E +#define DA732X_REG_CLK_EN5 0x5F +#define DA732X_REG_AIF_MCLK 0x60 +#define DA732X_REG_AIFA1 0x61 +#define DA732X_REG_AIFA2 0x62 +#define DA732X_REG_AIFA3 0x63 +#define DA732X_REG_AIFB1 0x64 +#define DA732X_REG_AIFB2 0x65 +#define DA732X_REG_AIFB3 0x66 +#define DA732X_REG_PC_CTRL 0x6A +#define DA732X_REG_DATA_ROUTE 0x70 +#define DA732X_REG_DSP_CTRL 0x71 +#define DA732X_REG_CIF_CTRL2 0x74 +#define DA732X_REG_HANDSHAKE 0x75 +#define DA732X_REG_MBOX0 0x76 +#define DA732X_REG_MBOX1 0x77 +#define DA732X_REG_MBOX2 0x78 +#define DA732X_REG_MBOX_STATUS 0x79 +#define DA732X_REG_SPARE1_OUT 0x7D +#define DA732X_REG_SPARE2_OUT 0x7E +#define DA732X_REG_SPARE1_IN 0x7F +#define DA732X_REG_ID 0x81 +#define DA732X_REG_ADC1_PD 0x90 +#define DA732X_REG_ADC1_HPF 0x93 +#define DA732X_REG_ADC1_SEL 0x94 +#define DA732X_REG_ADC1_EQ12 0x95 +#define DA732X_REG_ADC1_EQ34 0x96 +#define DA732X_REG_ADC1_EQ5 0x97 +#define DA732X_REG_ADC2_PD 0x98 +#define DA732X_REG_ADC2_HPF 0x9B +#define DA732X_REG_ADC2_SEL 0x9C +#define DA732X_REG_ADC2_EQ12 0x9D +#define DA732X_REG_ADC2_EQ34 0x9E +#define DA732X_REG_ADC2_EQ5 0x9F +#define DA732X_REG_DAC1_HPF 0xA0 +#define DA732X_REG_DAC1_L_VOL 0xA1 +#define DA732X_REG_DAC1_R_VOL 0xA2 +#define DA732X_REG_DAC1_SEL 0xA3 +#define DA732X_REG_DAC1_SOFTMUTE 0xA4 +#define DA732X_REG_DAC1_EQ12 0xA5 +#define DA732X_REG_DAC1_EQ34 0xA6 +#define DA732X_REG_DAC1_EQ5 0xA7 +#define DA732X_REG_DAC2_HPF 0xB0 +#define DA732X_REG_DAC2_L_VOL 0xB1 +#define DA732X_REG_DAC2_R_VOL 0xB2 +#define DA732X_REG_DAC2_SEL 0xB3 +#define DA732X_REG_DAC2_SOFTMUTE 0xB4 +#define DA732X_REG_DAC2_EQ12 0xB5 +#define DA732X_REG_DAC2_EQ34 0xB6 +#define DA732X_REG_DAC2_EQ5 0xB7 +#define DA732X_REG_DAC3_HPF 0xC0 +#define DA732X_REG_DAC3_VOL 0xC1 +#define DA732X_REG_DAC3_SEL 0xC3 +#define DA732X_REG_DAC3_SOFTMUTE 0xC4 +#define DA732X_REG_DAC3_EQ12 0xC5 +#define DA732X_REG_DAC3_EQ34 0xC6 +#define DA732X_REG_DAC3_EQ5 0xC7 +#define DA732X_REG_BIQ_BYP 0xD2 +#define DA732X_REG_DMA_CMD 0xD3 +#define DA732X_REG_DMA_ADDR0 0xD4 +#define DA732X_REG_DMA_ADDR1 0xD5 +#define DA732X_REG_DMA_DATA0 0xD6 +#define DA732X_REG_DMA_DATA1 0xD7 +#define DA732X_REG_DMA_DATA2 0xD8 +#define DA732X_REG_DMA_DATA3 0xD9 +#define DA732X_REG_DMA_STATUS 0xDA +#define DA732X_REG_BROWNOUT 0xDF +#define DA732X_REG_UNLOCK 0xE0 + +#define DA732X_MAX_REG DA732X_REG_UNLOCK +/* + * Bits + */ + +/* DA732X_REG_STATUS_EXT (addr=0x00) */ +#define DA732X_STATUS_EXT_DSP (1 << 4) +#define DA732X_STATUS_EXT_CLEAR (0 << 0) + +/* DA732X_REG_STATUS (addr=0x01) */ +#define DA732X_STATUS_PLL_LOCK (1 << 0) +#define DA732X_STATUS_PLL_MCLK_DET (1 << 1) +#define DA732X_STATUS_HPDET_OUT (1 << 2) +#define DA732X_STATUS_INP_MIXDET_1 (1 << 3) +#define DA732X_STATUS_INP_MIXDET_2 (1 << 4) +#define DA732X_STATUS_BO_STATUS (1 << 5) + +/* DA732X_REG_REF1 (addr=0x02) */ +#define DA732X_VMID_FASTCHG (1 << 1) +#define DA732X_VMID_FASTDISCHG (1 << 2) +#define DA732X_REFBUFX2_EN (1 << 6) +#define DA732X_REFBUFX2_DIS (0 << 6) + +/* DA732X_REG_BIAS_EN (addr=0x03) */ +#define DA732X_BIAS_BOOST_MASK (3 << 0) +#define DA732X_BIAS_BOOST_100PC (0 << 0) +#define DA732X_BIAS_BOOST_133PC (1 << 0) +#define DA732X_BIAS_BOOST_88PC (2 << 0) +#define DA732X_BIAS_BOOST_50PC (3 << 0) +#define DA732X_BIAS_EN (1 << 7) +#define DA732X_BIAS_DIS (0 << 7) + +/* DA732X_REG_BIAS1 (addr=0x04) */ +#define DA732X_BIAS1_HP_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS1_HP_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS1_HP_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS1_HP_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS2 (addr=0x05) */ +#define DA732X_BIAS2_LINE2_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS2_LINE2_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS2_LINE2_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS2_LINE2_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS3 (addr=0x06) */ +#define DA732X_BIAS3_LINE3_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS3_LINE3_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS3_LINE3_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS3_LINE3_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_BIAS4 (addr=0x07) */ +#define DA732X_BIAS4_LINE4_DAC_BIAS_MASK (3 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_100PC (0 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_150PC (1 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_50PC (2 << 0) +#define DA732X_BIAS4_LINE4_DAC_BIAS_75PC (3 << 0) +#define DA732X_BIAS4_LINE4_OUT_BIAS_MASK (7 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_100PC (0 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_125PC (1 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_150PC (2 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_175PC (3 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_200PC (4 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_250PC (5 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_300PC (6 << 4) +#define DA732X_BIAS4_LINE4_OUT_BIAS_350PC (7 << 4) + +/* DA732X_REG_SIF_VDD_SEL (addr=0x08) */ +#define DA732X_SIF_VDD_SEL_AIFA_VDD2 (1 << 0) +#define DA732X_SIF_VDD_SEL_AIFB_VDD2 (1 << 1) +#define DA732X_SIF_VDD_SEL_CIFA_VDD2 (1 << 4) + +/* DA732X_REG_MICBIAS2/1 (addr=0x0F/0x10) */ +#define DA732X_MICBIAS_VOLTAGE_MASK (0x0F << 0) +#define DA732X_MICBIAS_VOLTAGE_2V (0x00 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V05 (0x01 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V1 (0x02 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V15 (0x03 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V2 (0x04 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V25 (0x05 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V3 (0x06 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V35 (0x07 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V4 (0x08 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V45 (0x09 << 0) +#define DA732X_MICBIAS_VOLTAGE_2V5 (0x0A << 0) +#define DA732X_MICBIAS_EN (1 << 7) +#define DA732X_MICBIAS_EN_SHIFT 7 +#define DA732X_MICBIAS_VOLTAGE_SHIFT 0 +#define DA732X_MICBIAS_VOLTAGE_MAX 0x0B + +/* DA732X_REG_MICDET (addr=0x11) */ +#define DA732X_MICDET_INP_MICRES (1 << 0) +#define DA732X_MICDET_INP_MICHOOK (1 << 1) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_8MS (0 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_16MS (1 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_32MS (2 << 0) +#define DA732X_MICDET_INP_DEBOUNCE_PRD_64MS (3 << 0) +#define DA732X_MICDET_INP_MICDET_EN (1 << 7) + +/* DA732X_REG_MIC1/2/3_PRE (addr=0x11/0x14/0x18) */ +#define DA732X_MICBOOST_MASK 0x7 +#define DA732X_MICBOOST_SHIFT 0 +#define DA732X_MICBOOST_MIN 0x1 +#define DA732X_MICBOOST_MAX DA732X_MICBOOST_MASK + +/* DA732X_REG_MIC1/2/3 (addr=0x13/0x15/0x19) */ +#define DA732X_MIC_VOL_SHIFT 0 +#define DA732X_MIC_VOL_VAL_MASK 0x1F +#define DA732X_MIC_MUTE_SHIFT 6 +#define DA732X_MIC_EN_SHIFT 7 +#define DA732X_MIC_VOL_VAL_MIN 0x7 +#define DA732X_MIC_VOL_VAL_MAX DA732X_MIC_VOL_VAL_MASK + +/* DA732X_REG_AUX1L/R (addr=0x16/0x17) */ +#define DA732X_AUX_VOL_SHIFT 0 +#define DA732X_AUX_VOL_MASK 0x7 +#define DA732X_AUX_MUTE_SHIFT 6 +#define DA732X_AUX_EN_SHIFT 7 +#define DA732X_AUX_VOL_VAL_MAX DA732X_AUX_VOL_MASK + +/* DA732X_REG_INP_PINBIAS (addr=0x1A) */ +#define DA732X_INP_MICL_PINBIAS_EN (1 << 0) +#define DA732X_INP_MICR_PINBIAS_EN (1 << 1) +#define DA732X_INP_AUX1L_PINBIAS_EN (1 << 2) +#define DA732X_INP_AUX1R_PINBIAS_EN (1 << 3) +#define DA732X_INP_AUX2_PINBIAS_EN (1 << 4) + +/* DA732X_REG_INP_ZC_EN (addr=0x1B) */ +#define DA732X_MIC1_PRE_ZC_EN (1 << 0) +#define DA732X_MIC1_ZC_EN (1 << 1) +#define DA732X_MIC2_PRE_ZC_EN (1 << 2) +#define DA732X_MIC2_ZC_EN (1 << 3) +#define DA732X_AUXL_ZC_EN (1 << 4) +#define DA732X_AUXR_ZC_EN (1 << 5) +#define DA732X_MIC3_PRE_ZC_EN (1 << 6) +#define DA732X_MIC3_ZC_EN (1 << 7) + +/* DA732X_REG_INP_MUX (addr=0x1D) */ +#define DA732X_INP_ADC1L_MUX_SEL_AUX1L (0 << 0) +#define DA732X_INP_ADC1L_MUX_SEL_MIC1 (1 << 0) +#define DA732X_INP_ADC1R_MUX_SEL_MASK (3 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_AUX1R (0 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC2 (1 << 2) +#define DA732X_INP_ADC1R_MUX_SEL_MIC3 (2 << 2) +#define DA732X_INP_ADC2L_MUX_SEL_AUX1L (0 << 4) +#define DA732X_INP_ADC2L_MUX_SEL_MICL (1 << 4) +#define DA732X_INP_ADC2R_MUX_SEL_MASK (3 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX1R (0 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_MICR (1 << 6) +#define DA732X_INP_ADC2R_MUX_SEL_AUX2 (2 << 6) +#define DA732X_ADC1L_MUX_SEL_SHIFT 0 +#define DA732X_ADC1R_MUX_SEL_SHIFT 2 +#define DA732X_ADC2L_MUX_SEL_SHIFT 4 +#define DA732X_ADC2R_MUX_SEL_SHIFT 6 + +/* DA732X_REG_HP_DET (addr=0x20) */ +#define DA732X_HP_DET_AZ (1 << 0) +#define DA732X_HP_DET_SEL1 (1 << 1) +#define DA732X_HP_DET_IS_MASK (3 << 2) +#define DA732X_HP_DET_IS_0_5UA (0 << 2) +#define DA732X_HP_DET_IS_1UA (1 << 2) +#define DA732X_HP_DET_IS_2UA (2 << 2) +#define DA732X_HP_DET_IS_4UA (3 << 2) +#define DA732X_HP_DET_RS_MASK (3 << 4) +#define DA732X_HP_DET_RS_INFINITE (0 << 4) +#define DA732X_HP_DET_RS_100KOHM (1 << 4) +#define DA732X_HP_DET_RS_10KOHM (2 << 4) +#define DA732X_HP_DET_RS_1KOHM (3 << 4) +#define DA732X_HP_DET_EN (1 << 7) + +/* DA732X_REG_HPL_DAC_OFFSET (addr=0x21/0x26) */ +#define DA732X_HP_DAC_OFFSET_TRIM_MASK (0x3F << 0) +#define DA732X_HP_DAC_OFFSET_DAC_SIGN (1 << 6) + +/* DA732X_REG_HPL_DAC_OFF_CNTL (addr=0x22/0x27) */ +#define DA732X_HP_DAC_OFF_CNTL_CONT_MASK (7 << 0) +#define DA732X_HP_DAC_OFF_CNTL_COMPO (1 << 3) +#define DA732X_HP_DAC_OFF_CALIBRATION (1 << 0) +#define DA732X_HP_DAC_OFF_SCALE_STEPS (1 << 1) +#define DA732X_HP_DAC_OFF_MASK 0x7F +#define DA732X_HP_DAC_COMPO_SHIFT 3 + +/* DA732X_REG_HPL_OUT_OFFSET (addr=0x23/0x28) */ +#define DA732X_HP_OUT_OFFSET_MASK (0xFF << 0) +#define DA732X_HP_DAC_OFFSET_TRIM_VAL 0x7F + +/* DA732X_REG_HPL/R (addr=0x24/0x29) */ +#define DA732X_HP_OUT_SIGN (1 << 0) +#define DA732X_HP_OUT_COMP (1 << 1) +#define DA732X_HP_OUT_RESERVED (1 << 2) +#define DA732X_HP_OUT_COMPO (1 << 3) +#define DA732X_HP_OUT_DAC_EN (1 << 4) +#define DA732X_HP_OUT_HIZ_EN (1 << 5) +#define DA732X_HP_OUT_HIZ_DIS (0 << 5) +#define DA732X_HP_OUT_MUTE (1 << 6) +#define DA732X_HP_OUT_EN (1 << 7) +#define DA732X_HP_OUT_COMPO_SHIFT 3 +#define DA732X_HP_OUT_DAC_EN_SHIFT 4 +#define DA732X_HP_HIZ_SHIFT 5 +#define DA732X_HP_MUTE_SHIFT 6 +#define DA732X_HP_OUT_EN_SHIFT 7 + +#define DA732X_OUT_HIZ_EN (1 << 5) +#define DA732X_OUT_HIZ_DIS (0 << 5) + +/* DA732X_REG_HPL/R_VOL (addr=0x25/0x2A) */ +#define DA732X_HP_VOL_VAL_MASK 0xF +#define DA732X_HP_VOL_SHIFT 0 +#define DA732X_HP_VOL_VAL_MAX DA732X_HP_VOL_VAL_MASK + +/* DA732X_REG_LIN2/3/4 (addr=0x2B/0x2C/0x2D) */ +#define DA732X_LOUT_VOL_SHIFT 0 +#define DA732X_LOUT_VOL_MASK 0x0F +#define DA732X_LOUT_DAC_OFF (0 << 4) +#define DA732X_LOUT_DAC_EN (1 << 4) +#define DA732X_LOUT_HIZ_N_DIS (0 << 5) +#define DA732X_LOUT_HIZ_N_EN (1 << 5) +#define DA732X_LOUT_UNMUTED (0 << 6) +#define DA732X_LOUT_MUTED (1 << 6) +#define DA732X_LOUT_EN (0 << 7) +#define DA732X_LOUT_DIS (1 << 7) +#define DA732X_LOUT_DAC_EN_SHIFT 4 +#define DA732X_LOUT_MUTE_SHIFT 6 +#define DA732X_LIN_OUT_EN_SHIFT 7 +#define DA732X_LOUT_VOL_VAL_MAX DA732X_LOUT_VOL_MASK + +/* DA732X_REG_OUT_ZC_EN (addr=0x2E) */ +#define DA732X_HPL_ZC_EN_SHIFT 0 +#define DA732X_HPR_ZC_EN_SHIFT 1 +#define DA732X_HPL_ZC_EN (1 << 0) +#define DA732X_HPL_ZC_DIS (0 << 0) +#define DA732X_HPR_ZC_EN (1 << 1) +#define DA732X_HPR_ZC_DIS (0 << 1) +#define DA732X_LIN2_ZC_EN (1 << 2) +#define DA732X_LIN2_ZC_DIS (0 << 2) +#define DA732X_LIN3_ZC_EN (1 << 3) +#define DA732X_LIN3_ZC_DIS (0 << 3) +#define DA732X_LIN4_ZC_EN (1 << 4) +#define DA732X_LIN4_ZC_DIS (0 << 4) + +/* DA732X_REG_HP_LIN1_GNDSEL (addr=0x37) */ +#define DA732X_HP_OUT_GNDSEL (1 << 0) + +/* DA732X_REG_CP_HP2 (addr=0x3a) */ +#define DA732X_HP_CP_PULSESKIP (1 << 0) +#define DA732X_HP_CP_REG (1 << 1) +#define DA732X_HP_CP_EN (1 << 3) +#define DA732X_HP_CP_DIS (0 << 3) + +/* DA732X_REG_CP_CTRL1 (addr=0x40) */ +#define DA732X_CP_MODE_MASK (7 << 1) +#define DA732X_CP_CTRL_STANDBY (0 << 1) +#define DA732X_CP_CTRL_CPVDD6 (2 << 1) +#define DA732X_CP_CTRL_CPVDD5 (3 << 1) +#define DA732X_CP_CTRL_CPVDD4 (4 << 1) +#define DA732X_CP_CTRL_CPVDD3 (5 << 1) +#define DA732X_CP_CTRL_CPVDD2 (6 << 1) +#define DA732X_CP_CTRL_CPVDD1 (7 << 1) +#define DA723X_CP_DIS (0 << 7) +#define DA732X_CP_EN (1 << 7) + +/* DA732X_REG_CP_CTRL2 (addr=0x41) */ +#define DA732X_CP_BOOST (1 << 0) +#define DA732X_CP_MANAGE_MAGNITUDE (2 << 2) + +/* DA732X_REG_CP_CTRL3 (addr=0x42) */ +#define DA732X_CP_1MHZ (0 << 0) +#define DA732X_CP_500KHZ (1 << 0) +#define DA732X_CP_250KHZ (2 << 0) +#define DA732X_CP_125KHZ (3 << 0) +#define DA732X_CP_63KHZ (4 << 0) +#define DA732X_CP_0KHZ (5 << 0) + +/* DA732X_REG_PLL_CTRL (addr=0x53) */ +#define DA732X_PLL_INDIV_MASK (3 << 0) +#define DA732X_PLL_SRM_EN (1 << 2) +#define DA732X_PLL_EN (1 << 7) +#define DA732X_PLL_BYPASS (0 << 0) + +/* DA732X_REG_CLK_CTRL (addr=0x54) */ +#define DA732X_SR1_MASK (0xF) +#define DA732X_SR2_MASK (0xF0) + +/* DA732X_REG_CLK_DSP (addr=0x5A) */ +#define DA732X_DSP_FREQ_MASK (7 << 0) +#define DA732X_DSP_FREQ_12MHZ (0 << 0) +#define DA732X_DSP_FREQ_24MHZ (1 << 0) +#define DA732X_DSP_FREQ_36MHZ (2 << 0) +#define DA732X_DSP_FREQ_48MHZ (3 << 0) +#define DA732X_DSP_FREQ_60MHZ (4 << 0) +#define DA732X_DSP_FREQ_72MHZ (5 << 0) +#define DA732X_DSP_FREQ_84MHZ (6 << 0) +#define DA732X_DSP_FREQ_96MHZ (7 << 0) + +/* DA732X_REG_CLK_EN1 (addr=0x5B) */ +#define DA732X_DSP_CLK_EN (1 << 0) +#define DA732X_SYS3_CLK_EN (1 << 1) +#define DA732X_DSP12_CLK_EN (1 << 2) +#define DA732X_PC_CLK_EN (1 << 3) +#define DA732X_MCLK_SQR_EN (1 << 7) + +/* DA732X_REG_CLK_EN2 (addr=0x5C) */ +#define DA732X_UART_CLK_EN (1 << 1) +#define DA732X_CP_CLK_EN (1 << 2) +#define DA732X_CP_CLK_DIS (0 << 2) + +/* DA732X_REG_CLK_EN3 (addr=0x5D) */ +#define DA732X_ADCA_BB_CLK_EN (1 << 0) +#define DA732X_ADCC_BB_CLK_EN (1 << 4) + +/* DA732X_REG_CLK_EN4 (addr=0x5E) */ +#define DA732X_DACA_BB_CLK_EN (1 << 0) +#define DA732X_DACC_BB_CLK_EN (1 << 4) +#define DA732X_DACA_BB_CLK_SHIFT 0 +#define DA732X_DACC_BB_CLK_SHIFT 4 + +/* DA732X_REG_CLK_EN5 (addr=0x5F) */ +#define DA732X_DACE_BB_CLK_EN (1 << 0) +#define DA732X_DACE_BB_CLK_SHIFT 0 + +/* DA732X_REG_AIF_MCLK (addr=0x60) */ +#define DA732X_AIFM_FRAME_64 (1 << 2) +#define DA732X_AIFM_SRC_SEL_AIFA (1 << 6) +#define DA732X_CLK_GENERATION_AIF_A (1 << 4) +#define DA732X_NO_CLK_GENERATION 0x0 + +/* DA732X_REG_AIFA1 (addr=0x61) */ +#define DA732X_AIF_WORD_MASK (0x3 << 0) +#define DA732X_AIF_WORD_16 (0 << 0) +#define DA732X_AIF_WORD_20 (1 << 0) +#define DA732X_AIF_WORD_24 (2 << 0) +#define DA732X_AIF_WORD_32 (3 << 0) +#define DA732X_AIF_TDM_MONO_SHIFT (1 << 6) +#define DA732X_AIF1_CLK_MASK (1 << 7) +#define DA732X_AIF_SLAVE (0 << 7) +#define DA732X_AIF_CLK_FROM_SRC (1 << 7) + +/* DA732X_REG_AIFA3 (addr=0x63) */ +#define DA732X_AIF_MODE_SHIFT 0 +#define DA732X_AIF_MODE_MASK 0x3 +#define DA732X_AIF_I2S_MODE (0 << 0) +#define DA732X_AIF_LEFT_J_MODE (1 << 0) +#define DA732X_AIF_RIGHT_J_MODE (2 << 0) +#define DA732X_AIF_DSP_MODE (3 << 0) +#define DA732X_AIF_WCLK_INV (1 << 4) +#define DA732X_AIF_BCLK_INV (1 << 5) +#define DA732X_AIF_EN (1 << 7) +#define DA732X_AIF_EN_SHIFT 7 + +/* DA732X_REG_PC_CTRL (addr=0x6a) */ +#define DA732X_PC_PULSE_AIFA (0 << 0) +#define DA732X_PC_PULSE_AIFB (1 << 0) +#define DA732X_PC_RESYNC_AUT (1 << 6) +#define DA732X_PC_RESYNC_NOT_AUT (0 << 6) +#define DA732X_PC_SAME (1 << 7) + +/* DA732X_REG_DATA_ROUTE (addr=0x70) */ +#define DA732X_ADC1_TO_AIFA (0 << 0) +#define DA732X_DSP_TO_AIFA (1 << 0) +#define DA732X_ADC2_TO_AIFB (0 << 1) +#define DA732X_DSP_TO_AIFB (1 << 1) +#define DA732X_AIFA_TO_DAC1L (0 << 2) +#define DA732X_DSP_TO_DAC1L (1 << 2) +#define DA732X_AIFA_TO_DAC1R (0 << 3) +#define DA732X_DSP_TO_DAC1R (1 << 3) +#define DA732X_AIFB_TO_DAC2L (0 << 4) +#define DA732X_DSP_TO_DAC2L (1 << 4) +#define DA732X_AIFB_TO_DAC2R (0 << 5) +#define DA732X_DSP_TO_DAC2R (1 << 5) +#define DA732X_AIFB_TO_DAC3 (0 << 6) +#define DA732X_DSP_TO_DAC3 (1 << 6) +#define DA732X_BYPASS_DSP (0 << 0) +#define DA732X_ALL_TO_DSP (0x7F << 0) + +/* DA732X_REG_DSP_CTRL (addr=0x71) */ +#define DA732X_DIGITAL_EN (1 << 0) +#define DA732X_DIGITAL_RESET (0 << 0) +#define DA732X_DSP_CORE_EN (1 << 1) +#define DA732X_DSP_CORE_RESET (0 << 1) + +/* DA732X_REG_SPARE1_OUT (addr=0x7D)*/ +#define DA732X_HP_DRIVER_EN (1 << 0) +#define DA732X_HP_GATE_LOW (1 << 2) +#define DA732X_HP_LOOP_GAIN_CTRL (1 << 3) + +/* DA732X_REG_ID (addr=0x81)*/ +#define DA732X_ID_MINOR_MASK (0xF << 0) +#define DA732X_ID_MAJOR_MASK (0xF << 4) + +/* DA732X_REG_ADC1/2_PD (addr=0x90/0x98) */ +#define DA732X_ADC_RST_MASK (0x3 << 0) +#define DA732X_ADC_PD_MASK (0x3 << 2) +#define DA732X_ADC_SET_ACT (0x3 << 0) +#define DA732X_ADC_SET_RST (0x0 << 0) +#define DA732X_ADC_ON (0x3 << 2) +#define DA732X_ADC_OFF (0x0 << 2) + +/* DA732X_REG_ADC1/2_SEL (addr=0x94/0x9C) */ +#define DA732X_ADC_VOL_VAL_MASK 0x7 +#define DA732X_ADCL_VOL_SHIFT 0 +#define DA732X_ADCR_VOL_SHIFT 4 +#define DA732X_ADCL_EN_SHIFT 2 +#define DA732X_ADCR_EN_SHIFT 3 +#define DA732X_ADCL_EN (1 << 2) +#define DA732X_ADCR_EN (1 << 3) +#define DA732X_ADC_VOL_VAL_MAX DA732X_ADC_VOL_VAL_MASK + +/* + * DA732X_REG_ADC1/2_HPF (addr=0x93/0x9b) + * DA732x_REG_DAC1/2/3_HPG (addr=0xA5/0xB5/0xC5) + */ +#define DA732X_HPF_MUSIC_EN (1 << 3) +#define DA732X_HPF_VOICE_EN ((1 << 3) | (1 << 7)) +#define DA732X_HPF_MASK ((1 << 3) | (1 << 7)) +#define DA732X_HPF_DIS ((0 << 3) | (0 << 7)) + +/* DA732X_REG_DAC1/2/3_VOL */ +#define DA732X_DAC_VOL_VAL_MASK 0x7F +#define DA732X_DAC_VOL_SHIFT 0 +#define DA732X_DAC_VOL_VAL_MAX DA732X_DAC_VOL_VAL_MASK + +/* DA732X_REG_DAC1/2/3_SEL (addr=0xA3/0xB3/0xC3) */ +#define DA732X_DACL_EN_SHIFT 3 +#define DA732X_DACR_EN_SHIFT 7 +#define DA732X_DACL_MUTE_SHIFT 2 +#define DA732X_DACR_MUTE_SHIFT 6 +#define DA732X_DACL_EN (1 << 3) +#define DA732X_DACR_EN (1 << 7) +#define DA732X_DACL_SDM (1 << 0) +#define DA732X_DACR_SDM (1 << 4) +#define DA732X_DACL_MUTE (1 << 2) +#define DA732X_DACR_MUTE (1 << 6) + +/* DA732X_REG_DAC_SOFTMUTE (addr=0xA4/0xB4/0xC4) */ +#define DA732X_SOFTMUTE_EN (1 << 7) +#define DA732X_GAIN_RAMPED (1 << 6) +#define DA732X_16_SAMPLES (4 << 0) +#define DA732X_SOFTMUTE_MASK (1 << 7) +#define DA732X_SOFTMUTE_SHIFT 7 + +/* + * DA732x_REG_ADC1/2_EQ12 (addr=0x95/0x9D) + * DA732x_REG_ADC1/2_EQ34 (addr=0x96/0x9E) + * DA732x_REG_ADC1/2_EQ5 (addr=0x97/0x9F) + * DA732x_REG_DAC1/2/3_EQ12 (addr=0xA5/0xB5/0xC5) + * DA732x_REG_DAC1/2/3_EQ34 (addr=0xA6/0xB6/0xC6) + * DA732x_REG_DAC1/2/3_EQ5 (addr=0xA7/0xB7/0xB7) + */ +#define DA732X_EQ_VOL_VAL_MASK 0xF +#define DA732X_EQ_BAND1_SHIFT 0 +#define DA732X_EQ_BAND2_SHIFT 4 +#define DA732X_EQ_BAND3_SHIFT 0 +#define DA732X_EQ_BAND4_SHIFT 4 +#define DA732X_EQ_BAND5_SHIFT 0 +#define DA732X_EQ_OVERALL_SHIFT 4 +#define DA732X_EQ_OVERALL_VOL_VAL_MASK 0x3 +#define DA732X_EQ_DIS (0 << 7) +#define DA732X_EQ_EN (1 << 7) +#define DA732X_EQ_EN_SHIFT 7 +#define DA732X_EQ_VOL_VAL_MAX DA732X_EQ_VOL_VAL_MASK +#define DA732X_EQ_OVERALL_VOL_VAL_MAX DA732X_EQ_OVERALL_VOL_VAL_MASK + +/* DA732X_REG_DMA_CMD (addr=0xD3) */ +#define DA732X_SEL_DSP_DMA_MASK (3 << 0) +#define DA732X_SEL_DSP_DMA_DIS (0 << 0) +#define DA732X_SEL_DSP_DMA_PMEM (1 << 0) +#define DA732X_SEL_DSP_DMA_XMEM (2 << 0) +#define DA732X_SEL_DSP_DMA_YMEM (3 << 0) +#define DA732X_DSP_RW_MASK (1 << 4) +#define DA732X_DSP_DMA_WRITE (0 << 4) +#define DA732X_DSP_DMA_READ (1 << 4) + +/* DA732X_REG_DMA_STATUS (addr=0xDA) */ +#define DA732X_DSP_DMA_FREE (0 << 0) +#define DA732X_DSP_DMA_BUSY (1 << 0) + +#endif /* __DA732X_REG_H_ */ -- cgit v0.10.2 From 20c5fd399482ef5b87a41ab064b3255f1faaaee4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 10:03:20 +0800 Subject: ASoC: wm8903: Move pin configuration into I2C probe() function Ensure that the device pins are configured as soon as possible by moving the pin configration (including MICBIAS) into the I2C probe() function. This had been done in the CODEC probe() function when we were relying on the ASoC register I/O code. Signed-off-by: Mark Brown Tested-by: Stephen Warren diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 3abd450..64ca904 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1,7 +1,7 @@ /* * wm8903.c -- WM8903 ALSA SoC Audio driver * - * Copyright 2008-11 Wolfson Microelectronics + * Copyright 2008-12 Wolfson Microelectronics * Copyright 2011-2012 NVIDIA, Inc. * * Author: Mark Brown @@ -1880,10 +1880,9 @@ static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); struct wm8903_platform_data *pdata = wm8903->pdata; - int ret, i; + int ret; int trigger, irq_pol; u16 val; - bool mic_gpio = false; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -1894,47 +1893,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - /* Set up GPIOs, detect if any are MIC detect outputs */ - for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { - if ((!pdata->gpio_cfg[i]) || - (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) - continue; - - snd_soc_write(codec, WM8903_GPIO_CONTROL_1 + i, - pdata->gpio_cfg[i] & 0x7fff); - - val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) - >> WM8903_GP1_FN_SHIFT; - - switch (val) { - case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: - case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: - mic_gpio = true; - break; - default: - break; - } - } - - /* Set up microphone detection */ - snd_soc_write(codec, WM8903_MIC_BIAS_CONTROL_0, - pdata->micdet_cfg); - - /* Microphone detection needs the WSEQ clock */ - if (pdata->micdet_cfg) - snd_soc_update_bits(codec, WM8903_WRITE_SEQUENCER_0, - WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); - - /* If microphone detection is enabled by pdata but - * detected via IRQ then interrupts can be lost before - * the machine driver has set up microphone detection - * IRQs as the IRQs are clear on read. The detection - * will be enabled when the machine driver configures. - */ - WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); - - wm8903->mic_delay = pdata->micdet_delay; - if (wm8903->irq) { if (pdata->irq_active_low) { trigger = IRQF_TRIGGER_LOW; @@ -2115,8 +2073,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; + bool mic_gpio = false; unsigned int val; - int ret; + int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), GFP_KERNEL); @@ -2160,6 +2119,8 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } } + pdata = wm8903->pdata; + ret = regmap_read(wm8903->regmap, WM8903_SW_RESET_AND_ID, &val); if (ret != 0) { dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); @@ -2184,6 +2145,47 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903_init_gpio(wm8903); + /* Set up GPIO pin state, detect if any are MIC detect outputs */ + for (i = 0; i < ARRAY_SIZE(pdata->gpio_cfg); i++) { + if ((!pdata->gpio_cfg[i]) || + (pdata->gpio_cfg[i] > WM8903_GPIO_CONFIG_ZERO)) + continue; + + regmap_write(wm8903->regmap, WM8903_GPIO_CONTROL_1 + i, + pdata->gpio_cfg[i] & 0x7fff); + + val = (pdata->gpio_cfg[i] & WM8903_GP1_FN_MASK) + >> WM8903_GP1_FN_SHIFT; + + switch (val) { + case WM8903_GPn_FN_MICBIAS_CURRENT_DETECT: + case WM8903_GPn_FN_MICBIAS_SHORT_DETECT: + mic_gpio = true; + break; + default: + break; + } + } + + /* Set up microphone detection */ + regmap_write(wm8903->regmap, WM8903_MIC_BIAS_CONTROL_0, + pdata->micdet_cfg); + + /* Microphone detection needs the WSEQ clock */ + if (pdata->micdet_cfg) + regmap_update_bits(wm8903->regmap, WM8903_WRITE_SEQUENCER_0, + WM8903_WSEQ_ENA, WM8903_WSEQ_ENA); + + /* If microphone detection is enabled by pdata but + * detected via IRQ then interrupts can be lost before + * the machine driver has set up microphone detection + * IRQs as the IRQs are clear on read. The detection + * will be enabled when the machine driver configures. + */ + WARN_ON(!mic_gpio && (pdata->micdet_cfg & WM8903_MICDET_ENA)); + + wm8903->mic_delay = pdata->micdet_delay; + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) -- cgit v0.10.2 From e373cbfb2f7d194e48d528794b3b99274d4c1a97 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 10:06:11 +0800 Subject: ASoC: wm8903: Make interrupt handler use regmap directly There's no urgent need for the interrupt handler to use the ASoC I/O functions and it'll support a further move in where we request the interrupt so call the regmap APIs directly. Signed-off-by: Mark Brown Tested-by: Stephen Warren diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 64ca904..f5d47c8 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1636,17 +1636,27 @@ EXPORT_SYMBOL_GPL(wm8903_mic_detect); static irqreturn_t wm8903_irq(int irq, void *data) { - struct snd_soc_codec *codec = data; - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int mic_report; - int int_pol; - int int_val = 0; - int mask = ~snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1_MASK); + struct wm8903_priv *wm8903 = data; + int mic_report, ret; + unsigned int int_val, mask, int_pol; + + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1_MASK, + &mask); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ mask: %d\n", ret); + return IRQ_NONE; + } + + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_STATUS_1, &int_val); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read IRQ status: %d\n", ret); + return IRQ_NONE; + } - int_val = snd_soc_read(codec, WM8903_INTERRUPT_STATUS_1) & mask; + int_val &= ~mask; if (int_val & WM8903_WSEQ_BUSY_EINT) { - dev_warn(codec->dev, "Write sequencer done\n"); + dev_warn(wm8903->dev, "Write sequencer done\n"); } /* @@ -1657,22 +1667,28 @@ static irqreturn_t wm8903_irq(int irq, void *data) * the polarity register. */ mic_report = wm8903->mic_last_report; - int_pol = snd_soc_read(codec, WM8903_INTERRUPT_POLARITY_1); + ret = regmap_read(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + &int_pol); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to read interrupt polarity: %d\n", + ret); + return IRQ_HANDLED; + } #ifndef CONFIG_SND_SOC_WM8903_MODULE if (int_val & (WM8903_MICSHRT_EINT | WM8903_MICDET_EINT)) - trace_snd_soc_jack_irq(dev_name(codec->dev)); + trace_snd_soc_jack_irq(dev_name(wm8903->dev)); #endif if (int_val & WM8903_MICSHRT_EINT) { - dev_dbg(codec->dev, "Microphone short (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone short (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_short; int_pol ^= WM8903_MICSHRT_INV; } if (int_val & WM8903_MICDET_EINT) { - dev_dbg(codec->dev, "Microphone detect (pol=%x)\n", int_pol); + dev_dbg(wm8903->dev, "Microphone detect (pol=%x)\n", int_pol); mic_report ^= wm8903->mic_det; int_pol ^= WM8903_MICDET_INV; @@ -1680,8 +1696,8 @@ static irqreturn_t wm8903_irq(int irq, void *data) msleep(wm8903->mic_delay); } - snd_soc_update_bits(codec, WM8903_INTERRUPT_POLARITY_1, - WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_POLARITY_1, + WM8903_MICSHRT_INV | WM8903_MICDET_INV, int_pol); snd_soc_jack_report(wm8903->mic_jack, mic_report, wm8903->mic_short | wm8903->mic_det); @@ -1907,7 +1923,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq, trigger | IRQF_ONESHOT, - "wm8903", codec); + "wm8903", wm8903); if (ret != 0) { dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); @@ -1963,7 +1979,7 @@ static int wm8903_remove(struct snd_soc_codec *codec) wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); if (wm8903->irq) - free_irq(wm8903->irq, codec); + free_irq(wm8903->irq, wm8903); return 0; } -- cgit v0.10.2 From b7c95d9146c8201740e2ce9dca7fb1eb8b7b0053 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 10:15:10 +0800 Subject: ASoC: wm8903: Move interrupt request to I2C probe There's no reason to defer requesting of the interrupt until the CODEC probe and doing so results in more work if we hit an error as we'll have registered the CODEC with the core. It's neater to acquire as many of the resources we'll need as we can in the bus probe function. Signed-off-by: Mark Brown Tested-by: Stephen Warren diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index f5d47c8..7261a68 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1895,9 +1895,7 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - struct wm8903_platform_data *pdata = wm8903->pdata; int ret; - int trigger, irq_pol; u16 val; wm8903->codec = codec; @@ -1909,32 +1907,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) return ret; } - if (wm8903->irq) { - if (pdata->irq_active_low) { - trigger = IRQF_TRIGGER_LOW; - irq_pol = WM8903_IRQ_POL; - } else { - trigger = IRQF_TRIGGER_HIGH; - irq_pol = 0; - } - - snd_soc_update_bits(codec, WM8903_INTERRUPT_CONTROL, - WM8903_IRQ_POL, irq_pol); - - ret = request_threaded_irq(wm8903->irq, NULL, wm8903_irq, - trigger | IRQF_ONESHOT, - "wm8903", wm8903); - if (ret != 0) { - dev_err(codec->dev, "Failed to request IRQ: %d\n", - ret); - return ret; - } - - /* Enable write sequencer interrupts */ - snd_soc_update_bits(codec, WM8903_INTERRUPT_STATUS_1_MASK, - WM8903_IM_WSEQ_BUSY_EINT, 0); - } - /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1975,11 +1947,7 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* power down chip */ static int wm8903_remove(struct snd_soc_codec *codec) { - struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - wm8903_set_bias_level(codec, SND_SOC_BIAS_OFF); - if (wm8903->irq) - free_irq(wm8903->irq, wm8903); return 0; } @@ -2089,8 +2057,9 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, { struct wm8903_platform_data *pdata = dev_get_platdata(&i2c->dev); struct wm8903_priv *wm8903; + int trigger; bool mic_gpio = false; - unsigned int val; + unsigned int val, irq_pol; int ret, i; wm8903 = devm_kzalloc(&i2c->dev, sizeof(struct wm8903_priv), @@ -2108,7 +2077,6 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, wm8903); - wm8903->irq = i2c->irq; /* If no platform data was supplied, create storage for defaults */ if (pdata) { @@ -2202,6 +2170,33 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, wm8903->mic_delay = pdata->micdet_delay; + if (i2c->irq) { + if (pdata->irq_active_low) { + trigger = IRQF_TRIGGER_LOW; + irq_pol = WM8903_IRQ_POL; + } else { + trigger = IRQF_TRIGGER_HIGH; + irq_pol = 0; + } + + regmap_update_bits(wm8903->regmap, WM8903_INTERRUPT_CONTROL, + WM8903_IRQ_POL, irq_pol); + + ret = request_threaded_irq(i2c->irq, NULL, wm8903_irq, + trigger | IRQF_ONESHOT, + "wm8903", wm8903); + if (ret != 0) { + dev_err(wm8903->dev, "Failed to request IRQ: %d\n", + ret); + return ret; + } + + /* Enable write sequencer interrupts */ + regmap_update_bits(wm8903->regmap, + WM8903_INTERRUPT_STATUS_1_MASK, + WM8903_IM_WSEQ_BUSY_EINT, 0); + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) @@ -2216,6 +2211,8 @@ static __devexit int wm8903_i2c_remove(struct i2c_client *client) { struct wm8903_priv *wm8903 = i2c_get_clientdata(client); + if (client->irq) + free_irq(client->irq, wm8903); wm8903_free_gpio(wm8903); snd_soc_unregister_codec(&client->dev); -- cgit v0.10.2 From a89c3e956ae78cec8926b92f2d61b7a5b675e787 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 10:30:34 +0800 Subject: ASoC: wm8903: Move register default changes to I2C probe Also convert to use update_bits() while we're at it. No great need to do this, it's just a bit neater to do as much as possible in the I2C probe. Signed-off-by: Mark Brown Tested-by: Stephen Warren diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 7261a68..73f1c8d 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1896,7 +1896,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); int ret; - u16 val; wm8903->codec = codec; codec->control_data = wm8903->regmap; @@ -1910,37 +1909,6 @@ static int wm8903_probe(struct snd_soc_codec *codec) /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - /* Latch volume update bits */ - val = snd_soc_read(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT); - val |= WM8903_ADCVU; - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_ADC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT); - val |= WM8903_DACVU; - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_LEFT, val); - snd_soc_write(codec, WM8903_DAC_DIGITAL_VOLUME_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT1_LEFT); - val |= WM8903_HPOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT1_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT2_LEFT); - val |= WM8903_LINEOUTVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT2_RIGHT, val); - - val = snd_soc_read(codec, WM8903_ANALOGUE_OUT3_LEFT); - val |= WM8903_SPKVU; - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_LEFT, val); - snd_soc_write(codec, WM8903_ANALOGUE_OUT3_RIGHT, val); - - /* Enable DAC soft mute by default */ - snd_soc_update_bits(codec, WM8903_DAC_DIGITAL_1, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, - WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); - return ret; } @@ -2197,6 +2165,37 @@ static __devinit int wm8903_i2c_probe(struct i2c_client *i2c, WM8903_IM_WSEQ_BUSY_EINT, 0); } + /* Latch volume update bits */ + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_LEFT, + WM8903_ADCVU, WM8903_ADCVU); + regmap_update_bits(wm8903->regmap, WM8903_ADC_DIGITAL_VOLUME_RIGHT, + WM8903_ADCVU, WM8903_ADCVU); + + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_LEFT, + WM8903_DACVU, WM8903_DACVU); + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_VOLUME_RIGHT, + WM8903_DACVU, WM8903_DACVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_LEFT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT1_RIGHT, + WM8903_HPOUTVU, WM8903_HPOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_LEFT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT2_RIGHT, + WM8903_LINEOUTVU, WM8903_LINEOUTVU); + + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_LEFT, + WM8903_SPKVU, WM8903_SPKVU); + regmap_update_bits(wm8903->regmap, WM8903_ANALOGUE_OUT3_RIGHT, + WM8903_SPKVU, WM8903_SPKVU); + + /* Enable DAC soft mute by default */ + regmap_update_bits(wm8903->regmap, WM8903_DAC_DIGITAL_1, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE, + WM8903_DAC_MUTEMODE | WM8903_DAC_MUTE); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8903, &wm8903_dai, 1); if (ret != 0) -- cgit v0.10.2 From f242e50eee1ec7692c4854d94e8cd543991cce71 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Thu, 7 Jun 2012 14:00:46 +0200 Subject: mfd/ab8500: Move platform-data for ab8500-codec into mfd-driver The platform-data used by the Ux500 ASoC-driver is moved from the machine-driver context into the codec-driver context. This means adding the platform-data for 'ab8500-codec' into the main AB8500 platform-data. Signed-off-by: Ola Lilja Signed-off-by: Mark Brown diff --git a/arch/arm/mach-ux500/board-mop500.c b/arch/arm/mach-ux500/board-mop500.c index 9c74ac5..c8a8fde 100644 --- a/arch/arm/mach-ux500/board-mop500.c +++ b/arch/arm/mach-ux500/board-mop500.c @@ -25,6 +25,7 @@ #include #include #include +#include #include #include #include @@ -97,6 +98,18 @@ static struct ab8500_gpio_platform_data ab8500_gpio_pdata = { 0x7A, 0x00, 0x00}, }; +/* ab8500-codec */ +static struct ab8500_codec_platform_data ab8500_codec_pdata = { + .amics = { + .mic1_type = AMIC_TYPE_DIFFERENTIAL, + .mic2_type = AMIC_TYPE_DIFFERENTIAL, + .mic1a_micbias = AMIC_MICBIAS_VAMIC1, + .mic1b_micbias = AMIC_MICBIAS_VAMIC1, + .mic2_micbias = AMIC_MICBIAS_VAMIC2 + }, + .ear_cmv = EAR_CMV_0_95V +}; + static struct gpio_keys_button snowball_key_array[] = { { .gpio = 32, @@ -195,6 +208,7 @@ static struct ab8500_platform_data ab8500_platdata = { .regulator = ab8500_regulators, .num_regulator = ARRAY_SIZE(ab8500_regulators), .gpio = &ab8500_gpio_pdata, + .codec = &ab8500_codec_pdata, }; static struct resource ab8500_resources[] = { diff --git a/include/linux/mfd/abx500/ab8500-codec.h b/include/linux/mfd/abx500/ab8500-codec.h new file mode 100644 index 0000000..dc65292 --- /dev/null +++ b/include/linux/mfd/abx500/ab8500-codec.h @@ -0,0 +1,52 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef AB8500_CORE_CODEC_H +#define AB8500_CORE_CODEC_H + +/* Mic-types */ +enum amic_type { + AMIC_TYPE_SINGLE_ENDED, + AMIC_TYPE_DIFFERENTIAL +}; + +/* Mic-biases */ +enum amic_micbias { + AMIC_MICBIAS_VAMIC1, + AMIC_MICBIAS_VAMIC2 +}; + +/* Bias-voltage */ +enum ear_cm_voltage { + EAR_CMV_0_95V, + EAR_CMV_1_10V, + EAR_CMV_1_27V, + EAR_CMV_1_58V +}; + +/* Analog microphone settings */ +struct amic_settings { + enum amic_type mic1_type; + enum amic_type mic2_type; + enum amic_micbias mic1a_micbias; + enum amic_micbias mic1b_micbias; + enum amic_micbias mic2_micbias; +}; + +/* Platform data structure for the audio-parts of the AB8500 */ +struct ab8500_codec_platform_data { + struct amic_settings amics; + enum ear_cm_voltage ear_cmv; +}; + +#endif diff --git a/include/linux/mfd/abx500/ab8500.h b/include/linux/mfd/abx500/ab8500.h index 91dd3ef..bc9b84b 100644 --- a/include/linux/mfd/abx500/ab8500.h +++ b/include/linux/mfd/abx500/ab8500.h @@ -266,6 +266,7 @@ struct ab8500 { struct regulator_reg_init; struct regulator_init_data; struct ab8500_gpio_platform_data; +struct ab8500_codec_platform_data; /** * struct ab8500_platform_data - AB8500 platform data @@ -284,6 +285,7 @@ struct ab8500_platform_data { int num_regulator; struct regulator_init_data *regulator; struct ab8500_gpio_platform_data *gpio; + struct ab8500_codec_platform_data *codec; }; extern int __devinit ab8500_init(struct ab8500 *ab8500, -- cgit v0.10.2 From 85f243912b99b053ce0624c30609f5d8fd4445d2 Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Wed, 13 Jun 2012 10:09:51 +0200 Subject: ASoC: Ux500: Correct license strings GPLv2 -> GPL v2 Reported-by: Stephen Rothwell Signed-off-by: Ola Lilja Acked-by: Linus Walleij Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 95dc7d5..389dd66 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2518,4 +2518,4 @@ static struct platform_driver ab8500_codec_platform_driver = { }; module_platform_driver(ab8500_codec_platform_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 93c6c40..62ac028 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -840,4 +840,4 @@ static struct platform_driver msp_i2s_driver = { }; module_platform_driver(msp_i2s_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 496dec1..ee14d2d 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -739,4 +739,4 @@ void ux500_msp_i2s_cleanup_msp(struct platform_device *pdev, devm_kfree(&pdev->dev, msp); } -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 66b080e..97d8e4d 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -315,4 +315,4 @@ static struct platform_driver ux500_pcm_driver = { }; module_platform_driver(ux500_pcm_driver); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- cgit v0.10.2 From 8994a5e1d2443511e677d62e97d7de3718b71325 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Wed, 13 Jun 2012 14:36:07 +0800 Subject: ASoC: ml26124: Convert to devm_regmap_init_i2c This fixes a leak if snd_soc_register_codec fails. Signed-off-by: Axel Lin Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 22cb5bf..96aa5fa 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -638,7 +638,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, priv); - priv->regmap = regmap_init_i2c(i2c, &ml26124_i2c_regmap); + priv->regmap = devm_regmap_init_i2c(i2c, &ml26124_i2c_regmap); if (IS_ERR(priv->regmap)) { ret = PTR_ERR(priv->regmap); dev_err(&i2c->dev, "regmap_init_i2c() failed: %d\n", ret); @@ -651,10 +651,7 @@ static __devinit int ml26124_i2c_probe(struct i2c_client *i2c, static __devexit int ml26124_i2c_remove(struct i2c_client *client) { - struct ml26124_priv *priv = i2c_get_clientdata(client); - snd_soc_unregister_codec(&client->dev); - regmap_exit(priv->regmap); return 0; } -- cgit v0.10.2 From 7a824e214e25a49442fe868dac0af8a904b24f58 Mon Sep 17 00:00:00 2001 From: Zhangfei Gao Date: Mon, 11 Jun 2012 18:04:38 +0800 Subject: ASoC: mmp: add audio dma support mmp-pcm handle audio dma based on soc-dmaengine Support mmp and pxa910 Signed-off-by: Zhangfei Gao Signed-off-by: Leo Yan Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown diff --git a/include/linux/platform_data/mmp_audio.h b/include/linux/platform_data/mmp_audio.h new file mode 100644 index 0000000..0f25d16 --- /dev/null +++ b/include/linux/platform_data/mmp_audio.h @@ -0,0 +1,22 @@ +/* + * MMP Platform AUDIO Management + * + * Copyright (c) 2011 Marvell Semiconductors Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#ifndef MMP_AUDIO_H +#define MMP_AUDIO_H + +struct mmp_audio_platdata { + u32 period_max_capture; + u32 buffer_max_capture; + u32 period_max_playback; + u32 buffer_max_playback; +}; + +#endif /* MMP_AUDIO_H */ diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index a0f7d3c..5d76e29 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -8,6 +8,15 @@ config SND_PXA2XX_SOC the PXA2xx AC97, I2S or SSP interface. You will also need to select the audio interfaces to support below. +config SND_MMP_SOC + bool "Soc Audio for Marvell MMP chips" + depends on ARCH_MMP + select SND_SOC_DMAENGINE_PCM + select SND_ARM + help + Say Y if you want to add support for codecs attached to + the MMP SSPA interface. + config SND_PXA2XX_AC97 tristate select SND_AC97_CODEC diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index af35762..f913e9b 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -3,11 +3,13 @@ snd-soc-pxa2xx-objs := pxa2xx-pcm.o snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o +snd-soc-mmp-objs := mmp-pcm.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o +obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/mmp-pcm.c b/sound/soc/pxa/mmp-pcm.c new file mode 100644 index 0000000..73ac546 --- /dev/null +++ b/sound/soc/pxa/mmp-pcm.c @@ -0,0 +1,297 @@ +/* + * linux/sound/soc/pxa/mmp-pcm.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +struct mmp_dma_data { + int ssp_id; + struct resource *dma_res; +}; + +#define MMP_PCM_INFO (SNDRV_PCM_INFO_MMAP | \ + SNDRV_PCM_INFO_MMAP_VALID | \ + SNDRV_PCM_INFO_INTERLEAVED | \ + SNDRV_PCM_INFO_PAUSE | \ + SNDRV_PCM_INFO_RESUME) + +#define MMP_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_pcm_hardware mmp_pcm_hardware[] = { + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, + { + .info = MMP_PCM_INFO, + .formats = MMP_PCM_FORMATS, + .period_bytes_min = 1024, + .period_bytes_max = 2048, + .periods_min = 2, + .periods_max = 32, + .buffer_bytes_max = 4096, + .fifo_size = 32, + }, +}; + +static int mmp_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct pxa2xx_pcm_dma_params *dma_params; + struct dma_slave_config slave_config; + int ret; + + dma_params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + if (!dma_params) + return 0; + + ret = snd_hwparams_to_dma_slave_config(substream, params, &slave_config); + if (ret) + return ret; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr = dma_params->dev_addr; + slave_config.dst_maxburst = 4; + } else { + slave_config.src_addr = dma_params->dev_addr; + slave_config.src_maxburst = 4; + } + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret) + return ret; + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static bool filter(struct dma_chan *chan, void *param) +{ + struct mmp_dma_data *dma_data = param; + bool found = false; + char *devname; + + devname = kasprintf(GFP_KERNEL, "%s.%d", dma_data->dma_res->name, + dma_data->ssp_id); + if ((strcmp(dev_name(chan->device->dev), devname) == 0) && + (chan->chan_id == dma_data->dma_res->start)) { + found = true; + } + + kfree(devname); + return found; +} + +static int mmp_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct mmp_dma_data *dma_data; + struct resource *r; + int ret; + + r = platform_get_resource(pdev, IORESOURCE_DMA, substream->stream); + if (!r) + return -EBUSY; + + snd_soc_set_runtime_hwparams(substream, + &mmp_pcm_hardware[substream->stream]); + dma_data = devm_kzalloc(&pdev->dev, + sizeof(struct mmp_dma_data), GFP_KERNEL); + if (dma_data == NULL) + return -ENOMEM; + + dma_data->dma_res = r; + dma_data->ssp_id = cpu_dai->id; + + ret = snd_dmaengine_pcm_open(substream, filter, dma_data); + if (ret) { + devm_kfree(&pdev->dev, dma_data); + return ret; + } + + snd_dmaengine_pcm_set_data(substream, dma_data); + return 0; +} + +static int mmp_pcm_close(struct snd_pcm_substream *substream) +{ + struct mmp_dma_data *dma_data = snd_dmaengine_pcm_get_data(substream); + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct platform_device *pdev = to_platform_device(rtd->platform->dev); + + snd_dmaengine_pcm_close(substream); + devm_kfree(&pdev->dev, dma_data); + return 0; +} + +static int mmp_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long off = vma->vm_pgoff; + + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + return remap_pfn_range(vma, vma->vm_start, + __phys_to_pfn(runtime->dma_addr) + off, + vma->vm_end - vma->vm_start, vma->vm_page_prot); +} + +struct snd_pcm_ops mmp_pcm_ops = { + .open = mmp_pcm_open, + .close = mmp_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = mmp_pcm_hw_params, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = mmp_pcm_mmap, +}; + +static void mmp_pcm_free_dma_buffers(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + struct gen_pool *gpool; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return; + + for (stream = 0; stream < 2; stream++) { + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + gen_pool_free(gpool, (unsigned long)buf->area, size); + buf->area = NULL; + } + + return; +} + +static int mmp_pcm_preallocate_dma_buffer(struct snd_pcm_substream *substream, + int stream) +{ + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = mmp_pcm_hardware[stream].buffer_bytes_max; + struct gen_pool *gpool; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = substream->pcm->card->dev; + buf->private_data = NULL; + + gpool = sram_get_gpool("asram"); + if (!gpool) + return -ENOMEM; + + buf->area = (unsigned char *)gen_pool_alloc(gpool, size); + if (!buf->area) + return -ENOMEM; + buf->addr = gen_pool_virt_to_phys(gpool, (unsigned long)buf->area); + buf->bytes = size; + return 0; +} + +int mmp_pcm_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_pcm_substream *substream; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0, stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + + ret = mmp_pcm_preallocate_dma_buffer(substream, stream); + if (ret) + goto err; + } + + return 0; + +err: + mmp_pcm_free_dma_buffers(pcm); + return ret; +} + +struct snd_soc_platform_driver mmp_soc_platform = { + .ops = &mmp_pcm_ops, + .pcm_new = mmp_pcm_new, + .pcm_free = mmp_pcm_free_dma_buffers, +}; + +static __devinit int mmp_pcm_probe(struct platform_device *pdev) +{ + struct mmp_audio_platdata *pdata = pdev->dev.platform_data; + + if (pdata) { + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].buffer_bytes_max = + pdata->buffer_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_PLAYBACK].period_bytes_max = + pdata->period_max_playback; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].buffer_bytes_max = + pdata->buffer_max_capture; + mmp_pcm_hardware[SNDRV_PCM_STREAM_CAPTURE].period_bytes_max = + pdata->period_max_capture; + } + return snd_soc_register_platform(&pdev->dev, &mmp_soc_platform); +} + +static int __devexit mmp_pcm_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver mmp_pcm_driver = { + .driver = { + .name = "mmp-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = mmp_pcm_probe, + .remove = __devexit_p(mmp_pcm_remove), +}; + +module_platform_driver(mmp_pcm_driver); + +MODULE_AUTHOR("Leo Yan "); +MODULE_DESCRIPTION("MMP Soc Audio DMA module"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From fa375d42f0e531b7ca4316ea9fd5444e01d585e8 Mon Sep 17 00:00:00 2001 From: Zhangfei Gao Date: Mon, 11 Jun 2012 18:04:39 +0800 Subject: ASoC: mmp: add sspa support The SSPA is a configurable multi-channel audio serial (TDM) interface. It's configurable at runtime to support up to 128 channels and the number of bits per sample: 8, 12, 16, 20, 24 and 32 bits. It also support stereo format: I2S, left-justified or right-justified. Signed-off-by: Zhangfei Gao Signed-off-by: Leo Yan Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 5d76e29..6c3d00b 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -35,6 +35,9 @@ config SND_PXA_SOC_SSP tristate select PXA_SSP +config SND_MMP_SOC_SSPA + tristate + config SND_PXA2XX_SOC_CORGI tristate "SoC Audio support for Sharp Zaurus SL-C7x0" depends on SND_PXA2XX_SOC && PXA_SHARP_C7xx diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index f913e9b..07b8417 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -4,12 +4,14 @@ snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o snd-soc-pxa2xx-i2s-objs := pxa2xx-i2s.o snd-soc-pxa-ssp-objs := pxa-ssp.o snd-soc-mmp-objs := mmp-pcm.o +snd-soc-mmp-sspa-objs := mmp-sspa.o obj-$(CONFIG_SND_PXA2XX_SOC) += snd-soc-pxa2xx.o obj-$(CONFIG_SND_PXA2XX_SOC_AC97) += snd-soc-pxa2xx-ac97.o obj-$(CONFIG_SND_PXA2XX_SOC_I2S) += snd-soc-pxa2xx-i2s.o obj-$(CONFIG_SND_PXA_SOC_SSP) += snd-soc-pxa-ssp.o obj-$(CONFIG_SND_MMP_SOC) += snd-soc-mmp.o +obj-$(CONFIG_SND_MMP_SOC_SSPA) += snd-soc-mmp-sspa.o # PXA Machine Support snd-soc-corgi-objs := corgi.o diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c new file mode 100644 index 0000000..4d6cb8a --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.c @@ -0,0 +1,480 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.c + * Base on pxa2xx-ssp.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "mmp-sspa.h" + +/* + * SSPA audio private data + */ +struct sspa_priv { + struct ssp_device *sspa; + struct pxa2xx_pcm_dma_params *dma_params; + struct clk *audio_clk; + struct clk *sysclk; + int dai_fmt; + int running_cnt; +}; + +static void mmp_sspa_write_reg(struct ssp_device *sspa, u32 reg, u32 val) +{ + __raw_writel(val, sspa->mmio_base + reg); +} + +static u32 mmp_sspa_read_reg(struct ssp_device *sspa, u32 reg) +{ + return __raw_readl(sspa->mmio_base + reg); +} + +static void mmp_sspa_tx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_tx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_TXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); +} + +static void mmp_sspa_rx_enable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp |= SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static void mmp_sspa_rx_disable(struct ssp_device *sspa) +{ + unsigned int sspa_sp; + + sspa_sp = mmp_sspa_read_reg(sspa, SSPA_RXSP); + sspa_sp &= ~SSPA_SP_S_EN; + sspa_sp |= SSPA_SP_WEN; + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); +} + +static int mmp_sspa_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_enable(priv->sysclk); + clk_enable(priv->sspa->clk); + + return 0; +} + +static void mmp_sspa_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(dai); + + clk_disable(priv->sspa->clk); + clk_disable(priv->sysclk); + + return; +} + +/* + * Set the SSP ports SYSCLK. + */ +static int mmp_sspa_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (clk_id) { + case MMP_SSPA_CLK_AUDIO: + ret = clk_set_rate(priv->audio_clk, freq); + if (ret) + return ret; + break; + case MMP_SSPA_CLK_PLL: + case MMP_SSPA_CLK_VCXO: + /* not support yet */ + return -EINVAL; + default: + return -EINVAL; + } + + return 0; +} + +static int mmp_sspa_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, + int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct sspa_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + int ret = 0; + + switch (pll_id) { + case MMP_SYSCLK: + ret = clk_set_rate(priv->sysclk, freq_out); + if (ret) + return ret; + break; + case MMP_SSPA_CLK: + ret = clk_set_rate(priv->sspa->clk, freq_out); + if (ret) + return ret; + break; + default: + return -ENODEV; + } + + return 0; +} + +/* + * Set up the sspa dai format. The sspa port must be inactive + * before calling this function as the physical + * interface format is changed. + */ +static int mmp_sspa_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(cpu_dai); + struct ssp_device *sspa = sspa_priv->sspa; + u32 sspa_sp, sspa_ctrl; + + /* check if we need to change anything at all */ + if (sspa_priv->dai_fmt == fmt) + return 0; + + /* we can only change the settings if the port is not in use */ + if ((mmp_sspa_read_reg(sspa, SSPA_TXSP) & SSPA_SP_S_EN) || + (mmp_sspa_read_reg(sspa, SSPA_RXSP) & SSPA_SP_S_EN)) { + dev_err(&sspa->pdev->dev, + "can't change hardware dai format: stream is in use\n"); + return -EINVAL; + } + + /* reset port settings */ + sspa_sp = SSPA_SP_WEN | SSPA_SP_S_RST | SSPA_SP_FFLUSH; + sspa_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + sspa_sp |= SSPA_SP_MSL; + break; + case SND_SOC_DAIFMT_CBM_CFM: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + sspa_sp |= SSPA_SP_FSP; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + sspa_sp |= SSPA_TXSP_FPER(63); + sspa_sp |= SSPA_SP_FWID(31); + sspa_ctrl |= SSPA_CTL_XDATDLY(1); + break; + default: + return -EINVAL; + } + + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + sspa_sp &= ~(SSPA_SP_S_RST | SSPA_SP_FFLUSH); + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + mmp_sspa_write_reg(sspa, SSPA_RXSP, sspa_sp); + + /* + * FIXME: hw issue, for the tx serial port, + * can not config the master/slave mode; + * so must clean this bit. + * The master/slave mode has been set in the + * rx port. + */ + sspa_sp &= ~SSPA_SP_MSL; + mmp_sspa_write_reg(sspa, SSPA_TXSP, sspa_sp); + + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + + /* Since we are configuring the timings for the format by hand + * we have to defer some things until hw_params() where we + * know parameters like the sample size. + */ + sspa_priv->dai_fmt = fmt; + return 0; +} + +/* + * Set the SSPA audio DMA parameters and sample size. + * Can be called multiple times by oss emulation. + */ +static int mmp_sspa_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + struct pxa2xx_pcm_dma_params *dma_params; + u32 sspa_ctrl; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_TXCTL); + else + sspa_ctrl = mmp_sspa_read_reg(sspa, SSPA_RXCTL); + + sspa_ctrl &= ~SSPA_CTL_XFRLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XFRLEN1(params_channels(params) - 1); + sspa_ctrl &= ~SSPA_CTL_XWDLEN1_MASK; + sspa_ctrl |= SSPA_CTL_XWDLEN1(SSPA_CTL_32_BITS); + sspa_ctrl &= ~SSPA_CTL_XSSZ1_MASK; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_8_BITS); + break; + case SNDRV_PCM_FORMAT_S16_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_16_BITS); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_20_BITS); + break; + case SNDRV_PCM_FORMAT_S24_3LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_24_BITS); + break; + case SNDRV_PCM_FORMAT_S32_LE: + sspa_ctrl |= SSPA_CTL_XSSZ1(SSPA_CTL_32_BITS); + break; + default: + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + mmp_sspa_write_reg(sspa, SSPA_TXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_TXFIFO_LL, 0x1); + } else { + mmp_sspa_write_reg(sspa, SSPA_RXCTL, sspa_ctrl); + mmp_sspa_write_reg(sspa, SSPA_RXFIFO_UL, 0x0); + } + + dma_params = &sspa_priv->dma_params[substream->stream]; + dma_params->dev_addr = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + (sspa->phys_base + SSPA_TXD) : + (sspa->phys_base + SSPA_RXD); + snd_soc_dai_set_dma_data(cpu_dai, substream, dma_params); + return 0; +} + +static int mmp_sspa_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct sspa_priv *sspa_priv = snd_soc_dai_get_drvdata(dai); + struct ssp_device *sspa = sspa_priv->sspa; + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* + * whatever playback or capture, must enable rx. + * this is a hw issue, so need check if rx has been + * enabled or not; if has been enabled by another + * stream, do not enable again. + */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_enable(sspa); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_enable(sspa); + + sspa_priv->running_cnt++; + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sspa_priv->running_cnt--; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + mmp_sspa_tx_disable(sspa); + + /* have no capture stream, disable rx port */ + if (!sspa_priv->running_cnt) + mmp_sspa_rx_disable(sspa); + break; + + default: + ret = -EINVAL; + } + + return ret; +} + +static int mmp_sspa_probe(struct snd_soc_dai *dai) +{ + struct sspa_priv *priv = dev_get_drvdata(dai->dev); + + snd_soc_dai_set_drvdata(dai, priv); + return 0; + +} + +#define MMP_SSPA_RATES SNDRV_PCM_RATE_8000_192000 +#define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops mmp_sspa_dai_ops = { + .startup = mmp_sspa_startup, + .shutdown = mmp_sspa_shutdown, + .trigger = mmp_sspa_trigger, + .hw_params = mmp_sspa_hw_params, + .set_sysclk = mmp_sspa_set_dai_sysclk, + .set_pll = mmp_sspa_set_dai_pll, + .set_fmt = mmp_sspa_set_dai_fmt, +}; + +struct snd_soc_dai_driver mmp_sspa_dai = { + .probe = mmp_sspa_probe, + .playback = { + .channels_min = 1, + .channels_max = 128, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = MMP_SSPA_RATES, + .formats = MMP_SSPA_FORMATS, + }, + .ops = &mmp_sspa_dai_ops, +}; + +static __devinit int asoc_mmp_sspa_probe(struct platform_device *pdev) +{ + struct sspa_priv *priv; + struct resource *res; + + priv = devm_kzalloc(&pdev->dev, + sizeof(struct sspa_priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->sspa = devm_kzalloc(&pdev->dev, + sizeof(struct ssp_device), GFP_KERNEL); + if (priv->sspa == NULL) + return -ENOMEM; + + priv->dma_params = devm_kzalloc(&pdev->dev, + 2 * sizeof(struct pxa2xx_pcm_dma_params), GFP_KERNEL); + if (priv->dma_params == NULL) + return -ENOMEM; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (res == NULL) + return -ENOMEM; + + priv->sspa->mmio_base = devm_request_and_ioremap(&pdev->dev, res); + if (priv->sspa->mmio_base == NULL) + return -ENODEV; + + priv->sspa->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(priv->sspa->clk)) + return PTR_ERR(priv->sspa->clk); + + priv->audio_clk = clk_get(NULL, "mmp-audio"); + if (IS_ERR(priv->audio_clk)) + return PTR_ERR(priv->audio_clk); + + priv->sysclk = clk_get(NULL, "mmp-sysclk"); + if (IS_ERR(priv->sysclk)) { + clk_put(priv->audio_clk); + return PTR_ERR(priv->sysclk); + } + clk_enable(priv->audio_clk); + priv->dai_fmt = (unsigned int) -1; + platform_set_drvdata(pdev, priv); + + return snd_soc_register_dai(&pdev->dev, &mmp_sspa_dai); +} + +static int __devexit asoc_mmp_sspa_remove(struct platform_device *pdev) +{ + struct sspa_priv *priv = platform_get_drvdata(pdev); + + clk_disable(priv->audio_clk); + clk_put(priv->audio_clk); + clk_put(priv->sysclk); + snd_soc_unregister_dai(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_mmp_sspa_driver = { + .driver = { + .name = "mmp-sspa-dai", + .owner = THIS_MODULE, + }, + .probe = asoc_mmp_sspa_probe, + .remove = __devexit_p(asoc_mmp_sspa_remove), +}; + +module_platform_driver(asoc_mmp_sspa_driver); + +MODULE_AUTHOR("Leo Yan "); +MODULE_DESCRIPTION("MMP SSPA SoC Interface"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/pxa/mmp-sspa.h b/sound/soc/pxa/mmp-sspa.h new file mode 100644 index 0000000..ea365cb --- /dev/null +++ b/sound/soc/pxa/mmp-sspa.h @@ -0,0 +1,92 @@ +/* + * linux/sound/soc/pxa/mmp-sspa.h + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#ifndef _MMP_SSPA_H +#define _MMP_SSPA_H + +/* + * SSPA Registers + */ +#define SSPA_RXD (0x00) +#define SSPA_RXID (0x04) +#define SSPA_RXCTL (0x08) +#define SSPA_RXSP (0x0c) +#define SSPA_RXFIFO_UL (0x10) +#define SSPA_RXINT_MASK (0x14) +#define SSPA_RXC (0x18) +#define SSPA_RXFIFO_NOFS (0x1c) +#define SSPA_RXFIFO_SIZE (0x20) + +#define SSPA_TXD (0x80) +#define SSPA_TXID (0x84) +#define SSPA_TXCTL (0x88) +#define SSPA_TXSP (0x8c) +#define SSPA_TXFIFO_LL (0x90) +#define SSPA_TXINT_MASK (0x94) +#define SSPA_TXC (0x98) +#define SSPA_TXFIFO_NOFS (0x9c) +#define SSPA_TXFIFO_SIZE (0xa0) + +/* SSPA Control Register */ +#define SSPA_CTL_XPH (1 << 31) /* Read Phase */ +#define SSPA_CTL_XFIG (1 << 15) /* Transmit Zeros when FIFO Empty */ +#define SSPA_CTL_JST (1 << 3) /* Audio Sample Justification */ +#define SSPA_CTL_XFRLEN2_MASK (7 << 24) +#define SSPA_CTL_XFRLEN2(x) ((x) << 24) /* Transmit Frame Length in Phase 2 */ +#define SSPA_CTL_XWDLEN2_MASK (7 << 21) +#define SSPA_CTL_XWDLEN2(x) ((x) << 21) /* Transmit Word Length in Phase 2 */ +#define SSPA_CTL_XDATDLY(x) ((x) << 19) /* Tansmit Data Delay */ +#define SSPA_CTL_XSSZ2_MASK (7 << 16) +#define SSPA_CTL_XSSZ2(x) ((x) << 16) /* Transmit Sample Audio Size */ +#define SSPA_CTL_XFRLEN1_MASK (7 << 8) +#define SSPA_CTL_XFRLEN1(x) ((x) << 8) /* Transmit Frame Length in Phase 1 */ +#define SSPA_CTL_XWDLEN1_MASK (7 << 5) +#define SSPA_CTL_XWDLEN1(x) ((x) << 5) /* Transmit Word Length in Phase 1 */ +#define SSPA_CTL_XSSZ1_MASK (7 << 0) +#define SSPA_CTL_XSSZ1(x) ((x) << 0) /* XSSZ1 */ + +#define SSPA_CTL_8_BITS (0x0) /* Sample Size */ +#define SSPA_CTL_12_BITS (0x1) +#define SSPA_CTL_16_BITS (0x2) +#define SSPA_CTL_20_BITS (0x3) +#define SSPA_CTL_24_BITS (0x4) +#define SSPA_CTL_32_BITS (0x5) + +/* SSPA Serial Port Register */ +#define SSPA_SP_WEN (1 << 31) /* Write Configuration Enable */ +#define SSPA_SP_MSL (1 << 18) /* Master Slave Configuration */ +#define SSPA_SP_CLKP (1 << 17) /* CLKP Polarity Clock Edge Select */ +#define SSPA_SP_FSP (1 << 16) /* FSP Polarity Clock Edge Select */ +#define SSPA_SP_FFLUSH (1 << 2) /* FIFO Flush */ +#define SSPA_SP_S_RST (1 << 1) /* Active High Reset Signal */ +#define SSPA_SP_S_EN (1 << 0) /* Serial Clock Domain Enable */ +#define SSPA_SP_FWID(x) ((x) << 20) /* Frame-Sync Width */ +#define SSPA_TXSP_FPER(x) ((x) << 4) /* Frame-Sync Active */ + +/* sspa clock sources */ +#define MMP_SSPA_CLK_PLL 0 +#define MMP_SSPA_CLK_VCXO 1 +#define MMP_SSPA_CLK_AUDIO 3 + +/* sspa pll id */ +#define MMP_SYSCLK 0 +#define MMP_SSPA_CLK 1 + +#endif /* _MMP_SSPA_H */ -- cgit v0.10.2 From 5ebf20ae286a7d2b02551757166247a901d705e5 Mon Sep 17 00:00:00 2001 From: Zhangfei Gao Date: Mon, 11 Jun 2012 18:04:40 +0800 Subject: ASoC: add mmp brownstone support Adds Alsa audio platform driver for mmp brownstone machine Signed-off-by: Zhangfei Gao Signed-off-by: Leo Yan Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 6c3d00b..d389fd5 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -206,3 +206,13 @@ config SND_PXA2XX_SOC_IMOTE2 help Say Y if you want to add support for SoC audio on the IMote 2. + +config SND_MMP_SOC_BROWNSTONE + tristate "SoC Audio support for Marvell Brownstone" + depends on SND_MMP_SOC && MACH_BROWNSTONE + select SND_MMP_SOC_SSPA + select MFD_WM8994 + select SND_SOC_WM8994 + help + Say Y if you want to add support for SoC audio on the + Marvell Brownstone reference platform. diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index 07b8417..c12aa2a 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -32,6 +32,7 @@ snd-soc-mioa701-objs := mioa701_wm9713.o snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o snd-soc-raumfeld-objs := raumfeld.o +snd-soc-brownstone-objs := brownstone.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -51,3 +52,4 @@ obj-$(CONFIG_SND_SOC_TAVOREVB3) += snd-soc-tavorevb3.o obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o +obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c new file mode 100644 index 0000000..5e666e0 --- /dev/null +++ b/sound/soc/pxa/brownstone.c @@ -0,0 +1,174 @@ +/* + * linux/sound/soc/pxa/brownstone.c + * + * Copyright (C) 2011 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + */ + +#include +#include +#include +#include +#include + +#include "../codecs/wm8994.h" +#include "mmp-sspa.h" + +static const struct snd_kcontrol_new brownstone_dapm_control[] = { + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static const struct snd_soc_dapm_widget brownstone_dapm_widgets[] = { + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Main Mic", NULL), +}; + +static const struct snd_soc_dapm_route brownstone_audio_map[] = { + {"Ext Spk", NULL, "SPKOUTLP"}, + {"Ext Spk", NULL, "SPKOUTLN"}, + {"Ext Spk", NULL, "SPKOUTRP"}, + {"Ext Spk", NULL, "SPKOUTRN"}, + + {"Headset Stereophone", NULL, "HPOUT1L"}, + {"Headset Stereophone", NULL, "HPOUT1R"}, + + {"IN1RN", NULL, "Headset Mic"}, + + {"DMIC1DAT", NULL, "MICBIAS1"}, + {"MICBIAS1", NULL, "Main Mic"}, +}; + +static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + snd_soc_dapm_enable_pin(dapm, "Ext Spk"); + snd_soc_dapm_enable_pin(dapm, "Headset Stereophone"); + snd_soc_dapm_enable_pin(dapm, "Headset Mic"); + snd_soc_dapm_enable_pin(dapm, "Main Mic"); + + /* set endpoints to not connected */ + snd_soc_dapm_nc_pin(dapm, "HPOUT2P"); + snd_soc_dapm_nc_pin(dapm, "HPOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT1P"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2N"); + snd_soc_dapm_nc_pin(dapm, "LINEOUT2P"); + snd_soc_dapm_nc_pin(dapm, "IN1LN"); + snd_soc_dapm_nc_pin(dapm, "IN1LP"); + snd_soc_dapm_nc_pin(dapm, "IN1RP"); + snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN"); + snd_soc_dapm_nc_pin(dapm, "IN2RN"); + snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); + snd_soc_dapm_nc_pin(dapm, "IN2LN"); + + snd_soc_dapm_sync(dapm); + + return 0; +} + +static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + int freq_out, sspa_mclk, sysclk; + int sspa_div; + + if (params_rate(params) > 11025) { + freq_out = params_rate(params) * 512; + sysclk = params_rate(params) * 256; + sspa_mclk = params_rate(params) * 64; + } else { + freq_out = params_rate(params) * 1024; + sysclk = params_rate(params) * 512; + sspa_mclk = params_rate(params) * 64; + } + sspa_div = freq_out; + do_div(sspa_div, sspa_mclk); + + snd_soc_dai_set_sysclk(cpu_dai, MMP_SSPA_CLK_AUDIO, freq_out, 0); + snd_soc_dai_set_pll(cpu_dai, MMP_SYSCLK, 0, freq_out, sysclk); + snd_soc_dai_set_pll(cpu_dai, MMP_SSPA_CLK, 0, freq_out, sspa_mclk); + + /* set wm8994 sysclk */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_MCLK1, sysclk, 0); + + return 0; +} + +/* machine stream operations */ +static struct snd_soc_ops brownstone_ops = { + .hw_params = brownstone_wm8994_hw_params, +}; + +static struct snd_soc_dai_link brownstone_wm8994_dai[] = { +{ + .name = "WM8994", + .stream_name = "WM8994 HiFi", + .cpu_dai_name = "mmp-sspa-dai.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "mmp-pcm-audio", + .codec_name = "wm8994-codec", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ops = &brownstone_ops, + .init = brownstone_wm8994_init, +}, +}; + +/* audio machine driver */ +static struct snd_soc_card brownstone = { + .name = "brownstone", + .dai_link = brownstone_wm8994_dai, + .num_links = ARRAY_SIZE(brownstone_wm8994_dai), + + .controls = brownstone_dapm_control, + .num_controls = ARRAY_SIZE(brownstone_dapm_control), + .dapm_widgets = brownstone_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets), + .dapm_routes = brownstone_audio_map, + .num_dapm_routes = ARRAY_SIZE(brownstone_audio_map), +}; + +static int __devinit brownstone_probe(struct platform_device *pdev) +{ + int ret; + + brownstone.dev = &pdev->dev; + ret = snd_soc_register_card(&brownstone); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + return ret; +} + +static int __devexit brownstone_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&brownstone); + return 0; +} + +static struct platform_driver mmp_driver = { + .driver = { + .name = "brownstone-audio", + .owner = THIS_MODULE, + }, + .probe = brownstone_probe, + .remove = __devexit_p(brownstone_remove), +}; + +module_platform_driver(mmp_driver); + +MODULE_AUTHOR("Leo Yan "); +MODULE_DESCRIPTION("ALSA SoC Brownstone"); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From b883f363495f3d2e237170f6b8814869a3dd16fe Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Mon, 11 Jun 2012 18:04:41 +0800 Subject: ASoC: add ttc-dkb machine support add ttc-dkb machine support for pxa910. It uses 88pm8607 as codec dai, mmp-pcm as platform and pxa-ssp as cpu dai. Signed-off-by: Qiao Zhou Signed-off-by: Mark Brown diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index d389fd5..4d2e46f 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -150,6 +150,26 @@ config SND_SOC_TAVOREVB3 Say Y if you want to add support for SoC audio on the Marvell Saarb reference platform. +config SND_PXA910_SOC + tristate "SoC Audio for Marvell PXA910 chip" + depends on ARCH_MMP && SND + select SND_PCM + help + Say Y if you want to add support for SoC audio on the + Marvell PXA910 reference platform. + +config SND_SOC_TTC_DKB + bool "SoC Audio support for TTC DKB" + depends on SND_PXA910_SOC && MACH_TTC_DKB + select PXA_SSP + select SND_PXA_SOC_SSP + select SND_MMP_SOC + select MFD_88PM860X + select SND_SOC_88PM860X + help + Say Y if you want to add support for SoC audio on TTC DKB + + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE diff --git a/sound/soc/pxa/Makefile b/sound/soc/pxa/Makefile index c12aa2a..d8a265d 100644 --- a/sound/soc/pxa/Makefile +++ b/sound/soc/pxa/Makefile @@ -33,6 +33,7 @@ snd-soc-z2-objs := z2.o snd-soc-imote2-objs := imote2.o snd-soc-raumfeld-objs := raumfeld.o snd-soc-brownstone-objs := brownstone.o +snd-soc-ttc-dkb-objs := ttc-dkb.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o @@ -53,3 +54,4 @@ obj-$(CONFIG_SND_SOC_ZYLONITE) += snd-soc-zylonite.o obj-$(CONFIG_SND_PXA2XX_SOC_IMOTE2) += snd-soc-imote2.o obj-$(CONFIG_SND_SOC_RAUMFELD) += snd-soc-raumfeld.o obj-$(CONFIG_SND_MMP_SOC_BROWNSTONE) += snd-soc-brownstone.o +obj-$(CONFIG_SND_SOC_TTC_DKB) += snd-soc-ttc-dkb.o diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c new file mode 100644 index 0000000..935491a --- /dev/null +++ b/sound/soc/pxa/ttc-dkb.c @@ -0,0 +1,173 @@ +/* + * linux/sound/soc/pxa/ttc_dkb.c + * + * Copyright (C) 2012 Marvell International Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ +#include +#include +#include +#include +#include +#include +#include +#include +#include "../codecs/88pm860x-codec.h" + +static struct snd_soc_jack hs_jack, mic_jack; + +static struct snd_soc_jack_pin hs_jack_pins[] = { + { .pin = "Headset Stereophone", .mask = SND_JACK_HEADPHONE, }, +}; + +static struct snd_soc_jack_pin mic_jack_pins[] = { + { .pin = "Headset Mic 2", .mask = SND_JACK_MICROPHONE, }, +}; + +/* ttc machine dapm widgets */ +static const struct snd_soc_dapm_widget ttc_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_LINE("Lineout Out 1", NULL), + SND_SOC_DAPM_LINE("Lineout Out 2", NULL), + SND_SOC_DAPM_SPK("Ext Speaker", NULL), + SND_SOC_DAPM_MIC("Ext Mic 1", NULL), + SND_SOC_DAPM_MIC("Headset Mic 2", NULL), + SND_SOC_DAPM_MIC("Ext Mic 3", NULL), +}; + +/* ttc machine audio map */ +static const struct snd_soc_dapm_route ttc_audio_map[] = { + {"Headset Stereophone", NULL, "HS1"}, + {"Headset Stereophone", NULL, "HS2"}, + + {"Ext Speaker", NULL, "LSP"}, + {"Ext Speaker", NULL, "LSN"}, + + {"Lineout Out 1", NULL, "LINEOUT1"}, + {"Lineout Out 2", NULL, "LINEOUT2"}, + + {"MIC1P", NULL, "Mic1 Bias"}, + {"MIC1N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Ext Mic 1"}, + + {"MIC2P", NULL, "Mic1 Bias"}, + {"MIC2N", NULL, "Mic1 Bias"}, + {"Mic1 Bias", NULL, "Headset Mic 2"}, + + {"MIC3P", NULL, "Mic3 Bias"}, + {"MIC3N", NULL, "Mic3 Bias"}, + {"Mic3 Bias", NULL, "Ext Mic 3"}, +}; + +static int ttc_pm860x_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_dapm_context *dapm = &codec->dapm; + + /* connected pins */ + snd_soc_dapm_enable_pin(dapm, "Ext Speaker"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 1"); + snd_soc_dapm_enable_pin(dapm, "Ext Mic 3"); + snd_soc_dapm_disable_pin(dapm, "Headset Mic 2"); + snd_soc_dapm_disable_pin(dapm, "Headset Stereophone"); + + /* Headset jack detection */ + snd_soc_jack_new(codec, "Headphone Jack", SND_JACK_HEADPHONE + | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2, + &hs_jack); + snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins), + hs_jack_pins); + snd_soc_jack_new(codec, "Microphone Jack", SND_JACK_MICROPHONE, + &mic_jack); + snd_soc_jack_add_pins(&mic_jack, ARRAY_SIZE(mic_jack_pins), + mic_jack_pins); + + /* headphone, microphone detection & headset short detection */ + pm860x_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADPHONE, + SND_JACK_BTN_0, SND_JACK_BTN_1, SND_JACK_BTN_2); + pm860x_mic_jack_detect(codec, &hs_jack, SND_JACK_MICROPHONE); + + return 0; +} + +/* ttc/td-dkb digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link ttc_pm860x_hifi_dai[] = { +{ + .name = "88pm860x i2s", + .stream_name = "audio playback", + .codec_name = "88pm860x-codec", + .platform_name = "mmp-pcm-audio", + .cpu_dai_name = "pxa-ssp-dai.1", + .codec_dai_name = "88pm860x-i2s", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, + .init = ttc_pm860x_init, +}, +}; + +/* ttc/td audio machine driver */ +static struct snd_soc_card ttc_dkb_card = { + .name = "ttc-dkb-hifi", + .dai_link = ttc_pm860x_hifi_dai, + .num_links = ARRAY_SIZE(ttc_pm860x_hifi_dai), + + .dapm_widgets = ttc_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ttc_dapm_widgets), + .dapm_routes = ttc_audio_map, + .num_dapm_routes = ARRAY_SIZE(ttc_audio_map), +}; + +static int __devinit ttc_dkb_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card = &ttc_dkb_card; + int ret; + + card->dev = &pdev->dev; + + ret = snd_soc_register_card(card); + if (ret) + dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", + ret); + + return ret; +} + +static int __devexit ttc_dkb_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; +} + +static struct platform_driver ttc_dkb_driver = { + .driver = { + .name = "ttc-dkb-audio", + .owner = THIS_MODULE, + }, + .probe = ttc_dkb_probe, + .remove = __devexit_p(ttc_dkb_remove), +}; + +module_platform_driver(ttc_dkb_driver); + +/* Module information */ +MODULE_AUTHOR("Qiao Zhou, "); +MODULE_DESCRIPTION("ALSA SoC TTC DKB"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:ttc-dkb-audio"); -- cgit v0.10.2 From a9db7dbee0436f0c741c6dfb39ab0241d4131539 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 8 Jun 2012 12:34:20 -0600 Subject: ASoC: when initializing CPU DAI, don't duplicate any CODEC init If the CPU-side of a DAI link is a CODEC rather than a standalone DAI, the codec initialization will call try_module_get() and create the DAI widgets. Ensure that this isn't duplicated when the CPU DAI itself is probed, if the CPU DAI is part of a CODEC. Note that this is not a complete fix on its own, since there's no guarantee that the CODEC itself will be initialized - currently that only happens if the CODEC is also used as the CODEC-side of a DAI link, and that initialization may happen before or after the DAIs within the CODEC are initialized. However, such a scenario doesn't necessarily currently work, and I don't think this change alone makes it any worse. This is fixed in a couple patches time. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3d803f3..448d4a7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -983,7 +983,9 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - module_put(cpu_dai->dev->driver->owner); + + if (!cpu_dai->codec) + module_put(cpu_dai->dev->driver->owner); } } @@ -1257,11 +1259,13 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the cpu_dai */ if (!cpu_dai->probed && cpu_dai->driver->probe_order == order) { - cpu_dai->dapm.card = card; - if (!try_module_get(cpu_dai->dev->driver->owner)) - return -ENODEV; + if (!cpu_dai->codec) { + cpu_dai->dapm.card = card; + if (!try_module_get(cpu_dai->dev->driver->owner)) + return -ENODEV; - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); + } if (cpu_dai->driver->probe) { ret = cpu_dai->driver->probe(cpu_dai); -- cgit v0.10.2 From 18d756440e8ed292b772682c4440a9d33643225b Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 8 Jun 2012 12:34:21 -0600 Subject: ASoC: when removing a CPU DAI, clean up its DAPM context When a standalone CPU DAI (one not part of a CODEC) is probed, widgets are created for it. Add a call to snd_soc_dapm_free() in order to clean these up when the CPU DAI is removed. In order for snd_soc_dapm_free() to work, the CPU DAI's DAPM context's list member must be initialized, since snd_soc_dapm_free() removes that from the list it's part of. Add it to the card's list of DAPM contexts. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 448d4a7..621c5bd 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -984,8 +984,10 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) cpu_dai->probed = 0; list_del(&cpu_dai->card_list); - if (!cpu_dai->codec) + if (!cpu_dai->codec) { + snd_soc_dapm_free(&cpu_dai->dapm); module_put(cpu_dai->dev->driver->owner); + } } } @@ -1264,6 +1266,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; + list_add(&cpu_dai->dapm.list, &card->dapm_list); snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } -- cgit v0.10.2 From d12cd198cba7949c70f596296297b772063175c0 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 8 Jun 2012 12:34:22 -0600 Subject: ASoC: factor out soc_remove_platform() This change simply factors out part of soc_remove_dai_link() into a standalone function. This makes platform and CODEC removal much more similar at the call-sites. Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 621c5bd..a539ade 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -898,6 +898,28 @@ static int soc_bind_dai_link(struct snd_soc_card *card, int num) return 0; } +static int soc_remove_platform(struct snd_soc_platform *platform) +{ + int ret; + + if (platform->driver->remove) { + ret = platform->driver->remove(platform); + if (ret < 0) + pr_err("asoc: failed to remove %s: %d\n", + platform->name, ret); + } + + /* Make sure all DAPM widgets are freed */ + snd_soc_dapm_free(&platform->dapm); + + soc_cleanup_platform_debugfs(platform); + platform->probed = 0; + list_del(&platform->card_list); + module_put(platform->dev->driver->owner); + + return 0; +} + static void soc_remove_codec(struct snd_soc_codec *codec) { int err; @@ -950,22 +972,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) /* remove the platform */ if (platform && platform->probed && - platform->driver->remove_order == order) { - if (platform->driver->remove) { - err = platform->driver->remove(platform); - if (err < 0) - pr_err("asoc: failed to remove %s: %d\n", - platform->name, err); - } - - /* Make sure all DAPM widgets are freed */ - snd_soc_dapm_free(&platform->dapm); - - soc_cleanup_platform_debugfs(platform); - platform->probed = 0; - list_del(&platform->card_list); - module_put(platform->dev->driver->owner); - } + platform->driver->remove_order == order) + soc_remove_platform(platform); /* remove the CODEC */ if (codec && codec->probed && -- cgit v0.10.2 From 62ae68fa5d6d6f93d8ca8d00e21ad7ac410f9d58 Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Fri, 8 Jun 2012 12:34:23 -0600 Subject: ASoC: probe CODECs and platforms before DAIs and links soc_probe_dai_link() currently inter-mixes the probing of CODECs, platforms, and DAIs. This can lead to problems such as a CODEC's DAI being probed before the CODEC, if that DAI is used as the CPU-side of a DAI link without any other of the CODEC's DAIs having been used as the CODEC-side of any DAI link that was probed earlier. To solve this, split soc_probe_dai_link() into soc_probe_link_components() and soc_probe_link_dais(). The former is used to probe all CODECs and platforms used by a card first, and then the latter is used to probe all the DAIs and links later. A similar change is made to soc_remove_dai_links(). Signed-off-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a539ade..fe16135 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -941,11 +941,9 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) +static void soc_remove_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; - struct snd_soc_codec *codec = rtd->codec; - struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int err; @@ -970,16 +968,6 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) list_del(&codec_dai->card_list); } - /* remove the platform */ - if (platform && platform->probed && - platform->driver->remove_order == order) - soc_remove_platform(platform); - - /* remove the CODEC */ - if (codec && codec->probed && - codec->driver->remove_order == order) - soc_remove_codec(codec); - /* remove the cpu_dai */ if (cpu_dai && cpu_dai->probed && cpu_dai->driver->remove_order == order) { @@ -999,6 +987,38 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) } } +static void soc_remove_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + struct snd_soc_codec *codec; + + /* remove the platform */ + if (platform && platform->probed && + platform->driver->remove_order == order) { + soc_remove_platform(platform); + } + + /* remove the CODEC-side CODEC */ + if (codec_dai) { + codec = codec_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); + } + + /* remove any CPU-side CODEC */ + if (cpu_dai) { + codec = cpu_dai->codec; + if (codec && codec->probed && + codec->driver->remove_order == order) + soc_remove_codec(codec); + } +} + static void soc_remove_dai_links(struct snd_soc_card *card) { int dai, order; @@ -1006,8 +1026,15 @@ static void soc_remove_dai_links(struct snd_soc_card *card) for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (dai = 0; dai < card->num_rtd; dai++) - soc_remove_dai_link(card, dai, order); + soc_remove_link_dais(card, dai, order); + } + + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_link_components(card, dai, order); } + card->num_rtd = 0; } @@ -1244,7 +1271,44 @@ out: return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) +static int soc_probe_link_components(struct snd_soc_card *card, int num, + int order) +{ + struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_platform *platform = rtd->platform; + int ret; + + /* probe the CPU-side component, if it is a CODEC */ + if (cpu_dai->codec && + !cpu_dai->codec->probed && + cpu_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, cpu_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the CODEC-side component */ + if (!codec_dai->codec->probed && + codec_dai->codec->driver->probe_order == order) { + ret = soc_probe_codec(card, codec_dai->codec); + if (ret < 0) + return ret; + } + + /* probe the platform */ + if (!platform->probed && + platform->driver->probe_order == order) { + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; + } + + return 0; +} + +static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1292,22 +1356,6 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) list_add(&cpu_dai->card_list, &card->dai_dev_list); } - /* probe the CODEC */ - if (!codec->probed && - codec->driver->probe_order == order) { - ret = soc_probe_codec(card, codec); - if (ret < 0) - return ret; - } - - /* probe the platform */ - if (!platform->probed && - platform->driver->probe_order == order) { - ret = soc_probe_platform(card, platform); - if (ret < 0) - return ret; - } - /* probe the CODEC DAI */ if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { @@ -1582,14 +1630,27 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - /* early DAI link probe */ + /* probe all components used by DAI links on this card */ for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; order++) { for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i, order); + ret = soc_probe_link_components(card, i, order); if (ret < 0) { pr_err("asoc: failed to instantiate card %s: %d\n", - card->name, ret); + card->name, ret); + goto probe_dai_err; + } + } + } + + /* probe all DAI links on this card */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_link_dais(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", + card->name, ret); goto probe_dai_err; } } -- cgit v0.10.2 From e0690385a86cac5403a62d91dc146f2508416ded Mon Sep 17 00:00:00 2001 From: Ola Lilja Date: Tue, 12 Jun 2012 08:50:08 +0200 Subject: ASoC: Ux500: Add machine-driver Add machine-driver for ST-Ericsson U8500 platform, including support for the AB8500-codec. Signed-off-by: Ola Lilja Acked-by: Linus Walleij Signed-off-by: Mark Brown diff --git a/sound/soc/ux500/Kconfig b/sound/soc/ux500/Kconfig index 1d38515..069330d 100644 --- a/sound/soc/ux500/Kconfig +++ b/sound/soc/ux500/Kconfig @@ -19,3 +19,14 @@ config SND_SOC_UX500_PLAT_DMA select SND_SOC_DMAENGINE_PCM help Say Y if you want to enable the Ux500 platform-driver. + ++config SND_SOC_UX500_MACH_MOP500 ++ tristate "Machine - MOP500 (Ux500 + AB8500)" + depends on AB8500_CORE && AB8500_GPADC && SND_SOC_UX500 + select SND_SOC_AB8500_CODEC + select SND_SOC_UX500_PLAT_MSP_I2S + select SND_SOC_UX500_PLAT_DMA + help + Select this to enable the MOP500 machine-driver. + This will enable platform-drivers for: Ux500 + This will enable codec-drivers for: AB8500 diff --git a/sound/soc/ux500/Makefile b/sound/soc/ux500/Makefile index 4634bf0..cce0c11 100644 --- a/sound/soc/ux500/Makefile +++ b/sound/soc/ux500/Makefile @@ -5,3 +5,6 @@ obj-$(CONFIG_SND_SOC_UX500_PLAT_MSP_I2S) += snd-soc-ux500-plat-msp-i2s.o snd-soc-ux500-plat-dma-objs := ux500_pcm.o obj-$(CONFIG_SND_SOC_UX500_PLAT_DMA) += snd-soc-ux500-plat-dma.o + +snd-soc-ux500-mach-mop500-objs := mop500.o mop500_ab8500.o +obj-$(CONFIG_SND_SOC_UX500_MACH_MOP500) += snd-soc-ux500-mach-mop500.o diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c new file mode 100644 index 0000000..31c4d26 --- /dev/null +++ b/sound/soc/ux500/mop500.c @@ -0,0 +1,113 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja (ola.o.lilja@stericsson.com) + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include + +#include +#include +#include + +#include +#include + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" + +#include + +/* Define the whole MOP500 soundcard, linking platform to the codec-drivers */ +struct snd_soc_dai_link mop500_dai_links[] = { + { + .name = "ab8500_0", + .stream_name = "ab8500_0", + .cpu_dai_name = "ux500-msp-i2s.1", + .codec_dai_name = "ab8500-codec-dai.0", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = mop500_ab8500_machine_init, + .ops = mop500_ab8500_ops, + }, + { + .name = "ab8500_1", + .stream_name = "ab8500_1", + .cpu_dai_name = "ux500-msp-i2s.3", + .codec_dai_name = "ab8500-codec-dai.1", + .platform_name = "ux500-pcm.0", + .codec_name = "ab8500-codec.0", + .init = NULL, + .ops = mop500_ab8500_ops, + }, +}; + +static struct snd_soc_card mop500_card = { + .name = "MOP500-card", + .probe = NULL, + .dai_link = mop500_dai_links, + .num_links = ARRAY_SIZE(mop500_dai_links), +}; + +static int __devinit mop500_probe(struct platform_device *pdev) +{ + int ret; + + pr_debug("%s: Enter.\n", __func__); + + dev_dbg(&pdev->dev, "%s: Enter.\n", __func__); + + mop500_card.dev = &pdev->dev; + + dev_dbg(&pdev->dev, "%s: Card %s: Set platform drvdata.\n", + __func__, mop500_card.name); + platform_set_drvdata(pdev, &mop500_card); + + snd_soc_card_set_drvdata(&mop500_card, NULL); + + dev_dbg(&pdev->dev, "%s: Card %s: num_links = %d\n", + __func__, mop500_card.name, mop500_card.num_links); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: name = %s\n", + __func__, mop500_card.name, mop500_card.dai_link[0].name); + dev_dbg(&pdev->dev, "%s: Card %s: DAI-link 0: stream_name = %s\n", + __func__, mop500_card.name, + mop500_card.dai_link[0].stream_name); + + ret = snd_soc_register_card(&mop500_card); + if (ret) + dev_err(&pdev->dev, + "Error: snd_soc_register_card failed (%d)!\n", + ret); + + return ret; +} + +static int __devexit mop500_remove(struct platform_device *pdev) +{ + struct snd_soc_card *mop500_card = platform_get_drvdata(pdev); + + pr_debug("%s: Enter.\n", __func__); + + snd_soc_unregister_card(mop500_card); + mop500_ab8500_remove(mop500_card); + + return 0; +} + +static struct platform_driver snd_soc_mop500_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "snd-soc-mop500", + }, + .probe = mop500_probe, + .remove = __devexit_p(mop500_remove), +}; + +module_platform_driver(snd_soc_mop500_driver); diff --git a/sound/soc/ux500/mop500_ab8500.c b/sound/soc/ux500/mop500_ab8500.c new file mode 100644 index 0000000..78cce23 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.c @@ -0,0 +1,431 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja , + * Kristoffer Karlsson + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#include +#include +#include +#include + +#include + +#include +#include +#include +#include + +#include "ux500_pcm.h" +#include "ux500_msp_dai.h" +#include "../codecs/ab8500-codec.h" + +#define TX_SLOT_MONO 0x0008 +#define TX_SLOT_STEREO 0x000a +#define RX_SLOT_MONO 0x0001 +#define RX_SLOT_STEREO 0x0003 +#define TX_SLOT_8CH 0x00FF +#define RX_SLOT_8CH 0x00FF + +#define DEF_TX_SLOTS TX_SLOT_STEREO +#define DEF_RX_SLOTS RX_SLOT_MONO + +#define DRIVERMODE_NORMAL 0 +#define DRIVERMODE_CODEC_ONLY 1 + +/* Slot configuration */ +static unsigned int tx_slots = DEF_TX_SLOTS; +static unsigned int rx_slots = DEF_RX_SLOTS; + +/* Clocks */ +static const char * const enum_mclk[] = { + "SYSCLK", + "ULPCLK" +}; +enum mclk { + MCLK_SYSCLK, + MCLK_ULPCLK, +}; + +static SOC_ENUM_SINGLE_EXT_DECL(soc_enum_mclk, enum_mclk); + +/* Private data for machine-part MOP500<->AB8500 */ +struct mop500_ab8500_drvdata { + /* Clocks */ + enum mclk mclk_sel; + struct clk *clk_ptr_intclk; + struct clk *clk_ptr_sysclk; + struct clk *clk_ptr_ulpclk; +}; + +static inline const char *get_mclk_str(enum mclk mclk_sel) +{ + switch (mclk_sel) { + case MCLK_SYSCLK: + return "SYSCLK"; + case MCLK_ULPCLK: + return "ULPCLK"; + default: + return "Unknown"; + } +} + +static int mop500_ab8500_set_mclk(struct device *dev, + struct mop500_ab8500_drvdata *drvdata) +{ + int status; + struct clk *clk_ptr; + + if (IS_ERR(drvdata->clk_ptr_intclk)) { + dev_err(dev, + "%s: ERROR: intclk not initialized!\n", __func__); + return -EIO; + } + + switch (drvdata->mclk_sel) { + case MCLK_SYSCLK: + clk_ptr = drvdata->clk_ptr_sysclk; + break; + case MCLK_ULPCLK: + clk_ptr = drvdata->clk_ptr_ulpclk; + break; + default: + return -EINVAL; + } + + if (IS_ERR(clk_ptr)) { + dev_err(dev, "%s: ERROR: %s not initialized!\n", __func__, + get_mclk_str(drvdata->mclk_sel)); + return -EIO; + } + + status = clk_set_parent(drvdata->clk_ptr_intclk, clk_ptr); + if (status) + dev_err(dev, + "%s: ERROR: Setting intclk parent to %s failed (ret = %d)!", + __func__, get_mclk_str(drvdata->mclk_sel), status); + else + dev_dbg(dev, + "%s: intclk parent changed to %s.\n", + __func__, get_mclk_str(drvdata->mclk_sel)); + + return status; +} + +/* + * Control-events + */ + +static int mclk_input_control_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + + ucontrol->value.enumerated.item[0] = drvdata->mclk_sel; + + return 0; +} + +static int mclk_input_control_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mop500_ab8500_drvdata *drvdata = + snd_soc_card_get_drvdata(codec->card); + unsigned int val = ucontrol->value.enumerated.item[0]; + + if (val > (unsigned int)MCLK_ULPCLK) + return -EINVAL; + if (drvdata->mclk_sel == val) + return 0; + + drvdata->mclk_sel = val; + + return 1; +} + +/* + * Controls + */ + +static struct snd_kcontrol_new mop500_ab8500_ctrls[] = { + SOC_ENUM_EXT("Master Clock Select", + soc_enum_mclk, + mclk_input_control_get, mclk_input_control_put), + /* Digital interface - Clocks */ + SOC_SINGLE("Digital Interface Master Generator Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENMASTGEN, + 1, 0), + SOC_SINGLE("Digital Interface 0 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK0, + 1, 0), + SOC_SINGLE("Digital Interface 1 Bit-clock Switch", + AB8500_DIGIFCONF1, AB8500_DIGIFCONF1_ENFSBITCLK1, + 1, 0), + SOC_DAPM_PIN_SWITCH("Headset Left"), + SOC_DAPM_PIN_SWITCH("Headset Right"), + SOC_DAPM_PIN_SWITCH("Earpiece"), + SOC_DAPM_PIN_SWITCH("Speaker Left"), + SOC_DAPM_PIN_SWITCH("Speaker Right"), + SOC_DAPM_PIN_SWITCH("LineOut Left"), + SOC_DAPM_PIN_SWITCH("LineOut Right"), + SOC_DAPM_PIN_SWITCH("Vibra 1"), + SOC_DAPM_PIN_SWITCH("Vibra 2"), + SOC_DAPM_PIN_SWITCH("Mic 1"), + SOC_DAPM_PIN_SWITCH("Mic 2"), + SOC_DAPM_PIN_SWITCH("LineIn Left"), + SOC_DAPM_PIN_SWITCH("LineIn Right"), + SOC_DAPM_PIN_SWITCH("DMic 1"), + SOC_DAPM_PIN_SWITCH("DMic 2"), + SOC_DAPM_PIN_SWITCH("DMic 3"), + SOC_DAPM_PIN_SWITCH("DMic 4"), + SOC_DAPM_PIN_SWITCH("DMic 5"), + SOC_DAPM_PIN_SWITCH("DMic 6"), +}; + +/* ASoC */ + +int mop500_ab8500_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + /* Set audio-clock source */ + return mop500_ab8500_set_mclk(rtd->card->dev, + snd_soc_card_get_drvdata(rtd->card)); +} + +void mop500_ab8500_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->card->dev; + + dev_dbg(dev, "%s: Enter\n", __func__); + + /* Reset slots configuration to default(s) */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + tx_slots = DEF_TX_SLOTS; + else + rx_slots = DEF_RX_SLOTS; +} + +int mop500_ab8500_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct device *dev = rtd->card->dev; + unsigned int fmt; + int channels, ret = 0, driver_mode, slots; + unsigned int sw_codec, sw_cpu; + bool is_playback; + + dev_dbg(dev, "%s: Enter\n", __func__); + + dev_dbg(dev, "%s: substream->pcm->name = %s\n" + "substream->pcm->id = %s.\n" + "substream->name = %s.\n" + "substream->number = %d.\n", + __func__, + substream->pcm->name, + substream->pcm->id, + substream->name, + substream->number); + + channels = params_channels(params); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S32_LE: + sw_cpu = 32; + break; + + case SNDRV_PCM_FORMAT_S16_LE: + sw_cpu = 16; + break; + + default: + return -EINVAL; + } + + /* Setup codec depending on driver-mode */ + if (channels == 8) + driver_mode = DRIVERMODE_CODEC_ONLY; + else + driver_mode = DRIVERMODE_NORMAL; + dev_dbg(dev, "%s: Driver-mode: %s.\n", __func__, + (driver_mode == DRIVERMODE_NORMAL) ? "NORMAL" : "CODEC_ONLY"); + + /* Setup format */ + + if (driver_mode == DRIVERMODE_NORMAL) { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CONT; + } else { + fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_CBM_CFM | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_GATED; + } + + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for codec_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + dev_err(dev, + "%s: ERROR: snd_soc_dai_set_fmt failed for cpu_dai (ret = %d)!\n", + __func__, ret); + return ret; + } + + /* Setup TDM-slots */ + + is_playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK); + switch (channels) { + case 1: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_MONO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_MONO; + break; + case 2: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_STEREO : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_STEREO; + break; + case 8: + slots = 16; + tx_slots = (is_playback) ? TX_SLOT_8CH : 0; + rx_slots = (is_playback) ? 0 : RX_SLOT_8CH; + break; + default: + return -EINVAL; + } + + if (driver_mode == DRIVERMODE_NORMAL) + sw_codec = sw_cpu; + else + sw_codec = 20; + + dev_dbg(dev, "%s: CPU-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, tx_slots, rx_slots, slots, + sw_cpu); + if (ret) + return ret; + + dev_dbg(dev, "%s: CODEC-DAI TDM: TX=0x%04X RX=0x%04x\n", __func__, + tx_slots, rx_slots); + ret = snd_soc_dai_set_tdm_slot(codec_dai, tx_slots, rx_slots, slots, + sw_codec); + if (ret) + return ret; + + return 0; +} + +struct snd_soc_ops mop500_ab8500_ops[] = { + { + .hw_params = mop500_ab8500_hw_params, + .startup = mop500_ab8500_startup, + .shutdown = mop500_ab8500_shutdown, + } +}; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + struct device *dev = rtd->card->dev; + struct mop500_ab8500_drvdata *drvdata; + int ret; + + dev_dbg(dev, "%s Enter.\n", __func__); + + /* Create driver private-data struct */ + drvdata = devm_kzalloc(dev, sizeof(struct mop500_ab8500_drvdata), + GFP_KERNEL); + snd_soc_card_set_drvdata(rtd->card, drvdata); + + /* Setup clocks */ + + drvdata->clk_ptr_sysclk = clk_get(dev, "sysclk"); + if (IS_ERR(drvdata->clk_ptr_sysclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'sysclk'!\n", + __func__); + drvdata->clk_ptr_ulpclk = clk_get(dev, "ulpclk"); + if (IS_ERR(drvdata->clk_ptr_ulpclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'ulpclk'!\n", + __func__); + drvdata->clk_ptr_intclk = clk_get(dev, "intclk"); + if (IS_ERR(drvdata->clk_ptr_intclk)) + dev_warn(dev, "%s: WARNING: clk_get failed for 'intclk'!\n", + __func__); + + /* Set intclk default parent to ulpclk */ + drvdata->mclk_sel = MCLK_ULPCLK; + ret = mop500_ab8500_set_mclk(dev, drvdata); + if (ret < 0) + dev_warn(dev, "%s: WARNING: mop500_ab8500_set_mclk!\n", + __func__); + + drvdata->mclk_sel = MCLK_ULPCLK; + + /* Add controls */ + ret = snd_soc_add_codec_controls(codec, mop500_ab8500_ctrls, + ARRAY_SIZE(mop500_ab8500_ctrls)); + if (ret < 0) { + pr_err("%s: Failed to add machine-controls (%d)!\n", + __func__, ret); + return ret; + } + + ret = snd_soc_dapm_disable_pin(&codec->dapm, "Earpiece"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Speaker Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineOut Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Vibra 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "Mic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Left"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "LineIn Right"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 1"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 2"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 3"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 4"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 5"); + ret |= snd_soc_dapm_disable_pin(&codec->dapm, "DMic 6"); + + return ret; +} + +void mop500_ab8500_remove(struct snd_soc_card *card) +{ + struct mop500_ab8500_drvdata *drvdata = snd_soc_card_get_drvdata(card); + + if (drvdata->clk_ptr_sysclk != NULL) + clk_put(drvdata->clk_ptr_sysclk); + if (drvdata->clk_ptr_ulpclk != NULL) + clk_put(drvdata->clk_ptr_ulpclk); + if (drvdata->clk_ptr_intclk != NULL) + clk_put(drvdata->clk_ptr_intclk); + + snd_soc_card_set_drvdata(card, drvdata); +} diff --git a/sound/soc/ux500/mop500_ab8500.h b/sound/soc/ux500/mop500_ab8500.h new file mode 100644 index 0000000..cca5b33 --- /dev/null +++ b/sound/soc/ux500/mop500_ab8500.h @@ -0,0 +1,22 @@ +/* + * Copyright (C) ST-Ericsson SA 2012 + * + * Author: Ola Lilja + * for ST-Ericsson. + * + * License terms: + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as published + * by the Free Software Foundation. + */ + +#ifndef MOP500_AB8500_H +#define MOP500_AB8500_H + +extern struct snd_soc_ops mop500_ab8500_ops[]; + +int mop500_ab8500_machine_init(struct snd_soc_pcm_runtime *runtime); +void mop500_ab8500_remove(struct snd_soc_card *card); + +#endif -- cgit v0.10.2 From 9f0ed7a7c547efbce2c15b5017744809e9bba23a Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Sat, 16 Jun 2012 16:19:27 +0300 Subject: ASoC: Ux500: unlock on an error path There is a missing mutex_unlock() here. The cleanup path also has more debug output. Signed-off-by: Dan Carpenter Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index 389dd66..3c79592 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1235,7 +1235,8 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", __func__, enum_anc_state[req]); - return -EINVAL; + status = -EINVAL; + goto cleanup; } apply_fir = req == ANC_APPLY_FIR || req == ANC_APPLY_FIR_IIR; apply_iir = req == ANC_APPLY_IIR || req == ANC_APPLY_FIR_IIR; -- cgit v0.10.2 From 629b15b95d5b12a47791147b7559eacbad04d507 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 18 Jun 2012 21:09:04 +0100 Subject: ASoC: wm5100: Remove stubs of ASoC-level register map code Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 3823af3..f481729 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2378,13 +2378,6 @@ static int wm5100_remove(struct snd_soc_codec *codec) return 0; } -static int wm5100_soc_volatile(struct snd_soc_codec *codec, - unsigned int reg) -{ - return true; -} - - static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .probe = wm5100_probe, .remove = wm5100_remove, @@ -2392,8 +2385,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm5100 = { .set_sysclk = wm5100_set_sysclk, .set_pll = wm5100_set_fll, .idle_bias_off = 1, - .reg_cache_size = WM5100_MAX_REGISTER, - .volatile_register = wm5100_soc_volatile, .seq_notifier = wm5100_seq_notifier, .controls = wm5100_snd_controls, -- cgit v0.10.2 From d2aae47f804830da904d2454d73959eda4ebb0fd Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Wed, 20 Jun 2012 17:00:30 -0400 Subject: ASoC: add sport driver for bf6xx soc The SPORT(Serial Port) module on bf6xx soc has a totally different ip comparing to bf5xx soc. An individual SPORT module consists of two independently configurable SPORT halves with identical functionality. Each SPORT half can be configured for either transmitter or receiver. Signed-off-by: Scott Jiang Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 9f6bc55..0374a39 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -1,7 +1,8 @@ config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI BF5xx chip" + tristate "SoC I2S Audio for the ADI Blackfin chip" depends on BLACKFIN - select SND_BF5XX_SOC_SPORT + select SND_BF5XX_SOC_SPORT if !BF60x + select SND_BF6XX_SOC_SPORT if BF60x help Say Y or M if you want to add support for codecs attached to the Blackfin SPORT (synchronous serial ports) interface in I2S @@ -162,6 +163,9 @@ config SND_BF5XX_SOC_AD1980 config SND_BF5XX_SOC_SPORT tristate +config SND_BF6XX_SOC_SPORT + tristate + config SND_BF5XX_SOC_I2S tristate @@ -173,7 +177,7 @@ config SND_BF5XX_SOC_AC97 config SND_BF5XX_SPORT_NUM int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_I2S || SND_BF5XX_AC97 || SND_BF5XX_TDM) + depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) range 0 3 if BF54x range 0 1 if !BF54x default 0 diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 1bf86cc..13b0922 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -3,6 +3,7 @@ snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o snd-bf5xx-tdm-objs := bf5xx-tdm-pcm.o snd-soc-bf5xx-sport-objs := bf5xx-sport.o +snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o @@ -11,6 +12,7 @@ obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_TDM) += snd-bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o +obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c new file mode 100644 index 0000000..f19a72b --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -0,0 +1,422 @@ +/* + * bf6xx_sport.c Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "bf6xx-sport.h" + +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; + sport->tx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_tx_params); + +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params) +{ + if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) + return -EBUSY; + sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; + sport->rx_regs->div = params->div; + SSYNC(); + return 0; +} +EXPORT_SYMBOL(sport_set_rx_params); + +static int compute_wdsize(size_t wdsize) +{ + switch (wdsize) { + case 1: + return WDSIZE_8 | PSIZE_8; + case 2: + return WDSIZE_16 | PSIZE_16; + default: + return WDSIZE_32 | PSIZE_32; + } +} + +void sport_tx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); + set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->tx_dma_chan); + sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_tx_start); + +void sport_rx_start(struct sport_device *sport) +{ + set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); + set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR + | compute_wdsize(sport->wdsize) | NDSIZE_6); + enable_dma(sport->rx_dma_chan); + sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; + SSYNC(); +} +EXPORT_SYMBOL(sport_rx_start); + +void sport_tx_stop(struct sport_device *sport) +{ + sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->tx_dma_chan); +} +EXPORT_SYMBOL(sport_tx_stop); + +void sport_rx_stop(struct sport_device *sport) +{ + sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; + SSYNC(); + disable_dma(sport->rx_dma_chan); +} +EXPORT_SYMBOL(sport_rx_stop); + +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data) +{ + sport->tx_callback = tx_callback; + sport->tx_data = tx_data; +} +EXPORT_SYMBOL(sport_set_tx_callback); + +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data) +{ + sport->rx_callback = rx_callback; + sport->rx_data = rx_data; +} +EXPORT_SYMBOL(sport_set_rx_callback); + +static void setup_desc(struct dmasg *desc, void *buf, int fragcount, + size_t fragsize, unsigned int cfg, + unsigned int count, size_t wdsize) +{ + + int i; + + for (i = 0; i < fragcount; ++i) { + desc[i].next_desc_addr = &(desc[i + 1]); + desc[i].start_addr = (unsigned long)buf + i*fragsize; + desc[i].cfg = cfg; + desc[i].x_count = count; + desc[i].x_modify = wdsize; + desc[i].y_count = 0; + desc[i].y_modify = 0; + } + + /* make circular */ + desc[fragcount-1].next_desc_addr = desc; +} + +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->tx_desc) + dma_free_coherent(NULL, sport->tx_desc_size, + sport->tx_desc, 0); + + sport->tx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->tx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->tx_desc) + return -ENOMEM; + + sport->tx_buf = buf; + sport->tx_fragsize = fragsize; + sport->tx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; + + setup_desc(sport->tx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_tx_dma); + +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize) +{ + unsigned int count; + unsigned int cfg; + dma_addr_t addr; + + count = fragsize/sport->wdsize; + + if (sport->rx_desc) + dma_free_coherent(NULL, sport->rx_desc_size, + sport->rx_desc, 0); + + sport->rx_desc = dma_alloc_coherent(NULL, + fragcount * sizeof(struct dmasg), &addr, 0); + sport->rx_desc_size = fragcount * sizeof(struct dmasg); + if (!sport->rx_desc) + return -ENOMEM; + + sport->rx_buf = buf; + sport->rx_fragsize = fragsize; + sport->rx_frags = fragcount; + cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) + | WNR | NDSIZE_6; + + setup_desc(sport->rx_desc, buf, fragcount, fragsize, + cfg|DMAEN, count, sport->wdsize); + + return 0; +} +EXPORT_SYMBOL(sport_config_rx_dma); + +unsigned long sport_curr_offset_tx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); + + return (unsigned char *)curr - sport->tx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_tx); + +unsigned long sport_curr_offset_rx(struct sport_device *sport) +{ + unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); + + return (unsigned char *)curr - sport->rx_buf; +} +EXPORT_SYMBOL(sport_curr_offset_rx); + +static irqreturn_t sport_tx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + static unsigned long status; + + status = get_dma_curr_irqstat(sport->tx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->tx_dma_chan); + SSYNC(); + } + if (sport->tx_callback) + sport->tx_callback(sport->tx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_rx_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + unsigned long status; + + status = get_dma_curr_irqstat(sport->rx_dma_chan); + if (status & (DMA_DONE|DMA_ERR)) { + clear_dma_irqstat(sport->rx_dma_chan); + SSYNC(); + } + if (sport->rx_callback) + sport->rx_callback(sport->rx_data); + return IRQ_HANDLED; +} + +static irqreturn_t sport_err_irq(int irq, void *dev_id) +{ + struct sport_device *sport = dev_id; + struct device *dev = &sport->pdev->dev; + + if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: TUVF\n"); + if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) + dev_dbg(dev, "sport error: ROVF\n"); + + return IRQ_HANDLED; +} + +static int sport_get_resource(struct sport_device *sport) +{ + struct platform_device *pdev = sport->pdev; + struct device *dev = &pdev->dev; + struct bfin_snd_platform_data *pdata = dev->platform_data; + struct resource *res; + + if (!pdata) { + dev_err(dev, "No platform data\n"); + return -ENODEV; + } + sport->pin_req = pdata->pin_req; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(dev, "No tx MEM resource\n"); + return -ENODEV; + } + sport->tx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 1); + if (!res) { + dev_err(dev, "No rx MEM resource\n"); + return -ENODEV; + } + sport->rx_regs = (struct sport_register *)res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(dev, "No tx DMA resource\n"); + return -ENODEV; + } + sport->tx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(dev, "No rx DMA resource\n"); + return -ENODEV; + } + sport->rx_dma_chan = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); + if (!res) { + dev_err(dev, "No tx error irq resource\n"); + return -ENODEV; + } + sport->tx_err_irq = res->start; + + res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); + if (!res) { + dev_err(dev, "No rx error irq resource\n"); + return -ENODEV; + } + sport->rx_err_irq = res->start; + + return 0; +} + +static int sport_request_resource(struct sport_device *sport) +{ + struct device *dev = &sport->pdev->dev; + int ret; + + ret = peripheral_request_list(sport->pin_req, "soc-audio"); + if (ret) { + dev_err(dev, "Unable to request sport pin\n"); + return ret; + } + + ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); + goto err_tx_dma; + } + set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); + + ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); + if (ret) { + dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); + goto err_rx_dma; + } + set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); + + ret = request_irq(sport->tx_err_irq, sport_err_irq, + 0, "SPORT TX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); + goto err_tx_irq; + } + + ret = request_irq(sport->rx_err_irq, sport_err_irq, + 0, "SPORT RX ERROR", sport); + if (ret) { + dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); + goto err_rx_irq; + } + + return 0; +err_rx_irq: + free_irq(sport->tx_err_irq, sport); +err_tx_irq: + free_dma(sport->rx_dma_chan); +err_rx_dma: + free_dma(sport->tx_dma_chan); +err_tx_dma: + peripheral_free_list(sport->pin_req); + return ret; +} + +static void sport_free_resource(struct sport_device *sport) +{ + free_irq(sport->rx_err_irq, sport); + free_irq(sport->tx_err_irq, sport); + free_dma(sport->rx_dma_chan); + free_dma(sport->tx_dma_chan); + peripheral_free_list(sport->pin_req); +} + +struct sport_device *sport_create(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct sport_device *sport; + int ret; + + sport = kzalloc(sizeof(*sport), GFP_KERNEL); + if (!sport) { + dev_err(dev, "Unable to allocate memory for sport device\n"); + return NULL; + } + sport->pdev = pdev; + + ret = sport_get_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + ret = sport_request_resource(sport); + if (ret) { + kfree(sport); + return NULL; + } + + dev_dbg(dev, "SPORT create success\n"); + return sport; +} +EXPORT_SYMBOL(sport_create); + +void sport_delete(struct sport_device *sport) +{ + sport_free_resource(sport); +} +EXPORT_SYMBOL(sport_delete); + +MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); +MODULE_AUTHOR("Scott Jiang "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h new file mode 100644 index 0000000..307d193 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-sport.h @@ -0,0 +1,82 @@ +/* + * bf6xx_sport - Analog Devices BF6XX SPORT driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef _BF6XX_SPORT_H_ +#define _BF6XX_SPORT_H_ + +#include +#include + +struct sport_device { + struct platform_device *pdev; + const unsigned short *pin_req; + struct sport_register *tx_regs; + struct sport_register *rx_regs; + int tx_dma_chan; + int rx_dma_chan; + int tx_err_irq; + int rx_err_irq; + + void (*tx_callback)(void *data); + void *tx_data; + void (*rx_callback)(void *data); + void *rx_data; + + struct dmasg *tx_desc; + struct dmasg *rx_desc; + unsigned int tx_desc_size; + unsigned int rx_desc_size; + unsigned char *tx_buf; + unsigned char *rx_buf; + unsigned int tx_fragsize; + unsigned int rx_fragsize; + unsigned int tx_frags; + unsigned int rx_frags; + unsigned int wdsize; +}; + +struct sport_params { + u32 spctl; + u32 div; +}; + +struct sport_device *sport_create(struct platform_device *pdev); +void sport_delete(struct sport_device *sport); +int sport_set_tx_params(struct sport_device *sport, + struct sport_params *params); +int sport_set_rx_params(struct sport_device *sport, + struct sport_params *params); +void sport_tx_start(struct sport_device *sport); +void sport_rx_start(struct sport_device *sport); +void sport_tx_stop(struct sport_device *sport); +void sport_rx_stop(struct sport_device *sport); +void sport_set_tx_callback(struct sport_device *sport, + void (*tx_callback)(void *), void *tx_data); +void sport_set_rx_callback(struct sport_device *sport, + void (*rx_callback)(void *), void *rx_data); +int sport_config_tx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +int sport_config_rx_dma(struct sport_device *sport, void *buf, + int fragcount, size_t fragsize); +unsigned long sport_curr_offset_tx(struct sport_device *sport); +unsigned long sport_curr_offset_rx(struct sport_device *sport); + + + +#endif -- cgit v0.10.2 From f62ae7bda434ac5d2bcd6feb4f5bdb5885633177 Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Wed, 20 Jun 2012 17:00:31 -0400 Subject: ASoC: add i2s dai driver for bf6xx soc This driver enables i2s mode support on blackfin bf6xx platform. We reuse bf5xx-i2s-pcm.c as its i2s pcm driver because it's the same for both bf5xx and bf6xx soc. Signed-off-by: Scott Jiang Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 0374a39..16b88f5 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -10,12 +10,14 @@ config SND_BF5XX_I2S You will also need to select the audio interfaces to support below. config SND_BF5XX_SOC_SSM2602 - tristate "SoC SSM2602 Audio support for BF52x ezkit" + tristate "SoC SSM2602 Audio Codec Add-On Card support" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) - select SND_BF5XX_SOC_I2S + select SND_BF5XX_SOC_I2S if !BF60x + select SND_BF6XX_SOC_I2S if BF60x select SND_SOC_SSM2602 help - Say Y if you want to add support for SoC audio on BF527-EZKIT. + Say Y if you want to add support for the Analog Devices + SSM2602 Audio Codec Add-On Card. config SND_SOC_BFIN_EVAL_ADAU1701 tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" @@ -169,6 +171,9 @@ config SND_BF6XX_SOC_SPORT config SND_BF5XX_SOC_I2S tristate +config SND_BF6XX_SOC_I2S + tristate + config SND_BF5XX_SOC_TDM tristate diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile index 13b0922..6fea1f4 100644 --- a/sound/soc/blackfin/Makefile +++ b/sound/soc/blackfin/Makefile @@ -6,6 +6,7 @@ snd-soc-bf5xx-sport-objs := bf5xx-sport.o snd-soc-bf6xx-sport-objs := bf6xx-sport.o snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o +snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o snd-soc-bf5xx-tdm-objs := bf5xx-tdm.o obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o @@ -15,6 +16,7 @@ obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o +obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o obj-$(CONFIG_SND_BF5XX_SOC_TDM) += snd-soc-bf5xx-tdm.o # Blackfin Machine Support diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c new file mode 100644 index 0000000..c3c2466 --- /dev/null +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -0,0 +1,234 @@ +/* + * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver + * + * Copyright (c) 2012 Analog Devices Inc. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#include +#include +#include +#include +#include +#include +#include +#include + +#include "bf6xx-sport.h" + +struct sport_params param; + +static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR + | SPORT_CTL_LFS | SPORT_CTL_LAFS); + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE + | SPORT_CTL_LFS; + break; + case SND_SOC_DAIFMT_DSP_A: + param.spctl |= SPORT_CTL_FSR; + break; + case SND_SOC_DAIFMT_LEFT_J: + param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS + | SPORT_CTL_LAFS; + break; + default: + dev_err(dev, "%s: Unknown DAI format type\n", __func__); + ret = -EINVAL; + break; + } + + param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFS: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + ret = -EINVAL; + break; + default: + dev_err(dev, "%s: Unknown DAI master type\n", __func__); + ret = -EINVAL; + break; + } + + return ret; +} + +static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret = 0; + + param.spctl &= ~SPORT_CTL_SLEN; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + param.spctl |= 0x70; + sport->wdsize = 1; + case SNDRV_PCM_FORMAT_S16_LE: + param.spctl |= 0xf0; + sport->wdsize = 2; + break; + case SNDRV_PCM_FORMAT_S24_LE: + param.spctl |= 0x170; + sport->wdsize = 3; + break; + case SNDRV_PCM_FORMAT_S32_LE: + param.spctl |= 0x1f0; + sport->wdsize = 4; + break; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + } else { + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + } + return 0; +} + +#ifdef CONFIG_PM +static int bfin_i2s_suspend(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + + if (dai->capture_active) + sport_rx_stop(sport); + if (dai->playback_active) + sport_tx_stop(sport); + return 0; +} + +static int bfin_i2s_resume(struct snd_soc_dai *dai) +{ + struct sport_device *sport = snd_soc_dai_get_drvdata(dai); + struct device *dev = &sport->pdev->dev; + int ret; + + ret = sport_set_tx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT tx is busy!\n"); + return ret; + } + ret = sport_set_rx_params(sport, ¶m); + if (ret) { + dev_err(dev, "SPORT rx is busy!\n"); + return ret; + } + + return 0; +} + +#else +#define bfin_i2s_suspend NULL +#define bfin_i2s_resume NULL +#endif + +#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000) + +#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_ops bfin_i2s_dai_ops = { + .hw_params = bfin_i2s_hw_params, + .set_fmt = bfin_i2s_set_dai_fmt, +}; + +static struct snd_soc_dai_driver bfin_i2s_dai = { + .suspend = bfin_i2s_suspend, + .resume = bfin_i2s_resume, + .playback = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .capture = { + .channels_min = 1, + .channels_max = 2, + .rates = BFIN_I2S_RATES, + .formats = BFIN_I2S_FORMATS, + }, + .ops = &bfin_i2s_dai_ops, +}; + +static int __devinit bfin_i2s_probe(struct platform_device *pdev) +{ + struct sport_device *sport; + struct device *dev = &pdev->dev; + int ret; + + sport = sport_create(pdev); + if (!sport) + return -ENODEV; + + /* register with the ASoC layers */ + ret = snd_soc_register_dai(dev, &bfin_i2s_dai); + if (ret) { + dev_err(dev, "Failed to register DAI: %d\n", ret); + sport_delete(sport); + return ret; + } + platform_set_drvdata(pdev, sport); + + return 0; +} + +static int __devexit bfin_i2s_remove(struct platform_device *pdev) +{ + struct sport_device *sport = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&pdev->dev); + sport_delete(sport); + + return 0; +} + +static struct platform_driver bfin_i2s_driver = { + .probe = bfin_i2s_probe, + .remove = __devexit_p(bfin_i2s_remove), + .driver = { + .name = "bfin-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(bfin_i2s_driver); + +MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); +MODULE_AUTHOR("Scott Jiang "); +MODULE_LICENSE("GPL v2"); -- cgit v0.10.2 From c32c44cb58d212513243744878423abd207bc8a8 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:40 +0200 Subject: dmaengine: Add wrapper for device_tx_status callback This patch adds a small inline wrapper for the devivce_tx_status callback of a dma device. This makes the source code of users of this function a bit more compact and a bit more legible. E.g.: -status = chan->device->device_tx_status(chan, cookie, &state) +status = dmaengine_tx_status(chan, cookie, &state) Signed-off-by: Lars-Peter Clausen Acked-by Vinod Koul Signed-off-by: Mark Brown diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h index 56377df..cc0756a 100644 --- a/include/linux/dmaengine.h +++ b/include/linux/dmaengine.h @@ -670,6 +670,12 @@ static inline int dmaengine_resume(struct dma_chan *chan) return dmaengine_device_control(chan, DMA_RESUME, 0); } +static inline enum dma_status dmaengine_tx_status(struct dma_chan *chan, + dma_cookie_t cookie, struct dma_tx_state *state) +{ + return chan->device->device_tx_status(chan, cookie, state); +} + static inline dma_cookie_t dmaengine_submit(struct dma_async_tx_descriptor *desc) { return desc->tx_submit(desc); -- cgit v0.10.2 From 9883ab229d61b884323f9186b1bd4a41373a491b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:41 +0200 Subject: ASoC: dmaengine-pcm: Rename and deprecate snd_dmaengine_pcm_pointer Currently the sound dmaengine pcm helper functions implement the pcm_pointer callback by trying to count the number of elapsed periods. This is done by advancing the stream position in the dmaengine callback by one period. Unfortunately there is no guarantee that the callback will be called for each elapsed period. It may be possible that under high system load it is only called once for multiple elapsed periods. This patch renames the current implementation and documents its shortcomings and that it should not be used anymore in new drivers. The next patch will introduce a new snd_dmaengine_pcm_pointer which will be implemented based on querying the current stream position from the dma device. Signed-off-by: Lars-Peter Clausen Acked-by Vinod Koul Acked-by: Dong Aisheng diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index a8fcaa6..ea57915 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -38,7 +38,7 @@ void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream); int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); -snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream); +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream); int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data); diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index 162dbb7..4eea98b 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -136,7 +136,7 @@ static struct snd_pcm_ops ep93xx_pcm_ops = { .hw_params = ep93xx_pcm_hw_params, .hw_free = ep93xx_pcm_hw_free, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = ep93xx_pcm_mmap, }; diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index f3c0a5e..48f9d88 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -141,7 +141,7 @@ static struct snd_pcm_ops imx_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_imx_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_imx_pcm_mmap, }; diff --git a/sound/soc/mxs/mxs-pcm.c b/sound/soc/mxs/mxs-pcm.c index 373dec9..f82d766 100644 --- a/sound/soc/mxs/mxs-pcm.c +++ b/sound/soc/mxs/mxs-pcm.c @@ -141,7 +141,7 @@ static struct snd_pcm_ops mxs_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_mxs_pcm_hw_params, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = snd_mxs_pcm_mmap, }; diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 4756952..7c0877e 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -200,18 +200,18 @@ int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd) EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_trigger); /** - * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation + * snd_dmaengine_pcm_pointer_no_residue - dmaengine based PCM pointer implementation * @substream: PCM substream * - * This function can be used as the PCM pointer callback for dmaengine based PCM - * driver implementations. + * This function is deprecated and should not be used by new drivers, as its + * results may be unreliable. */ -snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) +snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream) { struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); return bytes_to_frames(substream->runtime, prtd->pos); } -EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index 97d8e4d..1a04e24 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -261,7 +261,7 @@ static struct snd_pcm_ops ux500_pcm_ops = { .hw_params = ux500_pcm_hw_params, .hw_free = ux500_pcm_hw_free, .trigger = snd_dmaengine_pcm_trigger, - .pointer = snd_dmaengine_pcm_pointer, + .pointer = snd_dmaengine_pcm_pointer_no_residue, .mmap = ux500_pcm_mmap }; -- cgit v0.10.2 From 3528f27a5d4ac299e2d8cbe7297c1e9edd601ee6 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 11 Jun 2012 20:11:42 +0200 Subject: ASoC: dmaengine-pcm: Add support for querying stream position from DMA driver Currently the sound dmaengine pcm helper functions implement the pcm_pointer callback by trying to count the number of elapsed periods. This is done by advancing the stream position in the dmaengine callback by one period. Unfortunately there is no guarantee that the callback will be called for each elapsed period. It may be possible that under high system load it is only called once for multiple elapsed periods. This patch addresses the issue by implementing support for querying the current stream position directly from the dmaengine driver. Since not all dmaengine drivers support reporting the stream position yet the old period counting implementation is kept for now. Furthermore the new mechanism allows to report the stream position with a sub-period granularity, given that the dmaengine driver supports this. Signed-off-by: Lars-Peter Clausen Acked-by: Vinod Koul Signed-off-by: Mark Brown diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index ea57915..b877334 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -38,6 +38,7 @@ void *snd_dmaengine_pcm_get_data(struct snd_pcm_substream *substream); int snd_hwparams_to_dma_slave_config(const struct snd_pcm_substream *substream, const struct snd_pcm_hw_params *params, struct dma_slave_config *slave_config); int snd_dmaengine_pcm_trigger(struct snd_pcm_substream *substream, int cmd); +snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream); snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream *substream); int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 7c0877e..2995334d 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -30,6 +30,7 @@ struct dmaengine_pcm_runtime_data { struct dma_chan *dma_chan; + dma_cookie_t cookie; unsigned int pos; @@ -153,7 +154,7 @@ static int dmaengine_pcm_prepare_and_submit(struct snd_pcm_substream *substream) desc->callback = dmaengine_pcm_dma_complete; desc->callback_param = substream; - dmaengine_submit(desc); + prtd->cookie = dmaengine_submit(desc); return 0; } @@ -213,6 +214,32 @@ snd_pcm_uframes_t snd_dmaengine_pcm_pointer_no_residue(struct snd_pcm_substream } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer_no_residue); +/** + * snd_dmaengine_pcm_pointer - dmaengine based PCM pointer implementation + * @substream: PCM substream + * + * This function can be used as the PCM pointer callback for dmaengine based PCM + * driver implementations. + */ +snd_pcm_uframes_t snd_dmaengine_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct dmaengine_pcm_runtime_data *prtd = substream_to_prtd(substream); + struct dma_tx_state state; + enum dma_status status; + unsigned int buf_size; + unsigned int pos = 0; + + status = dmaengine_tx_status(prtd->dma_chan, prtd->cookie, &state); + if (status == DMA_IN_PROGRESS || status == DMA_PAUSED) { + buf_size = snd_pcm_lib_buffer_bytes(substream); + if (state.residue > 0 && state.residue <= buf_size) + pos = buf_size - state.residue; + } + + return bytes_to_frames(substream->runtime, pos); +} +EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_pointer); + static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd, dma_filter_fn filter_fn, void *filter_data) { -- cgit v0.10.2 From f4a6348391fa029c0e230230adfafb7f33d4683e Mon Sep 17 00:00:00 2001 From: Scott Jiang Date: Thu, 21 Jun 2012 13:51:58 -0400 Subject: ASoC: bfin: use dev_err to print error log instead of dev_dbg Signed-off-by: Scott Jiang Signed-off-by: Mark Brown diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c index f19a72b..318c5ba 100644 --- a/sound/soc/blackfin/bf6xx-sport.c +++ b/sound/soc/blackfin/bf6xx-sport.c @@ -256,9 +256,9 @@ static irqreturn_t sport_err_irq(int irq, void *dev_id) struct device *dev = &sport->pdev->dev; if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: TUVF\n"); + dev_err(dev, "sport error: TUVF\n"); if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) - dev_dbg(dev, "sport error: ROVF\n"); + dev_err(dev, "sport error: ROVF\n"); return IRQ_HANDLED; } -- cgit v0.10.2 From 3a9cf8efd7b64f26f1e0f02afb70382f90cc11ca Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Thu, 21 Jun 2012 15:54:51 +0530 Subject: ASoC: Add support for synopsys i2s controller as per ASoC framework. This patch add support for synopsys I2S controller as per the ASoC framework. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h new file mode 100644 index 0000000..26f406e --- /dev/null +++ b/include/sound/designware_i2s.h @@ -0,0 +1,69 @@ +/* + * Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +#ifndef __SOUND_DESIGNWARE_I2S_H +#define __SOUND_DESIGNWARE_I2S_H + +#include +#include + +/* + * struct i2s_clk_config_data - represent i2s clk configuration data + * @chan_nr: number of channel + * @data_width: number of bits per sample (8/16/24/32 bit) + * @sample_rate: sampling frequency (8Khz, 16Khz, 32Khz, 44Khz, 48Khz) + */ +struct i2s_clk_config_data { + int chan_nr; + u32 data_width; + u32 sample_rate; +}; + +struct i2s_platform_data { + #define DWC_I2S_PLAY (1 << 0) + #define DWC_I2S_RECORD (1 << 1) + unsigned int cap; + int channel; + u32 snd_fmts; + u32 snd_rates; + + void *play_dma_data; + void *capture_dma_data; + bool (*filter)(struct dma_chan *chan, void *slave); + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +struct i2s_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +/* I2S DMA registers */ +#define I2S_RXDMA 0x01C0 +#define I2S_TXDMA 0x01C8 + +#define TWO_CHANNEL_SUPPORT 2 /* up to 2.0 */ +#define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ +#define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ +#define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ + +#endif /* __SOUND_DESIGNWARE_I2S_H */ diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 40b2ad1..c5de0a8 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -33,6 +33,7 @@ source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/blackfin/Kconfig" source "sound/soc/davinci/Kconfig" +source "sound/soc/dwc/Kconfig" source "sound/soc/ep93xx/Kconfig" source "sound/soc/fsl/Kconfig" source "sound/soc/jz4740/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 70990f4..00a555a 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -11,6 +11,7 @@ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += davinci/ +obj-$(CONFIG_SND_SOC) += dwc/ obj-$(CONFIG_SND_SOC) += ep93xx/ obj-$(CONFIG_SND_SOC) += fsl/ obj-$(CONFIG_SND_SOC) += jz4740/ diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig new file mode 100644 index 0000000..93e9fc3 --- /dev/null +++ b/sound/soc/dwc/Kconfig @@ -0,0 +1,8 @@ +config SND_DESIGNWARE_I2S + tristate "Synopsys I2S Device Driver" + help + Say Y or M if you want to add support for I2S driver for + Synopsys desigwnware I2S device. The device supports upto + maximum of 8 channels each for play and record. + + diff --git a/sound/soc/dwc/Makefile b/sound/soc/dwc/Makefile new file mode 100644 index 0000000..319371f --- /dev/null +++ b/sound/soc/dwc/Makefile @@ -0,0 +1,3 @@ +# SYNOPSYS Platform Support +obj-$(CONFIG_SND_DESIGNWARE_I2S) += designware_i2s.o + diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c new file mode 100644 index 0000000..e667e2b --- /dev/null +++ b/sound/soc/dwc/designware_i2s.c @@ -0,0 +1,454 @@ +/* + * ALSA SoC Synopsys I2S Audio Layer + * + * sound/soc/spear/designware_i2s.c + * + * Copyright (C) 2010 ST Microelectronics + * Rajeev Kumar + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +/* common register for all channel */ +#define IER 0x000 +#define IRER 0x004 +#define ITER 0x008 +#define CER 0x00C +#define CCR 0x010 +#define RXFFR 0x014 +#define TXFFR 0x018 + +/* I2STxRxRegisters for all channels */ +#define LRBR_LTHR(x) (0x40 * x + 0x020) +#define RRBR_RTHR(x) (0x40 * x + 0x024) +#define RER(x) (0x40 * x + 0x028) +#define TER(x) (0x40 * x + 0x02C) +#define RCR(x) (0x40 * x + 0x030) +#define TCR(x) (0x40 * x + 0x034) +#define ISR(x) (0x40 * x + 0x038) +#define IMR(x) (0x40 * x + 0x03C) +#define ROR(x) (0x40 * x + 0x040) +#define TOR(x) (0x40 * x + 0x044) +#define RFCR(x) (0x40 * x + 0x048) +#define TFCR(x) (0x40 * x + 0x04C) +#define RFF(x) (0x40 * x + 0x050) +#define TFF(x) (0x40 * x + 0x054) + +/* I2SCOMPRegisters */ +#define I2S_COMP_PARAM_2 0x01F0 +#define I2S_COMP_PARAM_1 0x01F4 +#define I2S_COMP_VERSION 0x01F8 +#define I2S_COMP_TYPE 0x01FC + +#define MAX_CHANNEL_NUM 8 +#define MIN_CHANNEL_NUM 2 + +struct dw_i2s_dev { + void __iomem *i2s_base; + struct clk *clk; + int active; + unsigned int capability; + struct device *dev; + + /* data related to DMA transfers b/w i2s and DMAC */ + struct i2s_dma_data play_dma_data; + struct i2s_dma_data capture_dma_data; + struct i2s_clk_config_data config; + int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); +}; + +static inline void i2s_write_reg(void *io_base, int reg, u32 val) +{ + writel(val, io_base + reg); +} + +static inline u32 i2s_read_reg(void *io_base, int reg) +{ + return readl(io_base + reg); +} + +static inline void i2s_disable_channels(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TER(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, RER(i), 0); + } +} + +static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) +{ + u32 i = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, TOR(i), 0); + } else { + for (i = 0; i < 4; i++) + i2s_write_reg(dev->i2s_base, ROR(i), 0); + } +} + +void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) +{ + + i2s_write_reg(dev->i2s_base, IER, 1); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + i2s_write_reg(dev->i2s_base, ITER, 1); + else + i2s_write_reg(dev->i2s_base, IRER, 1); + + i2s_write_reg(dev->i2s_base, CER, 1); +} + +static void i2s_stop(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) +{ + u32 i = 0, irq; + + i2s_clear_irqs(dev, substream->stream); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, ITER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x30); + } + } else { + i2s_write_reg(dev->i2s_base, IRER, 0); + + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq | 0x03); + } + } + + if (!dev->active) { + i2s_write_reg(dev->i2s_base, CER, 0); + i2s_write_reg(dev->i2s_base, IER, 0); + } +} + +static int dw_i2s_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(cpu_dai); + struct i2s_dma_data *dma_data = NULL; + + if (!(dev->capability & DWC_I2S_RECORD) && + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) + return -EINVAL; + + if (!(dev->capability & DWC_I2S_PLAY) && + (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dma_data = &dev->play_dma_data; + else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) + dma_data = &dev->capture_dma_data; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)dma_data); + + return 0; +} + +static int dw_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + struct i2s_clk_config_data *config = &dev->config; + u32 ccr, xfer_resolution, ch_reg, irq; + int ret; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + config->data_width = 16; + ccr = 0x00; + xfer_resolution = 0x02; + break; + + case SNDRV_PCM_FORMAT_S24_LE: + config->data_width = 24; + ccr = 0x08; + xfer_resolution = 0x04; + break; + + case SNDRV_PCM_FORMAT_S32_LE: + config->data_width = 32; + ccr = 0x10; + xfer_resolution = 0x05; + break; + + default: + dev_err(dev->dev, "designware-i2s: unsuppted PCM fmt"); + return -EINVAL; + } + + config->chan_nr = params_channels(params); + + switch (config->chan_nr) { + case EIGHT_CHANNEL_SUPPORT: + ch_reg = 3; + case SIX_CHANNEL_SUPPORT: + ch_reg = 2; + case FOUR_CHANNEL_SUPPORT: + ch_reg = 1; + case TWO_CHANNEL_SUPPORT: + ch_reg = 0; + break; + default: + dev_err(dev->dev, "channel not supported\n"); + } + + i2s_disable_channels(dev, substream->stream); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + i2s_write_reg(dev->i2s_base, TCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, TFCR(ch_reg), 0x02); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x30); + i2s_write_reg(dev->i2s_base, TER(ch_reg), 1); + } else { + i2s_write_reg(dev->i2s_base, RCR(ch_reg), xfer_resolution); + i2s_write_reg(dev->i2s_base, RFCR(ch_reg), 0x07); + irq = i2s_read_reg(dev->i2s_base, IMR(ch_reg)); + i2s_write_reg(dev->i2s_base, IMR(ch_reg), irq & ~0x03); + i2s_write_reg(dev->i2s_base, RER(ch_reg), 1); + } + + i2s_write_reg(dev->i2s_base, CCR, ccr); + + config->sample_rate = params_rate(params); + + if (!dev->i2s_clk_cfg) + return -EINVAL; + + ret = dev->i2s_clk_cfg(config); + if (ret < 0) { + dev_err(dev->dev, "runtime audio clk config fail\n"); + return ret; + } + + return 0; +} + +static void dw_i2s_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int dw_i2s_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + int ret = 0; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + dev->active++; + i2s_start(dev, substream); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + dev->active--; + i2s_stop(dev, substream); + break; + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops dw_i2s_dai_ops = { + .startup = dw_i2s_startup, + .shutdown = dw_i2s_shutdown, + .hw_params = dw_i2s_hw_params, + .trigger = dw_i2s_trigger, +}; + +#ifdef CONFIG_PM + +static int dw_i2s_suspend(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_disable(dev->clk); + return 0; +} + +static int dw_i2s_resume(struct snd_soc_dai *dai) +{ + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(dai); + + clk_enable(dev->clk); + return 0; +} + +#else +#define dw_i2s_suspend NULL +#define dw_i2s_resume NULL +#endif + +static int dw_i2s_probe(struct platform_device *pdev) +{ + const struct i2s_platform_data *pdata = pdev->dev.platform_data; + struct dw_i2s_dev *dev; + struct resource *res; + int ret; + unsigned int cap; + struct snd_soc_dai_driver *dw_i2s_dai; + + if (!pdata) { + dev_err(&pdev->dev, "Invalid platform data\n"); + return -EINVAL; + } + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "no i2s resource defined\n"); + return -ENODEV; + } + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_err(&pdev->dev, "i2s region already claimed\n"); + return -EBUSY; + } + + dev = devm_kzalloc(&pdev->dev, sizeof(*dev), GFP_KERNEL); + if (!dev) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + dev->i2s_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!dev->i2s_base) { + dev_err(&pdev->dev, "ioremap fail for i2s_region\n"); + return -ENOMEM; + } + + cap = pdata->cap; + dev->capability = cap; + dev->i2s_clk_cfg = pdata->i2s_clk_cfg; + + /* Set DMA slaves info */ + + dev->play_dma_data.data = pdata->play_dma_data; + dev->capture_dma_data.data = pdata->capture_dma_data; + dev->play_dma_data.addr = res->start + I2S_TXDMA; + dev->capture_dma_data.addr = res->start + I2S_RXDMA; + dev->play_dma_data.max_burst = 16; + dev->capture_dma_data.max_burst = 16; + dev->play_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->capture_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + dev->play_dma_data.filter = pdata->filter; + dev->capture_dma_data.filter = pdata->filter; + + dev->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(dev->clk)) + return PTR_ERR(dev->clk); + + ret = clk_enable(dev->clk); + if (ret < 0) + goto err_clk_put; + + dw_i2s_dai = devm_kzalloc(&pdev->dev, sizeof(*dw_i2s_dai), GFP_KERNEL); + if (!dw_i2s_dai) { + dev_err(&pdev->dev, "mem allocation failed for dai driver\n"); + ret = -ENOMEM; + goto err_clk_disable; + } + + if (cap & DWC_I2S_PLAY) { + dev_dbg(&pdev->dev, " SPEAr: play supported\n"); + dw_i2s_dai->playback.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->playback.channels_max = pdata->channel; + dw_i2s_dai->playback.formats = pdata->snd_fmts; + dw_i2s_dai->playback.rates = pdata->snd_rates; + } + + if (cap & DWC_I2S_RECORD) { + dev_dbg(&pdev->dev, "SPEAr: record supported\n"); + dw_i2s_dai->capture.channels_min = MIN_CHANNEL_NUM; + dw_i2s_dai->capture.channels_max = pdata->channel; + dw_i2s_dai->capture.formats = pdata->snd_fmts; + dw_i2s_dai->capture.rates = pdata->snd_rates; + } + + dw_i2s_dai->ops = &dw_i2s_dai_ops; + dw_i2s_dai->suspend = dw_i2s_suspend; + dw_i2s_dai->resume = dw_i2s_resume; + + dev->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, dev); + ret = snd_soc_register_dai(&pdev->dev, dw_i2s_dai); + if (ret != 0) { + dev_err(&pdev->dev, "not able to register dai\n"); + goto err_set_drvdata; + } + + return 0; + +err_set_drvdata: + dev_set_drvdata(&pdev->dev, NULL); +err_clk_disable: + clk_disable(dev->clk); +err_clk_put: + clk_put(dev->clk); + return ret; +} + +static int dw_i2s_remove(struct platform_device *pdev) +{ + struct dw_i2s_dev *dev = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(dev->clk); + + return 0; +} + +static struct platform_driver dw_i2s_driver = { + .probe = dw_i2s_probe, + .remove = dw_i2s_remove, + .driver = { + .name = "designware-i2s", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(dw_i2s_driver); + +MODULE_AUTHOR("Rajeev Kumar "); +MODULE_DESCRIPTION("DESIGNWARE I2S SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:designware_i2s"); -- cgit v0.10.2 From 241b446f30de171b627524c107ce03e5ecee0124 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Thu, 21 Jun 2012 15:54:52 +0530 Subject: ASoC: Add support for SPEAr ASoC pcm layer. This patch add support for the SPEAr ASoC pcm layer in ASoC framework. The pcm layer uses common snd_dmaengine framework. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/include/sound/spear_dma.h b/include/sound/spear_dma.h new file mode 100644 index 0000000..1b365bf --- /dev/null +++ b/include/sound/spear_dma.h @@ -0,0 +1,35 @@ +/* +* linux/spear_dma.h +* +* Copyright (ST) 2012 Rajeev Kumar (rajeev-dlh.kumar@st.com) +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License as published by +* the Free Software Foundation; either version 2 of the License, or +* (at your option) any later version. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#ifndef SPEAR_DMA_H +#define SPEAR_DMA_H + +#include + +struct spear_dma_data { + void *data; + dma_addr_t addr; + u32 max_burst; + enum dma_slave_buswidth addr_width; + bool (*filter)(struct dma_chan *chan, void *slave); +}; + +#endif /* SPEAR_DMA_H */ diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c new file mode 100644 index 0000000..97c2cac --- /dev/null +++ b/sound/soc/spear/spear_pcm.c @@ -0,0 +1,214 @@ +/* + * ALSA PCM interface for ST SPEAr Processors + * + * sound/soc/spear/spear_pcm.c + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +struct snd_pcm_hardware spear_pcm_hardware = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME), + .buffer_bytes_max = 16 * 1024, /* max buffer size */ + .period_bytes_min = 2 * 1024, /* 1 msec data minimum period size */ + .period_bytes_max = 2 * 1024, /* maximum period size */ + .periods_min = 1, /* min # periods */ + .periods_max = 8, /* max # of periods */ + .fifo_size = 0, /* fifo size in bytes */ +}; + +static int spear_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int spear_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int spear_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + + struct spear_dma_data *dma_data = (struct spear_dma_data *) + snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + int ret; + + ret = snd_soc_set_runtime_hwparams(substream, &spear_pcm_hardware); + if (ret) + return ret; + + ret = snd_dmaengine_pcm_open(substream, dma_data->filter, dma_data); + if (ret) + return ret; + + snd_dmaengine_pcm_set_data(substream, dma_data); + + return 0; +} + +static int spear_pcm_close(struct snd_pcm_substream *substream) +{ + + snd_dmaengine_pcm_close(substream); + + return 0; +} + +static int spear_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops spear_pcm_ops = { + .open = spear_pcm_open, + .close = spear_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = spear_pcm_hw_params, + .hw_free = spear_pcm_hw_free, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = spear_pcm_mmap, +}; + +static int +spear_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream, + size_t size) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + + dev_info(buf->dev.dev, + " preallocate_dma_buffer: area=%p, addr=%p, size=%d\n", + (void *)buf->area, (void *)buf->addr, size); + + buf->bytes = size; + return 0; +} + +static void spear_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf && !buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +static u64 spear_pcm_dmamask = DMA_BIT_MASK(32); + +static int spear_pcm_new(struct snd_card *card, + struct snd_soc_dai *dai, struct snd_pcm *pcm) +{ + int ret; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &spear_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + if (dai->driver->capture.channels_min) { + ret = spear_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE, + spear_pcm_hardware.buffer_bytes_max); + if (ret) + return ret; + } + + return 0; +} + +struct snd_soc_platform_driver spear_soc_platform = { + .ops = &spear_pcm_ops, + .pcm_new = spear_pcm_new, + .pcm_free = spear_pcm_free, +}; + +static int __devinit spear_soc_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &spear_soc_platform); +} + +static int __devexit spear_soc_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + + return 0; +} + +static struct platform_driver spear_pcm_driver = { + .driver = { + .name = "spear-pcm-audio", + .owner = THIS_MODULE, + }, + + .probe = spear_soc_platform_probe, + .remove = __devexit_p(spear_soc_platform_remove), +}; + +module_platform_driver(spear_pcm_driver); + +MODULE_AUTHOR("Rajeev Kumar "); +MODULE_DESCRIPTION("SPEAr PCM DMA module"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spear-pcm-audio"); -- cgit v0.10.2 From ace36d85809f6005b559802f194d44c6aa61af88 Mon Sep 17 00:00:00 2001 From: Vipin Kumar Date: Thu, 21 Jun 2012 15:54:53 +0530 Subject: ASoC: SPEAr spdif_in: Add spdif IN support This patch implements the spdif IN driver for ST peripheral Signed-off-by: Vipin Kumar Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/include/sound/spear_spdif.h b/include/sound/spear_spdif.h new file mode 100644 index 0000000..a12f396 --- /dev/null +++ b/include/sound/spear_spdif.h @@ -0,0 +1,29 @@ +/* + * Copyright (ST) 2012 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef __SOUND_SPDIF_H +#define __SOUND_SPDIF_H + +struct spear_spdif_platform_data { + /* DMA params */ + void *dma_params; + bool (*filter)(struct dma_chan *chan, void *slave); + void (*reset_perip)(void); +}; + +#endif /* SOUND_SPDIF_H */ diff --git a/sound/soc/spear/spdif_in.c b/sound/soc/spear/spdif_in.c new file mode 100644 index 0000000..c7c4b20 --- /dev/null +++ b/sound/soc/spear/spdif_in.c @@ -0,0 +1,297 @@ +/* + * ALSA SoC SPDIF In Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "spdif_in_regs.h" + +struct spdif_in_params { + u32 format; +}; + +struct spdif_in_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_in_params saved_params; + void *io_base; + struct device *dev; + void (*reset_perip)(void); + int irq; +}; + +static void spdif_in_configure(struct spdif_in_dev *host) +{ + u32 ctrl = SPDIF_IN_PRTYEN | SPDIF_IN_STATEN | SPDIF_IN_USREN | + SPDIF_IN_VALEN | SPDIF_IN_BLKEN; + ctrl |= SPDIF_MODE_16BIT | SPDIF_FIFO_THRES_16; + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); +} + +static int spdif_in_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + return 0; +} + +static void spdif_in_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return; + + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_in_format(struct spdif_in_dev *host, u32 format) +{ + u32 ctrl = readl(host->io_base + SPDIF_IN_CTRL); + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + ctrl |= SPDIF_XTRACT_16BIT; + break; + + case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: + ctrl &= ~SPDIF_XTRACT_16BIT; + break; + } + + writel(ctrl, host->io_base + SPDIF_IN_CTRL); +} + +static int spdif_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 format; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + format = params_format(params); + host->saved_params.format = format; + + return 0; +} + +static int spdif_in_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_in_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + clk_enable(host->clk); + spdif_in_configure(host); + spdif_in_format(host, host->saved_params.format); + + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl |= SPDIF_IN_SAMPLE | SPDIF_IN_ENB; + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0xF, host->io_base + SPDIF_IN_IRQ_MASK); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_IN_CTRL); + ctrl &= ~(SPDIF_IN_SAMPLE | SPDIF_IN_ENB); + writel(ctrl, host->io_base + SPDIF_IN_CTRL); + writel(0x0, host->io_base + SPDIF_IN_IRQ_MASK); + + if (host->reset_perip) + host->reset_perip(); + clk_disable(host->clk); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static struct snd_soc_dai_ops spdif_in_dai_ops = { + .startup = spdif_in_startup, + .shutdown = spdif_in_shutdown, + .trigger = spdif_in_trigger, + .hw_params = spdif_in_hw_params, +}; + +struct snd_soc_dai_driver spdif_in_dai = { + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE, + }, + .ops = &spdif_in_dai_ops, +}; + +static irqreturn_t spdif_in_irq(int irq, void *arg) +{ + struct spdif_in_dev *host = (struct spdif_in_dev *)arg; + + u32 irq_status = readl(host->io_base + SPDIF_IN_IRQ); + + if (!irq_status) + return IRQ_NONE; + + if (irq_status & SPDIF_IRQ_FIFOWRITE) + dev_err(host->dev, "spdif in: fifo write error"); + if (irq_status & SPDIF_IRQ_EMPTYFIFOREAD) + dev_err(host->dev, "spdif in: empty fifo read error"); + if (irq_status & SPDIF_IRQ_FIFOFULL) + dev_err(host->dev, "spdif in: fifo full error"); + if (irq_status & SPDIF_IRQ_OUTOFRANGE) + dev_err(host->dev, "spdif in: out of range error"); + + writel(0, host->io_base + SPDIF_IN_IRQ); + + return IRQ_HANDLED; +} + +static int spdif_in_probe(struct platform_device *pdev) +{ + struct spdif_in_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res, *res_fifo; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + res_fifo = platform_get_resource(pdev, IORESOURCE_IO, 0); + if (!res_fifo) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->irq = platform_get_irq(pdev, 0); + if (host->irq < 0) + return -EINVAL; + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + if (!pdata) + return -EINVAL; + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res_fifo->start; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + host->reset_perip = pdata->reset_perip; + + host->dev = &pdev->dev; + dev_set_drvdata(&pdev->dev, host); + + ret = devm_request_irq(&pdev->dev, host->irq, spdif_in_irq, 0, + "spdif-in", host); + if (ret) { + clk_put(host->clk); + dev_warn(&pdev->dev, "request_irq failed\n"); + return ret; + } + + ret = snd_soc_register_dai(&pdev->dev, &spdif_in_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_in_remove(struct platform_device *pdev) +{ + struct spdif_in_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + + +static struct platform_driver spdif_in_driver = { + .probe = spdif_in_probe, + .remove = spdif_in_remove, + .driver = { + .name = "spdif-in", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_in_driver); + +MODULE_AUTHOR("Vipin Kumar "); +MODULE_DESCRIPTION("SPEAr SPDIF IN SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_in"); diff --git a/sound/soc/spear/spdif_in_regs.h b/sound/soc/spear/spdif_in_regs.h new file mode 100644 index 0000000..37af7bc --- /dev/null +++ b/sound/soc/spear/spdif_in_regs.h @@ -0,0 +1,60 @@ +/* + * SPEAr SPDIF IN controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_IN_REGS_H +#define SPDIF_IN_REGS_H + +#define SPDIF_IN_CTRL 0x00 + #define SPDIF_IN_PRTYEN (1 << 20) + #define SPDIF_IN_STATEN (1 << 19) + #define SPDIF_IN_USREN (1 << 18) + #define SPDIF_IN_VALEN (1 << 17) + #define SPDIF_IN_BLKEN (1 << 16) + + #define SPDIF_MODE_24BIT (8 << 12) + #define SPDIF_MODE_23BIT (7 << 12) + #define SPDIF_MODE_22BIT (6 << 12) + #define SPDIF_MODE_21BIT (5 << 12) + #define SPDIF_MODE_20BIT (4 << 12) + #define SPDIF_MODE_19BIT (3 << 12) + #define SPDIF_MODE_18BIT (2 << 12) + #define SPDIF_MODE_17BIT (1 << 12) + #define SPDIF_MODE_16BIT (0 << 12) + #define SPDIF_MODE_MASK (0x0F << 12) + + #define SPDIF_IN_VALID (1 << 11) + #define SPDIF_IN_SAMPLE (1 << 10) + #define SPDIF_DATA_SWAP (1 << 9) + #define SPDIF_IN_ENB (1 << 8) + #define SPDIF_DATA_REVERT (1 << 7) + #define SPDIF_XTRACT_16BIT (1 << 6) + #define SPDIF_FIFO_THRES_16 (16 << 0) + +#define SPDIF_IN_IRQ_MASK 0x04 +#define SPDIF_IN_IRQ 0x08 + #define SPDIF_IRQ_FIFOWRITE (1 << 0) + #define SPDIF_IRQ_EMPTYFIFOREAD (1 << 1) + #define SPDIF_IRQ_FIFOFULL (1 << 2) + #define SPDIF_IRQ_OUTOFRANGE (1 << 3) + +#define SPDIF_IN_STA 0x0C + #define SPDIF_IN_LOCK (0x1 << 0) + +#endif /* SPDIF_IN_REGS_H */ -- cgit v0.10.2 From b2a4ec3d48fb53c99cb2e332f6e58eef770f1ed9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 11:34:49 +0100 Subject: ASoC: da732x: Staticise non-exported symbol soc_codec_dev_da732x Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 04af369..01be2a3 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1543,7 +1543,7 @@ static int da732x_remove(struct snd_soc_codec *codec) return 0; } -struct snd_soc_codec_driver soc_codec_dev_da732x = { +static struct snd_soc_codec_driver soc_codec_dev_da732x = { .probe = da732x_probe, .remove = da732x_remove, .set_bias_level = da732x_set_bias_level, -- cgit v0.10.2 From bb1591b3de7c9c6b28f337e214100a394a126ab2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 11:33:38 +0100 Subject: ASoC: isabelle: Staticise non-exported isabelle_dai Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 0d62f3b..5d8f39e 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1036,7 +1036,7 @@ static struct snd_soc_dai_ops isabelle_ul_dai_ops = { }; /* ISABELLE dai structure */ -struct snd_soc_dai_driver isabelle_dai[] = { +static struct snd_soc_dai_driver isabelle_dai[] = { { .name = "isabelle-dl1", .playback = { -- cgit v0.10.2 From 229e3fdc1ba49b747e9434b55b3f6bd68a4db251 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 11:40:55 +0100 Subject: ASoC: core: Add DOUBLE_R variants of the _RANGE controls The code handles this fine already, we just need new macros in the header for drivers to create the controls. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc.h b/include/sound/soc.h index e4348d2..e063380 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -42,6 +42,10 @@ ((unsigned long)&(struct soc_mixer_control) \ {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ .max = xmax, .platform_max = xmax, .invert = xinvert}) +#define SOC_DOUBLE_R_RANGE_VALUE(xlreg, xrreg, xshift, xmin, xmax, xinvert) \ + ((unsigned long)&(struct soc_mixer_control) \ + {.reg = xlreg, .rreg = xrreg, .shift = xshift, .rshift = xshift, \ + .min = xmin, .max = xmax, .platform_max = xmax, .invert = xinvert}) #define SOC_SINGLE(xname, reg, shift, max, invert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_volsw, .get = snd_soc_get_volsw,\ @@ -96,6 +100,13 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_TLV(xname, reg, shift_left, shift_right, max, invert, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ @@ -114,6 +125,16 @@ .get = snd_soc_get_volsw, .put = snd_soc_put_volsw, \ .private_value = SOC_DOUBLE_R_VALUE(reg_left, reg_right, xshift, \ xmax, xinvert) } +#define SOC_DOUBLE_R_RANGE_TLV(xname, reg_left, reg_right, xshift, xmin, \ + xmax, xinvert, tlv_array) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname),\ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |\ + SNDRV_CTL_ELEM_ACCESS_READWRITE,\ + .tlv.p = (tlv_array), \ + .info = snd_soc_info_volsw_range, \ + .get = snd_soc_get_volsw_range, .put = snd_soc_put_volsw_range, \ + .private_value = SOC_DOUBLE_R_RANGE_VALUE(reg_left, reg_right, \ + xshift, xmin, xmax, xinvert) } #define SOC_DOUBLE_R_SX_TLV(xname, xreg, xrreg, xshift, xmin, xmax, tlv_array) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \ -- cgit v0.10.2 From 62d4a4b99dfd647ef88b8434334eaa7497602857 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 12:21:49 +0100 Subject: ASoC: dapm: Try to add all routes even if one fails We may as well print as many errors as we can in one go rather than requiring developers to iterate through all their typos. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 7365fed..32fbf10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2276,15 +2276,15 @@ err: int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num) { - int i, ret = 0; + int i, r, ret = 0; mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); for (i = 0; i < num; i++) { - ret = snd_soc_dapm_add_route(dapm, route); - if (ret < 0) { + r = snd_soc_dapm_add_route(dapm, route); + if (r < 0) { dev_err(dapm->dev, "Failed to add route %s->%s\n", route->source, route->sink); - break; + ret = r; } route++; } -- cgit v0.10.2 From 9dfdd5abcf2b350d4fdb207c0dff3194e2fd73db Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 12:40:52 +0100 Subject: ASoC: io: Don't dereference regmap if we failed to get one Avoids a crash in invalid configurations. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 44d0174..29183ef 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -145,10 +145,13 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, if (!codec->control_data) codec->control_data = dev_get_regmap(codec->dev, NULL); - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte multiples */ - if (ret > 0) - codec->val_bytes = ret; + if (codec->control_data) { + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; + } break; default: -- cgit v0.10.2 From 07ed873e4c975a26c327a6bd306693678ef63351 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 18 Jun 2012 21:08:44 +0100 Subject: ASoC: Add shared code for Wolfson Arizona class devices The Wolfson Arizona series of audio hub CODECs can share a large amount of their driver code as the result of a common register map. This patch adds some of this core support, providing a basis for the initial WM5102 audio driver. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 43f5240..2ae8082 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -129,6 +129,11 @@ config SND_SOC_ALL_CODECS config SND_SOC_88PM860X tristate +config SND_SOC_ARIZONA + tristate + default y if SND_SOC_WM5102=y + default m if SND_SOC_WM5102=m + config SND_SOC_WM_HUBS tristate default y if SND_SOC_WM8993=y || SND_SOC_WM8994=y diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3d30654..3005ea6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o snd-soc-cs42l52-objs := cs42l52.o @@ -128,6 +129,7 @@ obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o +obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o obj-$(CONFIG_SND_SOC_CQ0093VC) += snd-soc-cq93vc.o obj-$(CONFIG_SND_SOC_CS42L51) += snd-soc-cs42l51.o obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c new file mode 100644 index 0000000..3b5730b --- /dev/null +++ b/sound/soc/codecs/arizona.c @@ -0,0 +1,781 @@ +/* + * arizona.c - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" + +#define ARIZONA_AIF_BCLK_CTRL 0x00 +#define ARIZONA_AIF_TX_PIN_CTRL 0x01 +#define ARIZONA_AIF_RX_PIN_CTRL 0x02 +#define ARIZONA_AIF_RATE_CTRL 0x03 +#define ARIZONA_AIF_FORMAT 0x04 +#define ARIZONA_AIF_TX_BCLK_RATE 0x05 +#define ARIZONA_AIF_RX_BCLK_RATE 0x06 +#define ARIZONA_AIF_FRAME_CTRL_1 0x07 +#define ARIZONA_AIF_FRAME_CTRL_2 0x08 +#define ARIZONA_AIF_FRAME_CTRL_3 0x09 +#define ARIZONA_AIF_FRAME_CTRL_4 0x0A +#define ARIZONA_AIF_FRAME_CTRL_5 0x0B +#define ARIZONA_AIF_FRAME_CTRL_6 0x0C +#define ARIZONA_AIF_FRAME_CTRL_7 0x0D +#define ARIZONA_AIF_FRAME_CTRL_8 0x0E +#define ARIZONA_AIF_FRAME_CTRL_9 0x0F +#define ARIZONA_AIF_FRAME_CTRL_10 0x10 +#define ARIZONA_AIF_FRAME_CTRL_11 0x11 +#define ARIZONA_AIF_FRAME_CTRL_12 0x12 +#define ARIZONA_AIF_FRAME_CTRL_13 0x13 +#define ARIZONA_AIF_FRAME_CTRL_14 0x14 +#define ARIZONA_AIF_FRAME_CTRL_15 0x15 +#define ARIZONA_AIF_FRAME_CTRL_16 0x16 +#define ARIZONA_AIF_FRAME_CTRL_17 0x17 +#define ARIZONA_AIF_FRAME_CTRL_18 0x18 +#define ARIZONA_AIF_TX_ENABLES 0x19 +#define ARIZONA_AIF_RX_ENABLES 0x1A +#define ARIZONA_AIF_FORCE_WRITE 0x1B + +#define arizona_fll_err(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_warn(_fll, fmt, ...) \ + dev_warn(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) +#define arizona_fll_dbg(_fll, fmt, ...) \ + dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) + +#define arizona_aif_err(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_warn(_dai, fmt, ...) \ + dev_warn(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +#define arizona_aif_dbg(_dai, fmt, ...) \ + dev_err(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) + +const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { + "None", + "Tone Generator 1", + "Tone Generator 2", + "Haptics", + "AEC", + "Mic Mute Mixer", + "Noise Generator", + "IN1L", + "IN1R", + "IN2L", + "IN2R", + "IN3L", + "IN3R", + "AIF1RX1", + "AIF1RX2", + "AIF1RX3", + "AIF1RX4", + "AIF1RX5", + "AIF1RX6", + "AIF1RX7", + "AIF1RX8", + "AIF2RX1", + "AIF2RX2", + "AIF3RX1", + "AIF3RX2", + "SLIMRX1", + "SLIMRX2", + "SLIMRX3", + "SLIMRX4", + "SLIMRX5", + "SLIMRX6", + "SLIMRX7", + "SLIMRX8", + "EQ1", + "EQ2", + "EQ3", + "EQ4", + "DRC1L", + "DRC1R", + "DRC2L", + "DRC2R", + "LHPF1", + "LHPF2", + "LHPF3", + "LHPF4", + "DSP1.1", + "DSP1.2", + "DSP1.3", + "DSP1.4", + "DSP1.5", + "DSP1.6", + "ASRC1L", + "ASRC1R", + "ASRC2L", + "ASRC2R", +}; +EXPORT_SYMBOL_GPL(arizona_mixer_texts); + +int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { + 0x00, /* None */ + 0x04, /* Tone */ + 0x05, + 0x06, /* Haptics */ + 0x08, /* AEC */ + 0x0c, /* Noise mixer */ + 0x0d, /* Comfort noise */ + 0x10, /* IN1L */ + 0x11, + 0x12, + 0x13, + 0x14, + 0x15, + 0x20, /* AIF1RX1 */ + 0x21, + 0x22, + 0x23, + 0x24, + 0x25, + 0x26, + 0x27, + 0x28, /* AIF2RX1 */ + 0x29, + 0x30, /* AIF3RX1 */ + 0x31, + 0x38, /* SLIMRX1 */ + 0x39, + 0x3a, + 0x3b, + 0x3c, + 0x3d, + 0x3e, + 0x3f, + 0x50, /* EQ1 */ + 0x51, + 0x52, + 0x53, + 0x58, /* DRC1L */ + 0x59, + 0x5a, + 0x5b, + 0x60, /* LHPF1 */ + 0x61, + 0x62, + 0x63, + 0x68, /* DSP1.1 */ + 0x69, + 0x6a, + 0x6b, + 0x6c, + 0x6d, + 0x90, /* ASRC1L */ + 0x91, + 0x92, + 0x93, +}; +EXPORT_SYMBOL_GPL(arizona_mixer_values); + +const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); +EXPORT_SYMBOL_GPL(arizona_mixer_tlv); + +static const char *arizona_lhpf_mode_text[] = { + "Low-pass", "High-pass" +}; + +const struct soc_enum arizona_lhpf1_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); + +const struct soc_enum arizona_lhpf2_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); + +const struct soc_enum arizona_lhpf3_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); + +const struct soc_enum arizona_lhpf4_mode = + SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, + arizona_lhpf_mode_text); +EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); + +int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_in_ev); + +int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + return 0; +} +EXPORT_SYMBOL_GPL(arizona_out_ev); + +int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + char *name; + unsigned int reg; + unsigned int mask = ARIZONA_SYSCLK_FREQ_MASK | ARIZONA_SYSCLK_SRC_MASK; + unsigned int val = source << ARIZONA_SYSCLK_SRC_SHIFT; + unsigned int *clk; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + name = "SYSCLK"; + reg = ARIZONA_SYSTEM_CLOCK_1; + clk = &priv->sysclk; + mask |= ARIZONA_SYSCLK_FRAC; + break; + case ARIZONA_CLK_ASYNCCLK: + name = "ASYNCCLK"; + reg = ARIZONA_ASYNC_CLOCK_1; + clk = &priv->asyncclk; + break; + default: + return -EINVAL; + } + + switch (freq) { + case 5644800: + case 6144000: + break; + case 11289600: + case 12288000: + val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 22579200: + case 24576000: + val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + case 45158400: + case 49152000: + val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + break; + default: + return -EINVAL; + } + + *clk = freq; + + if (freq % 6144000) + val |= ARIZONA_SYSCLK_FRAC; + + dev_dbg(arizona->dev, "%s set to %uHz", name, freq); + + return regmap_update_bits(arizona->regmap, reg, mask, val); +} +EXPORT_SYMBOL_GPL(arizona_set_sysclk); + +static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + int lrclk, bclk, mode, base; + + base = dai->driver->base; + + lrclk = 0; + bclk = 0; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + mode = 0; + break; + case SND_SOC_DAIFMT_DSP_B: + mode = 1; + break; + case SND_SOC_DAIFMT_I2S: + mode = 2; + break; + case SND_SOC_DAIFMT_LEFT_J: + mode = 3; + break; + default: + arizona_aif_err(dai, "Unsupported DAI format %d\n", + fmt & SND_SOC_DAIFMT_FORMAT_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + break; + case SND_SOC_DAIFMT_CBS_CFM: + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + break; + case SND_SOC_DAIFMT_CBM_CFM: + bclk |= ARIZONA_AIF1_BCLK_MSTR; + lrclk |= ARIZONA_AIF1TX_LRCLK_MSTR; + break; + default: + arizona_aif_err(dai, "Unsupported master mode %d\n", + fmt & SND_SOC_DAIFMT_MASTER_MASK); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + bclk |= ARIZONA_AIF1_BCLK_INV; + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + bclk |= ARIZONA_AIF1_BCLK_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + lrclk |= ARIZONA_AIF1TX_LRCLK_INV; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_INV | ARIZONA_AIF1_BCLK_MSTR, + bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_PIN_CTRL, + ARIZONA_AIF1TX_LRCLK_INV | + ARIZONA_AIF1TX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_PIN_CTRL, + ARIZONA_AIF1RX_LRCLK_INV | + ARIZONA_AIF1RX_LRCLK_MSTR, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FORMAT, + ARIZONA_AIF1_FMT_MASK, mode); + + return 0; +} + +static const int arizona_48k_rates[] = { + -1, + 48000, + 64000, + 96000, + 128000, + 192000, + 256000, + 384000, + 512000, + 768000, + 1024000, + 1536000, + 2048000, + 3072000, + 4096000, + 6144000, + 8192000, + 12288000, + 24576000, +}; + +static const int arizona_44k1_rates[] = { + -1, + 44100, + 58800, + 88200, + 117600, + 177640, + 235200, + 352800, + 470400, + 705600, + 940800, + 1411200, + 1881600, + 2882400, + 3763200, + 5644800, + 7526400, + 11289600, + 22579200, +}; + +static int arizona_sr_vals[] = { + 0, + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 0, + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static int arizona_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int base = dai->driver->base; + const int *rates; + int i; + int bclk, lrclk, wl, frame, sr_val; + + if (params_rate(params) % 8000) + rates = &arizona_44k1_rates[0]; + else + rates = &arizona_48k_rates[0]; + + for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + if (rates[i] == snd_soc_params_to_bclk(params)) { + bclk = i; + break; + } + } + if (i == ARRAY_SIZE(arizona_44k1_rates)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for the chip. + */ + for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) + if (arizona_sr_vals[i] == params_rate(params)) + break; + if (i == ARRAY_SIZE(arizona_sr_vals)) { + arizona_aif_err(dai, "Unsupported sample rate %dHz\n", + params_rate(params)); + return -EINVAL; + } + sr_val = i; + + lrclk = snd_soc_params_to_bclk(params) / params_rate(params); + + arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", + rates[bclk], rates[bclk] / lrclk); + + wl = snd_pcm_format_width(params_format(params)); + frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; + + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, + ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, + ARIZONA_AIF1TX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RX_BCLK_RATE, + ARIZONA_AIF1RX_BCPF_MASK, lrclk); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_1, + ARIZONA_AIF1TX_WL_MASK | + ARIZONA_AIF1TX_SLOT_LEN_MASK, frame); + snd_soc_update_bits(codec, base + ARIZONA_AIF_FRAME_CTRL_2, + ARIZONA_AIF1RX_WL_MASK | + ARIZONA_AIF1RX_SLOT_LEN_MASK, frame); + + return 0; +} + +const struct snd_soc_dai_ops arizona_dai_ops = { + .set_fmt = arizona_set_fmt, + .hw_params = arizona_hw_params, +}; + +static irqreturn_t arizona_fll_lock(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "Locked\n"); + + complete(&fll->lock); + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_fll_clock_ok(int irq, void *data) +{ + struct arizona_fll *fll = data; + + arizona_fll_dbg(fll, "clock OK\n"); + + complete(&fll->ok); + + return IRQ_HANDLED; +} + +static struct { + unsigned int min; + unsigned int max; + u16 fratio; + int ratio; +} fll_fratios[] = { + { 0, 64000, 4, 16 }, + { 64000, 128000, 3, 8 }, + { 128000, 256000, 2, 4 }, + { 256000, 1000000, 1, 2 }, + { 1000000, 13500000, 0, 1 }, +}; + +struct arizona_fll_cfg { + int n; + int theta; + int lambda; + int refdiv; + int outdiv; + int fratio; +}; + +static int arizona_calc_fll(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int target, div, gcd_fll; + int i, ratio; + + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + + /* Fref must be <=13.5MHz */ + div = 1; + cfg->refdiv = 0; + while ((Fref / div) > 13500000) { + div *= 2; + cfg->refdiv++; + + if (div > 8) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + } + + /* Apply the division for our remaining calculations */ + Fref /= div; + + /* Fvco should be 90-100MHz; don't check the upper bound */ + div = 1; + while (Fout * div < 90000000) { + div++; + if (div > 7) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + } + target = Fout * div; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + + /* Find an appropraite FLL_FRATIO and factor it out of the target */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + cfg->fratio = fll_fratios[i].fratio; + ratio = fll_fratios[i].ratio; + break; + } + } + if (i == ARRAY_SIZE(fll_fratios)) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return -EINVAL; + } + + cfg->n = target / (ratio * Fref); + + if (target % Fref) { + gcd_fll = gcd(target, ratio * Fref); + arizona_fll_dbg(fll, "GCD=%u\n", gcd_fll); + + cfg->theta = (target - (cfg->n * ratio * Fref)) + / gcd_fll; + cfg->lambda = (ratio * Fref) / gcd_fll; + } else { + cfg->theta = 0; + cfg->lambda = 0; + } + + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", + cfg->n, cfg->theta, cfg->lambda); + arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", + cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + + return 0; + +} + +static void arizona_apply_fll(struct arizona *arizona, unsigned int base, + struct arizona_fll_cfg *cfg, int source) +{ + regmap_update_bits(arizona->regmap, base + 3, + ARIZONA_FLL1_THETA_MASK, cfg->theta); + regmap_update_bits(arizona->regmap, base + 4, + ARIZONA_FLL1_LAMBDA_MASK, cfg->lambda); + regmap_update_bits(arizona->regmap, base + 5, + ARIZONA_FLL1_FRATIO_MASK, + cfg->fratio << ARIZONA_FLL1_FRATIO_SHIFT); + regmap_update_bits(arizona->regmap, base + 6, + ARIZONA_FLL1_CLK_REF_DIV_MASK | + ARIZONA_FLL1_CLK_REF_SRC_MASK, + cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | + source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); + + regmap_update_bits(arizona->regmap, base + 2, + ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, + ARIZONA_FLL1_CTRL_UPD | cfg->n); +} + +int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout) +{ + struct arizona *arizona = fll->arizona; + struct arizona_fll_cfg cfg, sync; + unsigned int reg, val; + int syncsrc; + bool ena; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, ®); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + ena = reg & ARIZONA_FLL1_ENA; + + if (Fout) { + /* Do we have a 32kHz reference? */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + syncsrc = -1; + } + + if (source == syncsrc) + syncsrc = -1; + + if (syncsrc >= 0) { + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + if (ret != 0) + return ret; + } else { + ret = arizona_calc_fll(fll, &cfg, Fref, Fout); + if (ret != 0) + return ret; + } + } else { + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (ena) + pm_runtime_put_autosuspend(arizona->dev); + + return 0; + } + + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + + if (syncsrc >= 0) { + arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); + arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); + } else { + arizona_apply_fll(arizona, fll->base, &cfg, source); + } + + if (!ena) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (syncsrc >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(25)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll); + +int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, + int ok_irq, struct arizona_fll *fll) +{ + int ret; + + init_completion(&fll->lock); + init_completion(&fll->ok); + + fll->id = id; + fll->base = base; + fll->arizona = arizona; + + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); + snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), + "FLL%d clock OK", id); + + ret = arizona_request_irq(arizona, lock_irq, fll->lock_name, + arizona_fll_lock, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d lock IRQ: %d\n", + id, ret); + } + + ret = arizona_request_irq(arizona, ok_irq, fll->clock_ok_name, + arizona_fll_clock_ok, fll); + if (ret != 0) { + dev_err(arizona->dev, "Failed to get FLL%d clock OK IRQ: %d\n", + id, ret); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_fll); + +MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h new file mode 100644 index 0000000..8c2ca1d --- /dev/null +++ b/sound/soc/codecs/arizona.h @@ -0,0 +1,149 @@ +/* + * arizona.h - Wolfson Arizona class device shared support + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _ASOC_ARIZONA_H +#define _ASOC_ARIZONA_H + +#include + +#include + +#define ARIZONA_CLK_SYSCLK 1 +#define ARIZONA_CLK_ASYNCCLK 2 + +#define ARIZONA_CLK_SRC_MCLK1 0x0 +#define ARIZONA_CLK_SRC_MCLK2 0x1 +#define ARIZONA_CLK_SRC_FLL1 0x4 +#define ARIZONA_CLK_SRC_FLL2 0x5 +#define ARIZONA_CLK_SRC_AIF1BCLK 0x8 +#define ARIZONA_CLK_SRC_AIF2BCLK 0x9 +#define ARIZONA_CLK_SRC_AIF3BCLK 0xa + +#define ARIZONA_FLL_SRC_MCLK1 0 +#define ARIZONA_FLL_SRC_MCLK2 1 +#define ARIZONA_FLL_SRC_SLIMCLK 2 +#define ARIZONA_FLL_SRC_FLL1 3 +#define ARIZONA_FLL_SRC_FLL2 4 +#define ARIZONA_FLL_SRC_AIF1BCLK 5 +#define ARIZONA_FLL_SRC_AIF2BCLK 6 +#define ARIZONA_FLL_SRC_AIF3BCLK 7 +#define ARIZONA_FLL_SRC_AIF1LRCLK 8 +#define ARIZONA_FLL_SRC_AIF2LRCLK 9 +#define ARIZONA_FLL_SRC_AIF3LRCLK 10 + +#define ARIZONA_MIXER_VOL_MASK 0x00FE +#define ARIZONA_MIXER_VOL_SHIFT 1 +#define ARIZONA_MIXER_VOL_WIDTH 7 + +struct arizona; + +struct arizona_priv { + struct arizona *arizona; + int sysclk; + int asyncclk; +}; + +#define ARIZONA_NUM_MIXER_INPUTS 55 + +extern const unsigned int arizona_mixer_tlv[]; +extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; +extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; + +#define ARIZONA_MIXER_CONTROLS(name, base) \ + SOC_SINGLE_RANGE_TLV(name " Input 1 Volume", base + 1, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 2 Volume", base + 3, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 3 Volume", base + 5, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv), \ + SOC_SINGLE_RANGE_TLV(name " Input 4 Volume", base + 7, \ + ARIZONA_MIXER_VOL_SHIFT, 0x20, 0x50, 0, \ + arizona_mixer_tlv) + +#define ARIZONA_MUX_ENUM_DECL(name, reg) \ + SOC_VALUE_ENUM_SINGLE_DECL(name, reg, 0, 0xff, \ + arizona_mixer_texts, arizona_mixer_values) + +#define ARIZONA_MUX_CTL_DECL(name) \ + const struct snd_kcontrol_new name##_mux = \ + SOC_DAPM_VALUE_ENUM("Route", name##_enum) + +#define ARIZONA_MIXER_ENUMS(name, base_reg) \ + static ARIZONA_MUX_ENUM_DECL(name##_in1_enum, base_reg); \ + static ARIZONA_MUX_ENUM_DECL(name##_in2_enum, base_reg + 2); \ + static ARIZONA_MUX_ENUM_DECL(name##_in3_enum, base_reg + 4); \ + static ARIZONA_MUX_ENUM_DECL(name##_in4_enum, base_reg + 6); \ + static ARIZONA_MUX_CTL_DECL(name##_in1); \ + static ARIZONA_MUX_CTL_DECL(name##_in2); \ + static ARIZONA_MUX_CTL_DECL(name##_in3); \ + static ARIZONA_MUX_CTL_DECL(name##_in4) + +#define ARIZONA_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +#define ARIZONA_MIXER_WIDGETS(name, name_str) \ + ARIZONA_MUX(name_str " Input 1", &name##_in1_mux), \ + ARIZONA_MUX(name_str " Input 2", &name##_in2_mux), \ + ARIZONA_MUX(name_str " Input 3", &name##_in3_mux), \ + ARIZONA_MUX(name_str " Input 4", &name##_in4_mux), \ + SND_SOC_DAPM_MIXER(name_str " Mixer", SND_SOC_NOPM, 0, 0, NULL, 0) + +#define ARIZONA_MIXER_ROUTES(widget, name) \ + { widget, NULL, name " Mixer" }, \ + { name " Mixer", NULL, name " Input 1" }, \ + { name " Mixer", NULL, name " Input 2" }, \ + { name " Mixer", NULL, name " Input 3" }, \ + { name " Mixer", NULL, name " Input 4" }, \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 1"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 2"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 3"), \ + ARIZONA_MIXER_INPUT_ROUTES(name " Input 4") + +extern const struct soc_enum arizona_lhpf1_mode; +extern const struct soc_enum arizona_lhpf2_mode; +extern const struct soc_enum arizona_lhpf3_mode; +extern const struct soc_enum arizona_lhpf4_mode; + +extern int arizona_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); +extern int arizona_out_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); + +extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir); + +extern const struct snd_soc_dai_ops arizona_dai_ops; + +#define ARIZONA_FLL_NAME_LEN 20 + +struct arizona_fll { + struct arizona *arizona; + int id; + unsigned int base; + struct completion lock; + struct completion ok; + + char lock_name[ARIZONA_FLL_NAME_LEN]; + char clock_ok_name[ARIZONA_FLL_NAME_LEN]; +}; + +extern int arizona_init_fll(struct arizona *arizona, int id, int base, + int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); + +#endif -- cgit v0.10.2 From 93e8791dd34ca0c3371d65c4488249d41de02776 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 19 Jun 2012 16:38:15 +0100 Subject: ASoC: wm5102: Initial driver The WM5102 is a highly-integrated low-power audio system for smartphones, tablets and other portable audio devices based on the Arizona platform. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2ae8082..1de24cc 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM2000 if I2C select SND_SOC_WM2200 if I2C select SND_SOC_WM5100 if I2C + select SND_SOC_WM5102 if MFD_WM5102 select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -330,6 +331,9 @@ config SND_SOC_WM2200 config SND_SOC_WM5100 tristate +config SND_SOC_WM5102 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3005ea6..d35ba7f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -63,6 +63,7 @@ snd-soc-wm1250-ev1-objs := wm1250-ev1.o snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o +snd-soc-wm5102-objs := wm5102.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -176,6 +177,7 @@ obj-$(CONFIG_SND_SOC_WM1250_EV1) += snd-soc-wm1250-ev1.o obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o +obj-$(CONFIG_SND_SOC_WM5102) += snd-soc-wm5102.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c new file mode 100644 index 0000000..9b9ea7f --- /dev/null +++ b/sound/soc/codecs/wm5102.c @@ -0,0 +1,870 @@ +/* + * wm5102.c -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" +#include "wm5102.h" + +struct wm5102_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5102_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_PGA_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_PGA_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_PGA_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" } + +static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, +}; + +static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5102_FLL1: + return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout); + case WM5102_FLL2: + return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5102_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5102_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5102_dai[] = { + { + .name = "wm5102-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5102-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5102_RATES, + .formats = WM5102_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5102_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5102_DIG_VU 0x0200 + +static unsigned int wm5102_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5102 = { + .probe = wm5102_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5102_set_fll, + + .controls = wm5102_snd_controls, + .num_controls = ARRAY_SIZE(wm5102_snd_controls), + .dapm_widgets = wm5102_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5102_dapm_widgets), + .dapm_routes = wm5102_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5102_dapm_routes), +}; + +static int __devinit wm5102_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5102_priv *wm5102; + int i; + + wm5102 = devm_kzalloc(&pdev->dev, sizeof(struct wm5102_priv), + GFP_KERNEL); + if (wm5102 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5102); + + wm5102->core.arizona = arizona; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5102->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5102->fll[1]); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], + WM5102_DIG_VU, WM5102_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5102, + wm5102_dai, ARRAY_SIZE(wm5102_dai)); +} + +static int __devexit wm5102_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5102_codec_driver = { + .driver = { + .name = "wm5102-codec", + .owner = THIS_MODULE, + }, + .probe = wm5102_probe, + .remove = __devexit_p(wm5102_remove), +}; + +module_platform_driver(wm5102_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5102 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5102-codec"); diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h new file mode 100644 index 0000000..d30477f --- /dev/null +++ b/sound/soc/codecs/wm5102.h @@ -0,0 +1,21 @@ +/* + * wm5102.h -- WM5102 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5102_H +#define _WM5102_H + +#include "arizona.h" + +#define WM5102_FLL1 1 +#define WM5102_FLL2 2 + +#endif -- cgit v0.10.2 From 1573ee81cb9ef24fa5acee6b7442e215e63ede2f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Uwe=20Kleine-K=C3=B6nig?= Date: Mon, 25 Jun 2012 09:28:46 +0200 Subject: ASoC: dmaengine_pcm: fix typo in comment MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Signed-off-by: Uwe Kleine-König Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dmaengine-pcm.c b/sound/soc/soc-dmaengine-pcm.c index 2995334d..5df529e 100644 --- a/sound/soc/soc-dmaengine-pcm.c +++ b/sound/soc/soc-dmaengine-pcm.c @@ -270,7 +270,7 @@ static int dmaengine_pcm_request_channel(struct dmaengine_pcm_runtime_data *prtd * Note that this function will use private_data field of the substream's * runtime. So it is not availabe to your pcm driver implementation. If you need * to keep additional data attached to a substream use - * snd_dmaeinge_pcm_{set,get}_data. + * snd_dmaengine_pcm_{set,get}_data. */ int snd_dmaengine_pcm_open(struct snd_pcm_substream *substream, dma_filter_fn filter_fn, void *filter_data) -- cgit v0.10.2 From fd88759a42dc10f8230b3933a1ceb40bd88fccea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:30:43 +0800 Subject: ASoC: wm8904: Rely entirely on the core for bias level management Even though the WM8904 is able to use idle_bias_off during both probe and resume we were needlessly leaving the device in standby mode. Instead power the device down as soon as we've confirmed that we can talk to it and don't manage the bias level at all over suspend and resume, the core will take us down to our minimum power level. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 5417b11..ecab871 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1945,25 +1945,6 @@ static struct snd_soc_dai_driver wm8904_dai = { .symmetric_rates = 1, }; -#ifdef CONFIG_PM -static int wm8904_suspend(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); - - return 0; -} - -static int wm8904_resume(struct snd_soc_codec *codec) -{ - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return 0; -} -#else -#define wm8904_suspend NULL -#define wm8904_resume NULL -#endif - static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -2143,7 +2124,10 @@ static int wm8904_probe(struct snd_soc_codec *codec) goto err_enable; } + /* Can leave the device powered off until we need it */ regcache_cache_only(wm8904->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_VU, WM8904_ADC_VU); @@ -2198,11 +2182,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_POBCTRL, 0); - wm8904_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* Bias level configuration will have done an extra enable */ - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); @@ -2220,7 +2199,6 @@ static int wm8904_remove(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - wm8904_set_bias_level(codec, SND_SOC_BIAS_OFF); regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); kfree(wm8904->retune_mobile_texts); kfree(wm8904->drc_texts); @@ -2231,8 +2209,6 @@ static int wm8904_remove(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_wm8904 = { .probe = wm8904_probe, .remove = wm8904_remove, - .suspend = wm8904_suspend, - .resume = wm8904_resume, .set_bias_level = wm8904_set_bias_level, .idle_bias_off = true, }; -- cgit v0.10.2 From 03862cf62ea36d6cf3d94eee84b89578cbcf0213 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:41:58 +0800 Subject: ASoC: wm8904: Move regulator acquisition and device identification to I2C It's more idiomatic to have the resource allocation at this level. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index ecab871..b178232 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -314,11 +314,6 @@ static bool wm8904_readable_register(struct device *dev, unsigned int reg) } } -static int wm8904_reset(struct snd_soc_codec *codec) -{ - return snd_soc_write(codec, WM8904_SW_RESET_AND_ID, 0); -} - static int wm8904_configure_clocking(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); @@ -2082,52 +2077,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) return ret; } - for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) - wm8904->supplies[i].supply = wm8904_supply_names[i]; - - ret = regulator_bulk_get(codec->dev, ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to request supplies: %d\n", ret); - return ret; - } - - ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), - wm8904->supplies); - if (ret != 0) { - dev_err(codec->dev, "Failed to enable supplies: %d\n", ret); - goto err_get; - } - - ret = snd_soc_read(codec, WM8904_SW_RESET_AND_ID); - if (ret < 0) { - dev_err(codec->dev, "Failed to read ID register\n"); - goto err_enable; - } - if (ret != 0x8904) { - dev_err(codec->dev, "Device is not a WM8904, ID is %x\n", ret); - ret = -EINVAL; - goto err_enable; - } - - ret = snd_soc_read(codec, WM8904_REVISION); - if (ret < 0) { - dev_err(codec->dev, "Failed to read device revision: %d\n", - ret); - goto err_enable; - } - dev_info(codec->dev, "revision %c\n", ret + 'A'); - - ret = wm8904_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - goto err_enable; - } - - /* Can leave the device powered off until we need it */ - regcache_cache_only(wm8904->regmap, true); - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, WM8904_ADC_VU, WM8904_ADC_VU); @@ -2187,19 +2136,12 @@ static int wm8904_probe(struct snd_soc_codec *codec) wm8904_add_widgets(codec); return 0; - -err_enable: - regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); -err_get: - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); - return ret; } static int wm8904_remove(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - regulator_bulk_free(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); kfree(wm8904->retune_mobile_texts); kfree(wm8904->drc_texts); @@ -2230,7 +2172,8 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm8904_priv *wm8904; - int ret; + unsigned int val; + int ret, i; wm8904 = devm_kzalloc(&i2c->dev, sizeof(struct wm8904_priv), GFP_KERNEL); @@ -2249,14 +2192,61 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, i2c_set_clientdata(i2c, wm8904); wm8904->pdata = i2c->dev.platform_data; + for (i = 0; i < ARRAY_SIZE(wm8904->supplies); i++) + wm8904->supplies[i].supply = wm8904_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(wm8904->supplies), + wm8904->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + ret = regmap_read(wm8904->regmap, WM8904_SW_RESET_AND_ID, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read ID register: %d\n", ret); + goto err_enable; + } + if (val != 0x8904) { + dev_err(&i2c->dev, "Device is not a WM8904, ID is %x\n", val); + ret = -EINVAL; + goto err_enable; + } + + ret = regmap_read(wm8904->regmap, WM8904_REVISION, &val); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to read device revision: %d\n", + ret); + goto err_enable; + } + dev_info(&i2c->dev, "revision %c\n", val + 'A'); + + ret = regmap_write(wm8904->regmap, WM8904_SW_RESET_AND_ID, 0); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_enable; + } + + /* Can leave the device powered off until we need it */ + regcache_cache_only(wm8904->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm8904, &wm8904_dai, 1); if (ret != 0) - goto err; + return ret; return 0; -err: +err_enable: + regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); return ret; } -- cgit v0.10.2 From 725e7a7b58fb27d8f97a1c4eae47cb5d37564725 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:57:37 +0800 Subject: ASoC: wm8904: Move register default setup into I2C probe() Get it done as early as possible, it's neater and minimises the time the pins aren't configured as requested. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index b178232..0013afe 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2054,8 +2054,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - struct wm8904_pdata *pdata = wm8904->pdata; - int ret, i; + int ret; codec->control_data = wm8904->regmap; @@ -2077,60 +2076,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) return ret; } - /* Change some default settings - latch VU and enable ZC */ - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_LEFT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_ADC_DIGITAL_VOLUME_RIGHT, - WM8904_ADC_VU, WM8904_ADC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_LEFT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_DAC_DIGITAL_VOLUME_RIGHT, - WM8904_DAC_VU, WM8904_DAC_VU); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_LEFT, - WM8904_HPOUT_VU | WM8904_HPOUTLZC, - WM8904_HPOUT_VU | WM8904_HPOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT1_RIGHT, - WM8904_HPOUT_VU | WM8904_HPOUTRZC, - WM8904_HPOUT_VU | WM8904_HPOUTRZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_LEFT, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); - snd_soc_update_bits(codec, WM8904_ANALOGUE_OUT2_RIGHT, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, - WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); - snd_soc_update_bits(codec, WM8904_CLOCK_RATES_0, - WM8904_SR_MODE, 0); - - /* Apply configuration from the platform data. */ - if (wm8904->pdata) { - for (i = 0; i < WM8904_GPIO_REGS; i++) { - if (!pdata->gpio_cfg[i]) - continue; - - regmap_update_bits(wm8904->regmap, - WM8904_GPIO_CONTROL_1 + i, - 0xffff, - pdata->gpio_cfg[i]); - } - - /* Zero is the default value for these anyway */ - for (i = 0; i < WM8904_MIC_REGS; i++) - regmap_update_bits(wm8904->regmap, - WM8904_MIC_BIAS_CONTROL_0 + i, - 0xffff, - pdata->mic_cfg[i]); - } - - /* Set Class W by default - this will be managed by the Class - * G widget at runtime where bypass paths are available. - */ - snd_soc_update_bits(codec, WM8904_CLASS_W_0, - WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); - - /* Use normal bias source */ - snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, - WM8904_POBCTRL, 0); - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); @@ -2234,6 +2179,60 @@ static __devinit int wm8904_i2c_probe(struct i2c_client *i2c, goto err_enable; } + /* Change some default settings - latch VU and enable ZC */ + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_LEFT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ADC_DIGITAL_VOLUME_RIGHT, + WM8904_ADC_VU, WM8904_ADC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_LEFT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_DAC_DIGITAL_VOLUME_RIGHT, + WM8904_DAC_VU, WM8904_DAC_VU); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_LEFT, + WM8904_HPOUT_VU | WM8904_HPOUTLZC, + WM8904_HPOUT_VU | WM8904_HPOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT1_RIGHT, + WM8904_HPOUT_VU | WM8904_HPOUTRZC, + WM8904_HPOUT_VU | WM8904_HPOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_LEFT, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTLZC); + regmap_update_bits(wm8904->regmap, WM8904_ANALOGUE_OUT2_RIGHT, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC, + WM8904_LINEOUT_VU | WM8904_LINEOUTRZC); + regmap_update_bits(wm8904->regmap, WM8904_CLOCK_RATES_0, + WM8904_SR_MODE, 0); + + /* Apply configuration from the platform data. */ + if (wm8904->pdata) { + for (i = 0; i < WM8904_GPIO_REGS; i++) { + if (!wm8904->pdata->gpio_cfg[i]) + continue; + + regmap_update_bits(wm8904->regmap, + WM8904_GPIO_CONTROL_1 + i, + 0xffff, + wm8904->pdata->gpio_cfg[i]); + } + + /* Zero is the default value for these anyway */ + for (i = 0; i < WM8904_MIC_REGS; i++) + regmap_update_bits(wm8904->regmap, + WM8904_MIC_BIAS_CONTROL_0 + i, + 0xffff, + wm8904->pdata->mic_cfg[i]); + } + + /* Set Class W by default - this will be managed by the Class + * G widget at runtime where bypass paths are available. + */ + regmap_update_bits(wm8904->regmap, WM8904_CLASS_W_0, + WM8904_CP_DYN_PWR, WM8904_CP_DYN_PWR); + + /* Use normal bias source */ + regmap_update_bits(wm8904->regmap, WM8904_BIAS_CONTROL_0, + WM8904_POBCTRL, 0); + /* Can leave the device powered off until we need it */ regcache_cache_only(wm8904->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); -- cgit v0.10.2 From 625c4888fff33c300779ed38963e1ee7ad878a12 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:42:06 +0800 Subject: ASoC: wm8996: Move regulator notifier callbacks into I2C level Now that we're using regmap the cache is available here. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 49e0e8d..e0cf5b0 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2644,21 +2644,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) goto err; } - wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; - wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; - wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; - - /* This should really be moved into the regulator core */ - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { - ret = regulator_register_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - if (ret != 0) { - dev_err(codec->dev, - "Failed to register regulator notifier: %d\n", - ret); - } - } - /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, @@ -2858,9 +2843,7 @@ err: static int wm8996_remove(struct snd_soc_codec *codec) { - struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i; snd_soc_update_bits(codec, WM8996_INTERRUPT_CONTROL, WM8996_IM_IRQ, WM8996_IM_IRQ); @@ -2868,10 +2851,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) if (i2c->irq) free_irq(i2c->irq, codec); - for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) - regulator_unregister_notifier(wm8996->supplies[i].consumer, - &wm8996->disable_nb[i]); - return 0; } @@ -2985,6 +2964,21 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, goto err_gpio; } + wm8996->disable_nb[0].notifier_call = wm8996_regulator_event_0; + wm8996->disable_nb[1].notifier_call = wm8996_regulator_event_1; + wm8996->disable_nb[2].notifier_call = wm8996_regulator_event_2; + + /* This should really be moved into the regulator core */ + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) { + ret = regulator_register_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + if (ret != 0) { + dev_err(&i2c->dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); if (ret != 0) { @@ -3062,6 +3056,7 @@ err: static __devexit int wm8996_i2c_remove(struct i2c_client *client) { struct wm8996_priv *wm8996 = i2c_get_clientdata(client); + int i; snd_soc_unregister_codec(&client->dev); wm8996_free_gpio(wm8996); @@ -3069,6 +3064,10 @@ static __devexit int wm8996_i2c_remove(struct i2c_client *client) gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); gpio_free(wm8996->pdata.ldo_ena); } + for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) + regulator_unregister_notifier(wm8996->supplies[i].consumer, + &wm8996->disable_nb[i]); + return 0; } -- cgit v0.10.2 From d4b3d0fbb7617a65cb919ac86110b0790ae568c5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:41:16 +0800 Subject: ASoC: wm8996: Inline wm8996_reset() and optimise cache-only usage There is only one caller and this allows us to cleanly leave the CODEC with the internal LDO powered down which is the default state we're looking for and means that we can robustly disable the register cache only when we either disable the LDO or power down the external regulators. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index e0cf5b0..1579880 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1528,18 +1528,6 @@ static bool wm8996_volatile_register(struct device *dev, unsigned int reg) } } -static int wm8996_reset(struct wm8996_priv *wm8996) -{ - if (wm8996->pdata.ldo_ena > 0) { - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 1); - return 0; - } else { - return regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, - 0x8915); - } -} - static const int bclk_divs[] = { 1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96 }; @@ -1631,8 +1619,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: regcache_cache_only(codec->control_data, true); - if (wm8996->pdata.ldo_ena >= 0) + if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(codec->control_data, true); + } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); break; @@ -3019,13 +3009,18 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - ret = wm8996_reset(wm8996); - if (ret < 0) { - dev_err(&i2c->dev, "Failed to issue reset\n"); - goto err_regmap; + if (wm8996->pdata.ldo_ena > 0) { + gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); + regcache_cache_only(wm8996->regmap, true); + } else { + ret = regmap_write(wm8996->regmap, WM8996_SOFTWARE_RESET, + 0x8915); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to issue reset: %d\n", ret); + goto err_regmap; + } } - regcache_cache_only(wm8996->regmap, true); regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); wm8996_init_gpio(wm8996); -- cgit v0.10.2 From ec8ffe1868f45a72eb243f1c9797806be58276fd Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 19:10:50 +0800 Subject: ASoC: wm8996: Move register default configuration to I2C probe This gets the registers set up as early as possible, mainly useful for the GPIOs to ensure that they're in the correct mode. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1579880..00f183d 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2619,7 +2619,7 @@ static int wm8996_probe(struct snd_soc_codec *codec) int ret; struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); struct i2c_client *i2c = to_i2c_client(codec->dev); - int i, irq_flags; + int irq_flags; wm8996->codec = codec; @@ -2634,162 +2634,12 @@ static int wm8996_probe(struct snd_soc_codec *codec) goto err; } - /* Apply platform data settings */ - snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, - WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, - wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | - wm8996->pdata.inr_mode); - - for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { - if (!wm8996->pdata.gpio_default[i]) - continue; - - snd_soc_write(codec, WM8996_GPIO_1 + i, - wm8996->pdata.gpio_default[i] & 0xffff); - } - - if (wm8996->pdata.spkmute_seq) - snd_soc_update_bits(codec, WM8996_PDM_SPEAKER_MUTE_SEQUENCE, - WM8996_SPK_MUTE_ENDIAN | - WM8996_SPK_MUTE_SEQ1_MASK, - wm8996->pdata.spkmute_seq); - - snd_soc_update_bits(codec, WM8996_ACCESSORY_DETECT_MODE_2, - WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | - WM8996_MICD_SRC, wm8996->pdata.micdet_def); - - /* Latch volume update bits */ - snd_soc_update_bits(codec, WM8996_LEFT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - snd_soc_update_bits(codec, WM8996_RIGHT_LINE_INPUT_VOLUME, - WM8996_IN1_VU, WM8996_IN1_VU); - - snd_soc_update_bits(codec, WM8996_DAC1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_DAC2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_DAC2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_OUTPUT1_LEFT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT1_RIGHT_VOLUME, - WM8996_DAC1_VU, WM8996_DAC1_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_LEFT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - snd_soc_update_bits(codec, WM8996_OUTPUT2_RIGHT_VOLUME, - WM8996_DAC2_VU, WM8996_DAC2_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_TX_LEFT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_TX_RIGHT_VOLUME, - WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_LEFT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_TX_RIGHT_VOLUME, - WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); - - snd_soc_update_bits(codec, WM8996_DSP1_RX_LEFT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP1_RX_RIGHT_VOLUME, - WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_LEFT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - snd_soc_update_bits(codec, WM8996_DSP2_RX_RIGHT_VOLUME, - WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); - - /* No support currently for the underclocked TDM modes and - * pick a default TDM layout with each channel pair working with - * slots 0 and 1. */ - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1RX_CHAN0_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1RX_CHAN1_SLOTS_MASK | - WM8996_AIF1RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1RX_CHAN2_SLOTS_MASK | - WM8996_AIF1RX_CHAN2_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1RX_CHAN3_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1RX_CHAN4_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1RX_CHAN5_SLOTS_MASK | - WM8996_AIF1RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2RX_CHAN0_SLOTS_MASK | - WM8996_AIF2RX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2RX_CHAN1_SLOTS_MASK | - WM8996_AIF2RX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF1TX_CHAN0_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF1TX_CHAN1_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, - WM8996_AIF1TX_CHAN2_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, - WM8996_AIF1TX_CHAN3_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, - WM8996_AIF1TX_CHAN4_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, - WM8996_AIF1TX_CHAN5_SLOTS_MASK | - WM8996_AIF1TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); - - snd_soc_update_bits(codec, WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, - WM8996_AIF2TX_CHAN0_SLOTS_MASK | - WM8996_AIF2TX_CHAN0_START_SLOT_MASK, - 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); - snd_soc_update_bits(codec, WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, - WM8996_AIF2TX_CHAN1_SLOTS_MASK | - WM8996_AIF2TX_CHAN1_START_SLOT_MASK, - 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else snd_soc_add_codec_controls(codec, wm8996_eq_controls, ARRAY_SIZE(wm8996_eq_controls)); - /* If the TX LRCLK pins are not in LRCLK mode configure the - * AIFs to source their clocks from the RX LRCLKs. - */ - if ((snd_soc_read(codec, WM8996_GPIO_1))) - snd_soc_update_bits(codec, WM8996_AIF1_TX_LRCLK_2, - WM8996_AIF1TX_LRCLK_MODE, - WM8996_AIF1TX_LRCLK_MODE); - - if ((snd_soc_read(codec, WM8996_GPIO_2))) - snd_soc_update_bits(codec, WM8996_AIF2_TX_LRCLK_2, - WM8996_AIF2TX_LRCLK_MODE, - WM8996_AIF2TX_LRCLK_MODE); - if (i2c->irq) { if (wm8996->pdata.irq_flags) irq_flags = wm8996->pdata.irq_flags; @@ -3023,6 +2873,185 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + /* Apply platform data settings */ + regmap_update_bits(wm8996->regmap, WM8996_LINE_INPUT_CONTROL, + WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, + wm8996->pdata.inl_mode << WM8996_INL_MODE_SHIFT | + wm8996->pdata.inr_mode); + + for (i = 0; i < ARRAY_SIZE(wm8996->pdata.gpio_default); i++) { + if (!wm8996->pdata.gpio_default[i]) + continue; + + regmap_write(wm8996->regmap, WM8996_GPIO_1 + i, + wm8996->pdata.gpio_default[i] & 0xffff); + } + + if (wm8996->pdata.spkmute_seq) + regmap_update_bits(wm8996->regmap, + WM8996_PDM_SPEAKER_MUTE_SEQUENCE, + WM8996_SPK_MUTE_ENDIAN | + WM8996_SPK_MUTE_SEQ1_MASK, + wm8996->pdata.spkmute_seq); + + regmap_update_bits(wm8996->regmap, WM8996_ACCESSORY_DETECT_MODE_2, + WM8996_MICD_BIAS_SRC | WM8996_HPOUT1FB_SRC | + WM8996_MICD_SRC, wm8996->pdata.micdet_def); + + /* Latch volume update bits */ + regmap_update_bits(wm8996->regmap, WM8996_LEFT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + regmap_update_bits(wm8996->regmap, WM8996_RIGHT_LINE_INPUT_VOLUME, + WM8996_IN1_VU, WM8996_IN1_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DAC1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_DAC2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_LEFT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT1_RIGHT_VOLUME, + WM8996_DAC1_VU, WM8996_DAC1_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_LEFT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + regmap_update_bits(wm8996->regmap, WM8996_OUTPUT2_RIGHT_VOLUME, + WM8996_DAC2_VU, WM8996_DAC2_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_LEFT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_TX_RIGHT_VOLUME, + WM8996_DSP1TX_VU, WM8996_DSP1TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_LEFT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_TX_RIGHT_VOLUME, + WM8996_DSP2TX_VU, WM8996_DSP2TX_VU); + + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_LEFT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP1_RX_RIGHT_VOLUME, + WM8996_DSP1RX_VU, WM8996_DSP1RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_LEFT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + regmap_update_bits(wm8996->regmap, WM8996_DSP2_RX_RIGHT_VOLUME, + WM8996_DSP2RX_VU, WM8996_DSP2RX_VU); + + /* No support currently for the underclocked TDM modes and + * pick a default TDM layout with each channel pair working with + * slots 0 and 1. */ + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1RX_CHAN0_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1RX_CHAN1_SLOTS_MASK | + WM8996_AIF1RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1RX_CHAN2_SLOTS_MASK | + WM8996_AIF1RX_CHAN2_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1RX_CHAN3_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1RX_CHAN4_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1RX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1RX_CHAN5_SLOTS_MASK | + WM8996_AIF1RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1RX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2RX_CHAN0_SLOTS_MASK | + WM8996_AIF2RX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF2RX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2RX_CHAN1_SLOTS_MASK | + WM8996_AIF2RX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF2RX_CHAN1_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF1TX_CHAN0_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF1TX_CHAN1_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_2_CONFIGURATION, + WM8996_AIF1TX_CHAN2_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN2_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_3_CONFIGURATION, + WM8996_AIF1TX_CHAN3_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN3_SLOTS_SHIFT | 1); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_4_CONFIGURATION, + WM8996_AIF1TX_CHAN4_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN4_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_5_CONFIGURATION, + WM8996_AIF1TX_CHAN5_SLOTS_MASK | + WM8996_AIF1TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN5_SLOTS_SHIFT | 1); + + regmap_update_bits(wm8996->regmap, + WM8996_AIF2TX_CHANNEL_0_CONFIGURATION, + WM8996_AIF2TX_CHAN0_SLOTS_MASK | + WM8996_AIF2TX_CHAN0_START_SLOT_MASK, + 1 << WM8996_AIF2TX_CHAN0_SLOTS_SHIFT | 0); + regmap_update_bits(wm8996->regmap, + WM8996_AIF1TX_CHANNEL_1_CONFIGURATION, + WM8996_AIF2TX_CHAN1_SLOTS_MASK | + WM8996_AIF2TX_CHAN1_START_SLOT_MASK, + 1 << WM8996_AIF1TX_CHAN1_SLOTS_SHIFT | 1); + + /* If the TX LRCLK pins are not in LRCLK mode configure the + * AIFs to source their clocks from the RX LRCLKs. + */ + ret = regmap_read(wm8996->regmap, WM8996_GPIO_1, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO1: %d\n", ret); + goto err_regmap; + } + + if (reg & WM8996_GP1_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF1_TX_LRCLK_2, + WM8996_AIF1TX_LRCLK_MODE, + WM8996_AIF1TX_LRCLK_MODE); + + ret = regmap_read(wm8996->regmap, WM8996_GPIO_2, ®); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read GPIO2: %d\n", ret); + goto err_regmap; + } + + if (reg & WM8996_GP2_FN_MASK) + regmap_update_bits(wm8996->regmap, WM8996_AIF2_TX_LRCLK_2, + WM8996_AIF2TX_LRCLK_MODE, + WM8996_AIF2TX_LRCLK_MODE); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, -- cgit v0.10.2 From d69d65226a7910d1cfd9f3841180a0f250eeb2e9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Jun 2012 10:01:27 +0100 Subject: ASoC: dwc: Bodge around continuing absence of clock API stubs The patches for stubbing out the generic clock API still haven't been applied so we need to either add ifdefs here or add a dependency until someone decides to actually apply them. Reported-by: Stephen Rothwell Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/Kconfig b/sound/soc/dwc/Kconfig index 93e9fc3..e334900 100644 --- a/sound/soc/dwc/Kconfig +++ b/sound/soc/dwc/Kconfig @@ -1,5 +1,6 @@ config SND_DESIGNWARE_I2S tristate "Synopsys I2S Device Driver" + depends on CLKDEV_LOOKUP help Say Y or M if you want to add support for I2S driver for Synopsys desigwnware I2S device. The device supports upto -- cgit v0.10.2 From 3dba1c265268950b1ddd22e53ea823c1cb57b684 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 27 Jun 2012 14:59:20 +0100 Subject: ASoC: wm5102: Remove unused platform data header Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 9b9ea7f..e76c41e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -25,7 +25,6 @@ #include #include #include -#include #include #include -- cgit v0.10.2 From 6b4a21b64ccc218a00dc0e38676092e64df159dc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 28 Jun 2012 13:11:47 +0100 Subject: ASoC: dwc: Add missing __iomem annotations Otherwise sparse gets very upset with us. Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index e667e2b..1bd042b 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -71,12 +71,12 @@ struct dw_i2s_dev { int (*i2s_clk_cfg)(struct i2s_clk_config_data *config); }; -static inline void i2s_write_reg(void *io_base, int reg, u32 val) +static inline void i2s_write_reg(void __iomem *io_base, int reg, u32 val) { writel(val, io_base + reg); } -static inline u32 i2s_read_reg(void *io_base, int reg) +static inline u32 i2s_read_reg(void __iomem *io_base, int reg) { return readl(io_base + reg); } -- cgit v0.10.2 From cdf605255c2b592d7dbc1de19688feae941b5567 Mon Sep 17 00:00:00 2001 From: Vipin Kumar Date: Thu, 28 Jun 2012 12:31:37 +0530 Subject: ASoC: spdif_receiver: Add support for spdif in Audio Codec This patch adds the support for spdif in audio codec. Signed-off-by: vipin Kumar Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d35ba7f..62c3d4d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,7 +45,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o +snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o diff --git a/sound/soc/codecs/spdif_receiver.c b/sound/soc/codecs/spdif_receiver.c new file mode 100644 index 0000000..dd8d856 --- /dev/null +++ b/sound/soc/codecs/spdif_receiver.c @@ -0,0 +1,67 @@ +/* + * ALSA SoC SPDIF DIR (Digital Interface Reciever) driver + * + * Based on ALSA SoC SPDIF DIT driver + * + * This driver is used by controllers which can operate in DIR (SPDI/F) where + * no codec is needed. This file provides stub codec that can be used + * in these configurations. SPEAr SPDIF IN Audio controller uses this driver. + * + * Author: Vipin Kumar, + * Copyright: (C) 2012 ST Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#define STUB_RATES SNDRV_PCM_RATE_8000_192000 +#define STUB_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) + +static struct snd_soc_codec_driver soc_codec_spdif_dir; + +static struct snd_soc_dai_driver dir_stub_dai = { + .name = "dir-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 384, + .rates = STUB_RATES, + .formats = STUB_FORMATS, + }, +}; + +static int spdif_dir_probe(struct platform_device *pdev) +{ + return snd_soc_register_codec(&pdev->dev, &soc_codec_spdif_dir, + &dir_stub_dai, 1); +} + +static int spdif_dir_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver spdif_dir_driver = { + .probe = spdif_dir_probe, + .remove = spdif_dir_remove, + .driver = { + .name = "spdif-dir", + .owner = THIS_MODULE, + }, +}; + +module_platform_driver(spdif_dir_driver); + +MODULE_DESCRIPTION("ASoC SPDIF DIR driver"); +MODULE_AUTHOR("Vipin Kumar "); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From adf643aba8ed620f8c8e2533f4ace3a90e5daecf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Jun 2012 02:34:46 +0100 Subject: ASoC: spdif: Build separate RX and TX objects Otherwise we fail to link when building as modules due to multiple init/exit functions. Reported-by: Fengguang Wu Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 62c3d4d..acf8088 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -45,7 +45,8 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-sn95031-objs := sn95031.o -snd-soc-spdif-objs := spdif_transciever.o spdif_receiver.o +snd-soc-spdif-tx-objs := spdif_transciever.o +snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o snd-soc-stac9766-objs := stac9766.o @@ -159,7 +160,7 @@ obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o -obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif.o +obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o -- cgit v0.10.2 From 8a720718b37d00cf8ab311902705ae7c7890bb95 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?= Date: Mon, 18 Jun 2012 22:41:28 +0200 Subject: ASoC: dapm: Fix snd_soc_dapm_put_volsw() connect MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit snd_soc_dapm_put_volsw() sets connect incorrectly in the case max > 1 with invert. In that case, the raw disconnect value should be max, which corresponds to the userspace value 0. This use case currently does not appear upstream, but it could break SOC_DAPM_SINGLE() or SOC_DAPM_SINGLE_TLV() elsewhere or in the future. Signed-off-by: Benoît Thébaudeau Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c2206bc..9670668 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2515,19 +2515,13 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, int wi; val = (ucontrol->value.integer.value[0] & mask); + connect = !!val; if (invert) val = max - val; mask = mask << shift; val = val << shift; - if (val) - /* new connection */ - connect = invert ? 0 : 1; - else - /* old connection must be powered down */ - connect = invert ? 1 : 0; - mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); change = snd_soc_test_bits(widget->codec, reg, mask, val); -- cgit v0.10.2 From 9c9acc91561221c30a530c9b84056609d0307c7c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 3 Jul 2012 14:04:04 +0530 Subject: ASoC: smdk_wm8994: Convert to use snd_soc_register_card() Current method for machine driver to register with the ASoC core is to use snd_soc_register_card() instead of creating a "soc-audio" platform device. Signed-off-by: Sachin Kamat Acked-by: Sangbeom Kim Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/smdk_wm8994.c b/sound/soc/samsung/smdk_wm8994.c index 8eb309f..48dd4dd 100644 --- a/sound/soc/samsung/smdk_wm8994.c +++ b/sound/soc/samsung/smdk_wm8994.c @@ -149,31 +149,41 @@ static struct snd_soc_card smdk = { .num_links = ARRAY_SIZE(smdk_dai), }; -static struct platform_device *smdk_snd_device; -static int __init smdk_audio_init(void) +static int __devinit smdk_audio_probe(struct platform_device *pdev) { int ret; + struct snd_soc_card *card = &smdk; - smdk_snd_device = platform_device_alloc("soc-audio", -1); - if (!smdk_snd_device) - return -ENOMEM; + card->dev = &pdev->dev; + ret = snd_soc_register_card(card); - platform_set_drvdata(smdk_snd_device, &smdk); - - ret = platform_device_add(smdk_snd_device); if (ret) - platform_device_put(smdk_snd_device); + dev_err(&pdev->dev, "snd_soc_register_card() failed:%d\n", ret); return ret; } -module_init(smdk_audio_init); -static void __exit smdk_audio_exit(void) +static int __devexit smdk_audio_remove(struct platform_device *pdev) { - platform_device_unregister(smdk_snd_device); + struct snd_soc_card *card = platform_get_drvdata(pdev); + + snd_soc_unregister_card(card); + + return 0; } -module_exit(smdk_audio_exit); + +static struct platform_driver smdk_audio_driver = { + .driver = { + .name = "smdk-audio", + .owner = THIS_MODULE, + }, + .probe = smdk_audio_probe, + .remove = __devexit_p(smdk_audio_remove), +}; + +module_platform_driver(smdk_audio_driver); MODULE_DESCRIPTION("ALSA SoC SMDK WM8994"); MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:smdk-audio"); -- cgit v0.10.2 From da602ab8a10e47c59be1a7ce524aaa76b77c23b6 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Beno=C3=AEt=20Th=C3=A9baudeau?= Date: Tue, 3 Jul 2012 20:18:17 +0200 Subject: ASoC: dapm: Remove incomplete stereo code MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Stereo is not yet supported by dapm widgets, so remove stereo code from snd_soc_dapm_get_volsw(), and warn if stereo controls are detected. Signed-off-by: Benoît Thébaudeau Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 9670668..912330b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2464,23 +2464,20 @@ int snd_soc_dapm_get_volsw(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; unsigned int reg = mc->reg; unsigned int shift = mc->shift; - unsigned int rshift = mc->rshift; int max = mc->max; - unsigned int invert = mc->invert; unsigned int mask = (1 << fls(max)) - 1; + unsigned int invert = mc->invert; + + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); ucontrol->value.integer.value[0] = (snd_soc_read(widget->codec, reg) >> shift) & mask; - if (shift != rshift) - ucontrol->value.integer.value[1] = - (snd_soc_read(widget->codec, reg) >> rshift) & mask; - if (invert) { + if (invert) ucontrol->value.integer.value[0] = max - ucontrol->value.integer.value[0]; - if (shift != rshift) - ucontrol->value.integer.value[1] = - max - ucontrol->value.integer.value[1]; - } return 0; } @@ -2514,6 +2511,11 @@ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; int wi; + if (snd_soc_volsw_is_stereo(mc)) + dev_warn(widget->dapm->dev, + "Control '%s' is stereo, which is not supported\n", + kcontrol->id.name); + val = (ucontrol->value.integer.value[0] & mask); connect = !!val; -- cgit v0.10.2 From df79f55df3992fdd5dd206de6aa9af6a8ec1f86f Mon Sep 17 00:00:00 2001 From: Laxman Dewangan Date: Fri, 29 Jun 2012 17:04:33 +0530 Subject: ASoC: tegra: use dmaengine based dma driver Use the dmaengine based Tegra APB DMA driver for data transfer between SPI fifo and memory in place of legacy Tegra APB DMA. Because generic soc-dmaengine-pcm uses the DMAs API based on dmaengine, using the exported APIs provided by this generic driver. The new driver is selected if legacy driver is not selected and new dma driver is enabled through config file. Signed-off-by: Laxman Dewangan Acked-by: Stephen Warren Tested-by: Stephen Warren Signed-off-by: Mark Brown diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index c1c8e95..7b6a1eb 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -1,7 +1,8 @@ config SND_SOC_TEGRA tristate "SoC Audio for the Tegra System-on-Chip" - depends on ARCH_TEGRA && TEGRA_SYSTEM_DMA + depends on ARCH_TEGRA && (TEGRA_SYSTEM_DMA || TEGRA20_APB_DMA) select REGMAP_MMIO + select SND_SOC_DMAENGINE_PCM if TEGRA20_APB_DMA help Say Y or M here if you want support for SoC audio on Tegra. diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 127348d..5658bce 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -36,6 +36,7 @@ #include #include #include +#include #include "tegra_pcm.h" @@ -56,6 +57,7 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { .fifo_size = 4, }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) static void tegra_pcm_queue_dma(struct tegra_runtime_data *prtd) { struct snd_pcm_substream *substream = prtd->substream; @@ -285,6 +287,119 @@ static struct snd_pcm_ops tegra_pcm_ops = { .pointer = tegra_pcm_pointer, .mmap = tegra_pcm_mmap, }; +#else +static int tegra_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + int ret; + + /* Set HW params now that initialization is complete */ + snd_soc_set_runtime_hwparams(substream, &tegra_pcm_hardware); + + ret = snd_dmaengine_pcm_open(substream, NULL, NULL); + if (ret) { + dev_err(dev, "dmaengine pcm open failed with err %d\n", ret); + return ret; + } + + return 0; +} + +static int tegra_pcm_close(struct snd_pcm_substream *substream) +{ + snd_dmaengine_pcm_close(substream); + return 0; +} + +static int tegra_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct device *dev = rtd->platform->dev; + struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + struct tegra_pcm_dma_params *dmap; + struct dma_slave_config slave_config; + int ret; + + dmap = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + ret = snd_hwparams_to_dma_slave_config(substream, params, + &slave_config); + if (ret) { + dev_err(dev, "hw params config failed with err %d\n", ret); + return ret; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.dst_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.dst_addr = dmap->addr; + slave_config.src_maxburst = 0; + } else { + slave_config.src_addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + slave_config.src_addr = dmap->addr; + slave_config.dst_maxburst = 0; + } + slave_config.slave_id = dmap->req_sel; + + ret = dmaengine_slave_config(chan, &slave_config); + if (ret < 0) { + dev_err(dev, "dma slave config failed with err %d\n", ret); + return ret; + } + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + return 0; +} + +static int tegra_pcm_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + return 0; +} + +static int tegra_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_START); + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + return snd_dmaengine_pcm_trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + default: + return -EINVAL; + } + return 0; +} + +static int tegra_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + return dma_mmap_writecombine(substream->pcm->card->dev, vma, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); +} + +static struct snd_pcm_ops tegra_pcm_ops = { + .open = tegra_pcm_open, + .close = tegra_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = tegra_pcm_hw_params, + .hw_free = tegra_pcm_hw_free, + .trigger = tegra_pcm_trigger, + .pointer = snd_dmaengine_pcm_pointer, + .mmap = tegra_pcm_mmap, +}; +#endif static int tegra_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) { diff --git a/sound/soc/tegra/tegra_pcm.h b/sound/soc/tegra/tegra_pcm.h index 985d418..a3a4503 100644 --- a/sound/soc/tegra/tegra_pcm.h +++ b/sound/soc/tegra/tegra_pcm.h @@ -40,6 +40,7 @@ struct tegra_pcm_dma_params { unsigned long req_sel; }; +#if defined(CONFIG_TEGRA_SYSTEM_DMA) struct tegra_runtime_data { struct snd_pcm_substream *substream; spinlock_t lock; @@ -51,6 +52,7 @@ struct tegra_runtime_data { struct tegra_dma_req dma_req[2]; struct tegra_dma_channel *dma_chan; }; +#endif int tegra_pcm_platform_register(struct device *dev); void tegra_pcm_platform_unregister(struct device *dev); -- cgit v0.10.2 From 081413f206876e9d3755e1673828c7742fd00184 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 2 Jul 2012 18:19:58 +0100 Subject: ASoC: wm8962: Log AIF configuration requested by hw_params() Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 27da4d7..beb709b 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2580,6 +2580,9 @@ static int wm8962_hw_params(struct snd_pcm_substream *substream, WM8962_SAMPLE_RATE_INT_MODE | WM8962_SAMPLE_RATE_MASK, adctl3); + dev_dbg(codec->dev, "hw_params set BCLK %dHz LRCLK %dHz\n", + wm8962->bclk, wm8962->lrclk); + if (codec->dapm.bias_level == SND_SOC_BIAS_ON) wm8962_configure_bclk(codec); -- cgit v0.10.2 From 38cbf9598feba97de9f9b43efa9153fd7c1a2ec9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Jun 2012 13:04:02 +0100 Subject: ASoC: core: Try to use regmap if the driver doesn't set up any I/O Since most new drivers are expected to use regmap and since frequently the only thing we need to do in the CODEC probe function is configure the I/O try to initialise the register I/O using regmap if the driver hasn't done so after probe(). Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe16135..64b464c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1095,6 +1095,10 @@ static int soc_probe_codec(struct snd_soc_card *card, } } + /* If the driver didn't set I/O up try regmap */ + if (!codec->control_data) + snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (driver->controls) snd_soc_add_codec_controls(codec, driver->controls, driver->num_controls); -- cgit v0.10.2 From 2974d6b1aa5261d8db1b614437cc6bafd3ddf0f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Jun 2012 11:06:53 +0100 Subject: ASoC: wm8994: Don't suspend accessory detection Leave it up to the machine driver to disable accessory detection if desired, the common pattern is to have accessory detection be a wake source. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index a6e82d0..7bb8752 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2970,23 +2970,8 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_codec_suspend(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->wm8994; int i, ret; - switch (control->type) { - case WM8994: - snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); - break; - case WM1811: - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, 0); - /* Fall through */ - case WM8958: - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, 0); - break; - } - for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -3036,28 +3021,6 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) i + 1, ret); } - switch (control->type) { - case WM8994: - if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) - snd_soc_update_bits(codec, WM8994_MICBIAS, - WM8994_MICD_ENA, WM8994_MICD_ENA); - break; - case WM1811: - if (wm8994->jackdet && wm8994->jack_cb) { - /* Restart from idle */ - snd_soc_update_bits(codec, WM8994_ANTIPOP_2, - WM1811_JACKDET_MODE_MASK, - WM1811_JACKDET_MODE_JACK); - break; - } - break; - case WM8958: - if (wm8994->jack_cb) - snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, - WM8958_MICD_ENA, WM8958_MICD_ENA); - break; - } - return 0; } #else -- cgit v0.10.2 From 784a897e2310410ed169b5b331f2b7f06b7d58b7 Mon Sep 17 00:00:00 2001 From: Jiri Prchal Date: Wed, 4 Jul 2012 08:12:50 +0200 Subject: ASoC: tlv320aic3x: add missing registers and bits Adds register and bit shift definitions in header file. Changes are for TLV320AIC310x based on data sheet. Signed-off-by: Jiri Prchal Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb..5da5eb3 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0 -- cgit v0.10.2 From c9e8e8d2541cbf0331400e2fa2fdca404e3569d4 Mon Sep 17 00:00:00 2001 From: Jiri Prchal Date: Wed, 4 Jul 2012 08:12:51 +0200 Subject: ASoC: tlv320aic3x: extending registers cache Adds missing register default values to cache. Signed-off-by: Jiri Prchal Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 58ef59d..174de66 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -118,7 +118,9 @@ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { 0x00, 0x00, 0x00, 0x00, /* 88 */ 0x00, 0x00, 0x00, 0x00, /* 92 */ 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x02, 0x00, /* 100 */ + 0x00, 0x00, 0x00, 0x00, /* 104 */ + 0x00, 0x00, /* 108 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ -- cgit v0.10.2 From 3be58dbb92871442191188ae51b449e1a9f0fe64 Mon Sep 17 00:00:00 2001 From: Rajeev Kumar Date: Wed, 4 Jul 2012 16:11:12 +0530 Subject: ASoC: STA529: Add support for STA529 Audio Codec The STA529 is a digital stereo class-D audio amplifier. It includes an audio DSP, an ST proprietary high-efficiency class-D driver and CMOS power output stage. It is intended for high-efficiency digital-to-power-audio conversion for portable applications. Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 1de24cc..bbcb038 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -57,6 +57,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_SPDIF select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI select SND_SOC_STA32X if I2C + select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER @@ -284,6 +285,9 @@ config SND_SOC_SSM2602 config SND_SOC_STA32X tristate +config SND_SOC_STA529 + tristate + config SND_SOC_STAC9766 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index acf8088..8da3d22 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -49,6 +49,7 @@ snd-soc-spdif-tx-objs := spdif_transciever.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o +snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o @@ -163,6 +164,7 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o +obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c new file mode 100644 index 0000000..a9f34c7 --- /dev/null +++ b/sound/soc/codecs/sta529.c @@ -0,0 +1,441 @@ +/* + * ASoC codec driver for spear platform + * + * sound/soc/codecs/sta529.c -- spear ALSA Soc codec driver + * + * Copyright (C) 2012 ST Microelectronics + * Rajeev Kumar + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include +#include + +/* STA529 Register offsets */ +#define STA529_FFXCFG0 0x00 +#define STA529_FFXCFG1 0x01 +#define STA529_MVOL 0x02 +#define STA529_LVOL 0x03 +#define STA529_RVOL 0x04 +#define STA529_TTF0 0x05 +#define STA529_TTF1 0x06 +#define STA529_TTP0 0x07 +#define STA529_TTP1 0x08 +#define STA529_S2PCFG0 0x0A +#define STA529_S2PCFG1 0x0B +#define STA529_P2SCFG0 0x0C +#define STA529_P2SCFG1 0x0D +#define STA529_PLLCFG0 0x14 +#define STA529_PLLCFG1 0x15 +#define STA529_PLLCFG2 0x16 +#define STA529_PLLCFG3 0x17 +#define STA529_PLLPFE 0x18 +#define STA529_PLLST 0x19 +#define STA529_ADCCFG 0x1E /*mic_select*/ +#define STA529_CKOCFG 0x1F +#define STA529_MISC 0x20 +#define STA529_PADST0 0x21 +#define STA529_PADST1 0x22 +#define STA529_FFXST 0x23 +#define STA529_PWMIN1 0x2D +#define STA529_PWMIN2 0x2E +#define STA529_POWST 0x32 + +#define STA529_MAX_REGISTER 0x32 + +#define STA529_RATES (SNDRV_PCM_RATE_8000 | \ + SNDRV_PCM_RATE_11025 | \ + SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | \ + SNDRV_PCM_RATE_32000 | \ + SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000) + +#define STA529_FORMAT (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) +#define S2PC_VALUE 0x98 +#define CLOCK_OUT 0x60 +#define LEFT_J_DATA_FORMAT 0x10 +#define I2S_DATA_FORMAT 0x12 +#define RIGHT_J_DATA_FORMAT 0x14 +#define CODEC_MUTE_VAL 0x80 + +#define POWER_CNTLMSAK 0x40 +#define POWER_STDBY 0x40 +#define FFX_MASK 0x80 +#define FFX_OFF 0x80 +#define POWER_UP 0x00 +#define FFX_CLK_ENB 0x01 +#define FFX_CLK_DIS 0x00 +#define FFX_CLK_MSK 0x01 +#define PLAY_FREQ_RANGE_MSK 0x70 +#define CAP_FREQ_RANGE_MSK 0x0C +#define PDATA_LEN_MSK 0xC0 +#define BCLK_TO_FS_MSK 0x30 +#define AUDIO_MUTE_MSK 0x80 + +static const struct reg_default sta529_reg_defaults[] = { + { 0, 0x35 }, /* R0 - FFX Configuration reg 0 */ + { 1, 0xc8 }, /* R1 - FFX Configuration reg 1 */ + { 2, 0x50 }, /* R2 - Master Volume */ + { 3, 0x00 }, /* R3 - Left Volume */ + { 4, 0x00 }, /* R4 - Right Volume */ + { 10, 0xb2 }, /* R10 - S2P Config Reg 0 */ + { 11, 0x41 }, /* R11 - S2P Config Reg 1 */ + { 12, 0x92 }, /* R12 - P2S Config Reg 0 */ + { 13, 0x41 }, /* R13 - P2S Config Reg 1 */ + { 30, 0xd2 }, /* R30 - ADC Config Reg */ + { 31, 0x40 }, /* R31 - clock Out Reg */ + { 32, 0x21 }, /* R32 - Misc Register */ +}; + +struct sta529 { + struct regmap *regmap; +}; + +static bool sta529_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + + case STA529_FFXCFG0: + case STA529_FFXCFG1: + case STA529_MVOL: + case STA529_LVOL: + case STA529_RVOL: + case STA529_S2PCFG0: + case STA529_S2PCFG1: + case STA529_P2SCFG0: + case STA529_P2SCFG1: + case STA529_ADCCFG: + case STA529_CKOCFG: + case STA529_MISC: + return true; + default: + return false; + } +} + + +static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", + "Phase-shift"}; + +static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); +static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); +static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); + +static const struct snd_kcontrol_new sta529_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, + 127, 0, out_gain_tlv), + SOC_SINGLE_TLV("Master Playback Volume", STA529_MVOL, 0, 127, 1, + master_vol_tlv), + SOC_ENUM("PWM Select", pwm_src), +}; + +static int sta529_set_bias_level(struct snd_soc_codec *codec, enum + snd_soc_bias_level level) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + snd_soc_update_bits(codec, STA529_FFXCFG0, POWER_CNTLMSAK, + POWER_UP); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_ENB); + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + regcache_sync(sta529->regmap); + snd_soc_update_bits(codec, STA529_FFXCFG0, + POWER_CNTLMSAK, POWER_STDBY); + /* Making FFX output to zero */ + snd_soc_update_bits(codec, STA529_FFXCFG0, FFX_MASK, + FFX_OFF); + snd_soc_update_bits(codec, STA529_MISC, FFX_CLK_MSK, + FFX_CLK_DIS); + break; + case SND_SOC_BIAS_OFF: + break; + } + + /* + * store the label for powers down audio subsystem for suspend.This is + * used by soc core layer + */ + codec->dapm.bias_level = level; + + return 0; + +} + +static int sta529_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + int pdata, play_freq_val, record_freq_val; + int bclk_to_fs_ratio; + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + pdata = 1; + bclk_to_fs_ratio = 0; + break; + case SNDRV_PCM_FORMAT_S24_LE: + pdata = 2; + bclk_to_fs_ratio = 1; + break; + case SNDRV_PCM_FORMAT_S32_LE: + pdata = 3; + bclk_to_fs_ratio = 2; + break; + default: + dev_err(codec->dev, "Unsupported format\n"); + return -EINVAL; + } + + switch (params_rate(params)) { + case 8000: + case 11025: + play_freq_val = 0; + record_freq_val = 2; + break; + case 16000: + case 22050: + play_freq_val = 1; + record_freq_val = 0; + break; + + case 32000: + case 44100: + case 48000: + play_freq_val = 2; + record_freq_val = 0; + break; + default: + dev_err(codec->dev, "Unsupported rate\n"); + return -EINVAL; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + snd_soc_update_bits(codec, STA529_S2PCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_S2PCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, PLAY_FREQ_RANGE_MSK, + play_freq_val << 4); + } else { + snd_soc_update_bits(codec, STA529_P2SCFG1, PDATA_LEN_MSK, + pdata << 6); + snd_soc_update_bits(codec, STA529_P2SCFG1, BCLK_TO_FS_MSK, + bclk_to_fs_ratio << 4); + snd_soc_update_bits(codec, STA529_MISC, CAP_FREQ_RANGE_MSK, + record_freq_val << 2); + } + + return 0; +} + +static int sta529_mute(struct snd_soc_dai *dai, int mute) +{ + u8 val = 0; + + if (mute) + val |= CODEC_MUTE_VAL; + + snd_soc_update_bits(dai->codec, STA529_FFXCFG0, AUDIO_MUTE_MSK, val); + + return 0; +} + +static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 mode = 0; + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + mode = LEFT_J_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_I2S: + mode = I2S_DATA_FORMAT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + mode = RIGHT_J_DATA_FORMAT; + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + + return 0; +} + +static const struct snd_soc_dai_ops sta529_dai_ops = { + .hw_params = sta529_hw_params, + .set_fmt = sta529_set_dai_fmt, + .digital_mute = sta529_mute, +}; + +static struct snd_soc_dai_driver sta529_dai = { + .name = "sta529-audio", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = STA529_RATES, + .formats = STA529_FORMAT, + }, + .ops = &sta529_dai_ops, +}; + +static int sta529_probe(struct snd_soc_codec *codec) +{ + struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); + int ret; + + codec->control_data = sta529->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +/* power down chip */ +static int sta529_remove(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_suspend(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_OFF); + + return 0; +} + +static int sta529_resume(struct snd_soc_codec *codec) +{ + sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return 0; +} + +struct snd_soc_codec_driver sta529_codec_driver = { + .probe = sta529_probe, + .remove = sta529_remove, + .set_bias_level = sta529_set_bias_level, + .suspend = sta529_suspend, + .resume = sta529_resume, + .controls = sta529_snd_controls, + .num_controls = ARRAY_SIZE(sta529_snd_controls), +}; + +static const struct regmap_config sta529_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = STA529_MAX_REGISTER, + .readable_reg = sta529_readable, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = sta529_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(sta529_reg_defaults), +}; + +static __devinit int sta529_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct sta529 *sta529; + int ret; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + sta529 = devm_kzalloc(&i2c->dev, sizeof(struct sta529), GFP_KERNEL); + if (sta529 == NULL) { + dev_err(&i2c->dev, "Can not allocate memory\n"); + return -ENOMEM; + } + + sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); + if (IS_ERR(sta529->regmap)) { + dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); + return PTR_ERR(sta529->regmap); + } + + i2c_set_clientdata(i2c, sta529); + + ret = snd_soc_register_codec(&i2c->dev, + &sta529_codec_driver, &sta529_dai, 1); + if (ret != 0) + dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); + + return ret; +} + +static int __devexit sta529_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +static const struct i2c_device_id sta529_i2c_id[] = { + { "sta529", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, sta529_i2c_id); + +static struct i2c_driver sta529_i2c_driver = { + .driver = { + .name = "sta529", + .owner = THIS_MODULE, + }, + .probe = sta529_i2c_probe, + .remove = __devexit_p(sta529_i2c_remove), + .id_table = sta529_i2c_id, +}; + +module_i2c_driver(sta529_i2c_driver); + +MODULE_DESCRIPTION("ASoC STA529 codec driver"); +MODULE_AUTHOR("Rajeev Kumar "); +MODULE_LICENSE("GPL"); -- cgit v0.10.2 From e584f9b4c2a919eb665fea8536ecd2bd7260e876 Mon Sep 17 00:00:00 2001 From: Vipin Kumar Date: Wed, 4 Jul 2012 16:11:13 +0530 Subject: ASoC: SPEAr spdif_out: Add spdif out support This patch implements the spdif out driver for ST peripheral. This peripheral implements IEC60958 standard Signed-off-by: Vipin Kumar Signed-off-by: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/sound/soc/spear/spdif_out.c b/sound/soc/spear/spdif_out.c new file mode 100644 index 0000000..5eac4cd --- /dev/null +++ b/sound/soc/spear/spdif_out.c @@ -0,0 +1,389 @@ +/* + * ALSA SoC SPDIF Out Audio Layer for spear processors + * + * Copyright (C) 2012 ST Microelectronics + * Vipin Kumar + * + * This file is licensed under the terms of the GNU General Public + * License version 2. This program is licensed "as is" without any + * warranty of any kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include "spdif_out_regs.h" + +struct spdif_out_params { + u32 rate; + u32 core_freq; + u32 mute; +}; + +struct spdif_out_dev { + struct clk *clk; + struct spear_dma_data dma_params; + struct spdif_out_params saved_params; + u32 running; + void __iomem *io_base; +}; + +static void spdif_out_configure(struct spdif_out_dev *host) +{ + writel(SPDIF_OUT_RESET, host->io_base + SPDIF_OUT_SOFT_RST); + mdelay(1); + writel(readl(host->io_base + SPDIF_OUT_SOFT_RST) & ~SPDIF_OUT_RESET, + host->io_base + SPDIF_OUT_SOFT_RST); + + writel(SPDIF_OUT_FDMA_TRIG_16 | SPDIF_OUT_MEMFMT_16_16 | + SPDIF_OUT_VALID_HW | SPDIF_OUT_USER_HW | + SPDIF_OUT_CHNLSTA_HW | SPDIF_OUT_PARITY_HW, + host->io_base + SPDIF_OUT_CFG); + + writel(0x7F, host->io_base + SPDIF_OUT_INT_STA_CLR); + writel(0x7F, host->io_base + SPDIF_OUT_INT_EN_CLR); +} + +static int spdif_out_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + int ret; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + snd_soc_dai_set_dma_data(cpu_dai, substream, (void *)&host->dma_params); + + ret = clk_enable(host->clk); + if (ret) + return ret; + + host->running = true; + spdif_out_configure(host); + + return 0; +} + +static void spdif_out_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return; + + clk_disable(host->clk); + host->running = false; + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static void spdif_out_clock(struct spdif_out_dev *host, u32 core_freq, + u32 rate) +{ + u32 divider, ctrl; + + clk_set_rate(host->clk, core_freq); + divider = DIV_ROUND_CLOSEST(clk_get_rate(host->clk), (rate * 128)); + + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_DIVIDER_MASK; + ctrl |= (divider << SPDIF_DIVIDER_SHIFT) & SPDIF_DIVIDER_MASK; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); +} + +static int spdif_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 rate, core_freq; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + rate = params_rate(params); + + switch (rate) { + case 8000: + case 16000: + case 32000: + case 64000: + /* + * The clock is multiplied by 10 to bring it to feasible range + * of frequencies for sscg + */ + core_freq = 64000 * 128 * 10; /* 81.92 MHz */ + break; + case 5512: + case 11025: + case 22050: + case 44100: + case 88200: + case 176400: + core_freq = 176400 * 128; /* 22.5792 MHz */ + break; + case 48000: + case 96000: + case 192000: + default: + core_freq = 192000 * 128; /* 24.576 MHz */ + break; + } + + spdif_out_clock(host, core_freq, rate); + host->saved_params.core_freq = core_freq; + host->saved_params.rate = rate; + + return 0; +} + +static int spdif_out_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 ctrl; + int ret = 0; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + if (!host->saved_params.mute) + ctrl |= SPDIF_OPMODE_AUD_DATA | + SPDIF_STATE_NORMAL; + else + ctrl |= SPDIF_OPMODE_MUTE_PCM; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ctrl = readl(host->io_base + SPDIF_OUT_CTRL); + ctrl &= ~SPDIF_OPMODE_MASK; + ctrl |= SPDIF_OPMODE_OFF; + writel(ctrl, host->io_base + SPDIF_OUT_CTRL); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int spdif_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(dai); + u32 val; + + host->saved_params.mute = mute; + val = readl(host->io_base + SPDIF_OUT_CTRL); + val &= ~SPDIF_OPMODE_MASK; + + if (mute) + val |= SPDIF_OPMODE_MUTE_PCM; + else { + if (host->running) + val |= SPDIF_OPMODE_AUD_DATA | SPDIF_STATE_NORMAL; + else + val |= SPDIF_OPMODE_OFF; + } + + writel(val, host->io_base + SPDIF_OUT_CTRL); + return 0; +} + +static int spdif_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + ucontrol->value.integer.value[0] = host->saved_params.mute; + return 0; +} + +static int spdif_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct snd_soc_card *card = codec->card; + struct snd_soc_pcm_runtime *rtd = card->rtd; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct spdif_out_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + + if (host->saved_params.mute == ucontrol->value.integer.value[0]) + return 0; + + spdif_digital_mute(cpu_dai, ucontrol->value.integer.value[0]); + + return 1; +} +static const struct snd_kcontrol_new spdif_out_controls[] = { + SOC_SINGLE_BOOL_EXT("IEC958 Playback Switch", 0, + spdif_mute_get, spdif_mute_put), +}; + +int spdif_soc_dai_probe(struct snd_soc_dai *dai) +{ + return snd_soc_add_dai_controls(dai, spdif_out_controls, + ARRAY_SIZE(spdif_out_controls)); +} + +static const struct snd_soc_dai_ops spdif_out_dai_ops = { + .digital_mute = spdif_digital_mute, + .startup = spdif_out_startup, + .shutdown = spdif_out_shutdown, + .trigger = spdif_out_trigger, + .hw_params = spdif_out_hw_params, +}; + +static struct snd_soc_dai_driver spdif_out_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_192000), + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .probe = spdif_soc_dai_probe, + .ops = &spdif_out_dai_ops, +}; + +static int spdif_out_probe(struct platform_device *pdev) +{ + struct spdif_out_dev *host; + struct spear_spdif_platform_data *pdata; + struct resource *res; + int ret; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) + return -EINVAL; + + if (!devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name)) { + dev_warn(&pdev->dev, "Failed to get memory resourse\n"); + return -ENOENT; + } + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) { + dev_warn(&pdev->dev, "kzalloc fail\n"); + return -ENOMEM; + } + + host->io_base = devm_ioremap(&pdev->dev, res->start, + resource_size(res)); + if (!host->io_base) { + dev_warn(&pdev->dev, "ioremap failed\n"); + return -ENOMEM; + } + + host->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(host->clk)) + return PTR_ERR(host->clk); + + pdata = dev_get_platdata(&pdev->dev); + + host->dma_params.data = pdata->dma_params; + host->dma_params.addr = res->start + SPDIF_OUT_FIFO_DATA; + host->dma_params.max_burst = 16; + host->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + host->dma_params.filter = pdata->filter; + + dev_set_drvdata(&pdev->dev, host); + + ret = snd_soc_register_dai(&pdev->dev, &spdif_out_dai); + if (ret != 0) { + clk_put(host->clk); + return ret; + } + + return 0; +} + +static int spdif_out_remove(struct platform_device *pdev) +{ + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + snd_soc_unregister_dai(&pdev->dev); + dev_set_drvdata(&pdev->dev, NULL); + + clk_put(host->clk); + + return 0; +} + +#ifdef CONFIG_PM +static int spdif_out_suspend(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) + clk_disable(host->clk); + + return 0; +} + +static int spdif_out_resume(struct device *dev) +{ + struct platform_device *pdev = to_platform_device(dev); + struct spdif_out_dev *host = dev_get_drvdata(&pdev->dev); + + if (host->running) { + clk_enable(host->clk); + spdif_out_configure(host); + spdif_out_clock(host, host->saved_params.core_freq, + host->saved_params.rate); + } + return 0; +} + +static SIMPLE_DEV_PM_OPS(spdif_out_dev_pm_ops, spdif_out_suspend, \ + spdif_out_resume); + +#define SPDIF_OUT_DEV_PM_OPS (&spdif_out_dev_pm_ops) + +#else +#define SPDIF_OUT_DEV_PM_OPS NULL + +#endif + +static struct platform_driver spdif_out_driver = { + .probe = spdif_out_probe, + .remove = spdif_out_remove, + .driver = { + .name = "spdif-out", + .owner = THIS_MODULE, + .pm = SPDIF_OUT_DEV_PM_OPS, + }, +}; + +module_platform_driver(spdif_out_driver); + +MODULE_AUTHOR("Vipin Kumar "); +MODULE_DESCRIPTION("SPEAr SPDIF OUT SoC Interface"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:spdif_out"); diff --git a/sound/soc/spear/spdif_out_regs.h b/sound/soc/spear/spdif_out_regs.h new file mode 100644 index 0000000..a5e5332 --- /dev/null +++ b/sound/soc/spear/spdif_out_regs.h @@ -0,0 +1,79 @@ +/* + * SPEAr SPDIF OUT controller header file + * + * Copyright (ST) 2011 Vipin Kumar (vipin.kumar@st.com) + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#ifndef SPDIF_OUT_REGS_H +#define SPDIF_OUT_REGS_H + +#define SPDIF_OUT_SOFT_RST 0x00 + #define SPDIF_OUT_RESET (1 << 0) +#define SPDIF_OUT_FIFO_DATA 0x04 +#define SPDIF_OUT_INT_STA 0x08 +#define SPDIF_OUT_INT_STA_CLR 0x0C + #define SPDIF_INT_UNDERFLOW (1 << 0) + #define SPDIF_INT_EODATA (1 << 1) + #define SPDIF_INT_EOBLOCK (1 << 2) + #define SPDIF_INT_EOLATENCY (1 << 3) + #define SPDIF_INT_EOPD_DATA (1 << 4) + #define SPDIF_INT_MEMFULLREAD (1 << 5) + #define SPDIF_INT_EOPD_PAUSE (1 << 6) + +#define SPDIF_OUT_INT_EN 0x10 +#define SPDIF_OUT_INT_EN_SET 0x14 +#define SPDIF_OUT_INT_EN_CLR 0x18 +#define SPDIF_OUT_CTRL 0x1C + #define SPDIF_OPMODE_MASK (7 << 0) + #define SPDIF_OPMODE_OFF (0 << 0) + #define SPDIF_OPMODE_MUTE_PCM (1 << 0) + #define SPDIF_OPMODE_MUTE_PAUSE (2 << 0) + #define SPDIF_OPMODE_AUD_DATA (3 << 0) + #define SPDIF_OPMODE_ENCODE (4 << 0) + #define SPDIF_STATE_NORMAL (1 << 3) + #define SPDIF_DIVIDER_MASK (0xff << 5) + #define SPDIF_DIVIDER_SHIFT (5) + #define SPDIF_SAMPLEREAD_MASK (0x1ffff << 15) + #define SPDIF_SAMPLEREAD_SHIFT (15) +#define SPDIF_OUT_STA 0x20 +#define SPDIF_OUT_PA_PB 0x24 +#define SPDIF_OUT_PC_PD 0x28 +#define SPDIF_OUT_CL1 0x2C +#define SPDIF_OUT_CR1 0x30 +#define SPDIF_OUT_CL2_CR2_UV 0x34 +#define SPDIF_OUT_PAUSE_LAT 0x38 +#define SPDIF_OUT_FRMLEN_BRST 0x3C +#define SPDIF_OUT_CFG 0x40 + #define SPDIF_OUT_MEMFMT_16_0 (0 << 5) + #define SPDIF_OUT_MEMFMT_16_16 (1 << 5) + #define SPDIF_OUT_VALID_DMA (0 << 3) + #define SPDIF_OUT_VALID_HW (1 << 3) + #define SPDIF_OUT_USER_DMA (0 << 2) + #define SPDIF_OUT_USER_HW (1 << 2) + #define SPDIF_OUT_CHNLSTA_DMA (0 << 1) + #define SPDIF_OUT_CHNLSTA_HW (1 << 1) + #define SPDIF_OUT_PARITY_HW (0 << 0) + #define SPDIF_OUT_PARITY_DMA (1 << 0) + #define SPDIF_OUT_FDMA_TRIG_2 (2 << 8) + #define SPDIF_OUT_FDMA_TRIG_6 (6 << 8) + #define SPDIF_OUT_FDMA_TRIG_8 (8 << 8) + #define SPDIF_OUT_FDMA_TRIG_10 (10 << 8) + #define SPDIF_OUT_FDMA_TRIG_12 (12 << 8) + #define SPDIF_OUT_FDMA_TRIG_16 (16 << 8) + #define SPDIF_OUT_FDMA_TRIG_18 (18 << 8) + +#endif /* SPDIF_OUT_REGS_H */ -- cgit v0.10.2 From 1520ffd218f4aa53bc7652c0f6454da3cb428337 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 19:04:11 +0100 Subject: ASoC: dwc: Staticise non-exported i2s_start() Signed-off-by: Mark Brown diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index 1bd042b..1aa5130 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -107,7 +107,8 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) } } -void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) +static void i2s_start(struct dw_i2s_dev *dev, + struct snd_pcm_substream *substream) { i2s_write_reg(dev->i2s_base, IER, 1); -- cgit v0.10.2 From 949e6bc75fea779b433679601641ea641456283b Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 18:58:04 +0100 Subject: ASoC: arizona: Rename current rates tables to bclk_rates They're the rates for the BCLK, not for the sample rate, so rename so that we don't confuse ourselves. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 3b5730b..67760b4 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -363,7 +363,7 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) return 0; } -static const int arizona_48k_rates[] = { +static const int arizona_48k_bclk_rates[] = { -1, 48000, 64000, @@ -385,7 +385,7 @@ static const int arizona_48k_rates[] = { 24576000, }; -static const int arizona_44k1_rates[] = { +static const int arizona_44k1_bclk_rates[] = { -1, 44100, 58800, @@ -445,17 +445,17 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, int bclk, lrclk, wl, frame, sr_val; if (params_rate(params) % 8000) - rates = &arizona_44k1_rates[0]; + rates = &arizona_44k1_bclk_rates[0]; else - rates = &arizona_48k_rates[0]; + rates = &arizona_48k_bclk_rates[0]; - for (i = 0; i < ARRAY_SIZE(arizona_44k1_rates); i++) { + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] == snd_soc_params_to_bclk(params)) { bclk = i; break; } } - if (i == ARRAY_SIZE(arizona_44k1_rates)) { + if (i == ARRAY_SIZE(arizona_44k1_bclk_rates)) { arizona_aif_err(dai, "Unsupported sample rate %dHz\n", params_rate(params)); return -EINVAL; -- cgit v0.10.2 From 5b2eec3f98e08a8442ada41c4a63658b95a355f2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 17:32:05 +0100 Subject: ASoC: arizona: Implement AIF clock configuration Allow the user to select which of the system clocks each AIF is referenced to and constran the DAI to the set of frequencies which can be generated from that clock. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 67760b4..8e5246c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -385,6 +385,29 @@ static const int arizona_48k_bclk_rates[] = { 24576000, }; +static const unsigned int arizona_48k_rates[] = { + 12000, + 24000, + 48000, + 96000, + 192000, + 384000, + 768000, + 4000, + 8000, + 16000, + 32000, + 64000, + 128000, + 256000, + 512000, +}; + +static const struct snd_pcm_hw_constraint_list arizona_48k_constraint = { + .count = ARRAY_SIZE(arizona_48k_rates), + .list = arizona_48k_rates, +}; + static const int arizona_44k1_bclk_rates[] = { -1, 44100, @@ -407,6 +430,21 @@ static const int arizona_44k1_bclk_rates[] = { 22579200, }; +static const unsigned int arizona_44k1_rates[] = { + 11025, + 22050, + 44100, + 88200, + 176400, + 352800, + 705600, +}; + +static const struct snd_pcm_hw_constraint_list arizona_44k1_constraint = { + .count = ARRAY_SIZE(arizona_44k1_rates), + .list = arizona_44k1_rates, +}; + static int arizona_sr_vals[] = { 0, 12000, @@ -434,6 +472,36 @@ static int arizona_sr_vals[] = { 512000, }; +static int arizona_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + const struct snd_pcm_hw_constraint_list *constraint; + unsigned int base_rate; + + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + base_rate = priv->sysclk; + break; + case ARIZONA_CLK_ASYNCCLK: + base_rate = priv->asyncclk; + break; + default: + return 0; + } + + if (base_rate % 8000) + constraint = &arizona_44k1_constraint; + else + constraint = &arizona_48k_constraint; + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + constraint); +} + static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -501,11 +569,49 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return 0; } +static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + case ARIZONA_CLK_ASYNCCLK: + break; + default: + return -EINVAL; + } + + if (clk_id != dai_priv->clk && dai->active) { + dev_err(codec->dev, "Can't change clock on active DAI %d\n", + dai->id); + return -EBUSY; + } + + dai_priv->clk = clk_id; + + return 0; +} + const struct snd_soc_dai_ops arizona_dai_ops = { + .startup = arizona_startup, .set_fmt = arizona_set_fmt, .hw_params = arizona_hw_params, + .set_sysclk = arizona_dai_set_sysclk, }; +int arizona_init_dai(struct arizona_priv *priv, int id) +{ + struct arizona_dai_priv *dai_priv = &priv->dai[id]; + + dai_priv->clk = ARIZONA_CLK_SYSCLK; + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_dai); + static irqreturn_t arizona_fll_lock(int irq, void *data) { struct arizona_fll *fll = data; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 8c2ca1d..896711e 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -44,12 +44,19 @@ #define ARIZONA_MIXER_VOL_SHIFT 1 #define ARIZONA_MIXER_VOL_WIDTH 7 +#define ARIZONA_MAX_DAI 3 + struct arizona; +struct arizona_dai_priv { + int clk; +}; + struct arizona_priv { struct arizona *arizona; int sysclk; int asyncclk; + struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; }; #define ARIZONA_NUM_MIXER_INPUTS 55 @@ -146,4 +153,6 @@ extern int arizona_init_fll(struct arizona *arizona, int id, int base, extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); +extern int arizona_init_dai(struct arizona_priv *priv, int dai); + #endif diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index e76c41e..be74a78 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -832,6 +832,9 @@ static int __devinit wm5102_probe(struct platform_device *pdev) ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, &wm5102->fll[1]); + for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++) + arizona_init_dai(&wm5102->core, i); + /* Latch volume update bits */ for (i = 0; i < ARRAY_SIZE(wm5102_digital_vu); i++) regmap_update_bits(arizona->regmap, wm5102_digital_vu[i], -- cgit v0.10.2 From 5001765f992423fdfb82f42f548d3a51b9590186 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 19:07:09 +0100 Subject: ASoC: arizona: Be more forgiving in BCLK selection Allow any BCLK which can be divided down to generate LRCLK, not just the lowest possible BCLK to clock out the samples. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8e5246c..8e066eb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -518,7 +518,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, rates = &arizona_48k_bclk_rates[0]; for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { - if (rates[i] == snd_soc_params_to_bclk(params)) { + if (rates[i] >= snd_soc_params_to_bclk(params) && + rates[i] % params_rate(params) == 0) { bclk = i; break; } -- cgit v0.10.2 From ef3207c503519bf33a114af3a780dfd00cfd5ce4 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Wed, 4 Jul 2012 15:38:27 +0800 Subject: ASoC: fsl: remove unneeded AUDMUX register setting from imx-sgtl5000 If we don't set IMX_AUDMUX_V2_PTCR_TCLKDIR in the AUDMUX PTCR register (means Tx clock pin is input), we don't need to set IMX_AUDMUX_V2_PTCR_TCSEL as well. Since both i.MX35, i.MX51 and i.MX6 datasheet says "When Tx clock pin set as an input, the TCSEL settings are ignored". Signed-off-by: Hui Wang Acked-by: Shawn Guo Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 3a729ca..fb21b17 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -95,8 +95,7 @@ static int __devinit imx_sgtl5000_probe(struct platform_device *pdev) return ret; } imx_audmux_v2_configure_port(ext_port, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TCSEL(int_port), + IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); if (ret) { dev_err(&pdev->dev, "audmux external port setup failed\n"); -- cgit v0.10.2 From 42810d16220484a104317007e3d8fe5269df017b Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Tue, 3 Jul 2012 15:44:58 -0300 Subject: ASoC: imx-mc13783: Add audmux settings for mx27pdk mx27pdk board also has a mc13783 codec. Add support for it and do a run-time machine type check to perform the correct audiomux settings. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index f59c349..549b31f 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -111,22 +111,39 @@ static int __devinit imx_mc13783_probe(struct platform_device *pdev) return ret; } - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, - IMX_AUDMUX_V2_PTCR_SYN, - IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | - IMX_AUDMUX_V2_PDCR_MODE(1) | - IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); - imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, - IMX_AUDMUX_V2_PTCR_SYN | - IMX_AUDMUX_V2_PTCR_TFSDIR | - IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_TCLKDIR | - IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_RFSDIR | - IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | - IMX_AUDMUX_V2_PTCR_RCLKDIR | - IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), - IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + if (machine_is_mx31_3ds()) { + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | + IMX_AUDMUX_V2_PDCR_MODE(1) | + IMX_AUDMUX_V2_PDCR_INMMASK(0xfc)); + imx_audmux_v2_configure_port(MX31_AUDMUX_PORT1_SSI0, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_TCLKDIR | + IMX_AUDMUX_V2_PTCR_TCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RFSDIR | + IMX_AUDMUX_V2_PTCR_RFSEL(MX31_AUDMUX_PORT4_SSI_PINS_4) | + IMX_AUDMUX_V2_PTCR_RCLKDIR | + IMX_AUDMUX_V2_PTCR_RCSEL(MX31_AUDMUX_PORT4_SSI_PINS_4), + IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT4_SSI_PINS_4)); + } else if (machine_is_mx27_3ds()) { + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_TFSDIR | + IMX_AUDMUX_V1_PCR_TCLKDIR | + IMX_AUDMUX_V1_PCR_RFSDIR | + IMX_AUDMUX_V1_PCR_RCLKDIR | + IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) + ); + imx_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + IMX_AUDMUX_V1_PCR_SYN | + IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) + ); + } return ret; } -- cgit v0.10.2 From c013b27a174e8a83d3c8df799aa37c897842efcb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 20:05:57 +0100 Subject: ASoC: arizona: Enable ASYNCCLK domain for audio interfaces If an audio interface is configured to use ASYNCCLK then update the asynchronous sample rate rather than one of our primary sample rates. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8e066eb..d0bcca9 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -507,6 +507,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; int base = dai->driver->base; const int *rates; int i; @@ -530,10 +532,6 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - /* - * We will need to be more flexible than this in future, - * currently we use a single sample rate for the chip. - */ for (i = 0; i < ARRAY_SIZE(arizona_sr_vals); i++) if (arizona_sr_vals[i] == params_rate(params)) break; @@ -552,8 +550,28 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, wl = snd_pcm_format_width(params_format(params)); frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; - snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, - ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + /* + * We will need to be more flexible than this in future, + * currently we use a single sample rate for SYSCLK. + */ + switch (dai_priv->clk) { + case ARIZONA_CLK_SYSCLK: + snd_soc_update_bits(codec, ARIZONA_SAMPLE_RATE_1, + ARIZONA_SAMPLE_RATE_1_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 0); + break; + case ARIZONA_CLK_ASYNCCLK: + snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, + ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); + snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, + ARIZONA_AIF1_RATE_MASK, 8); + break; + default: + arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); + return -EINVAL; + } + snd_soc_update_bits(codec, base + ARIZONA_AIF_BCLK_CTRL, ARIZONA_AIF1_BCLK_FREQ_MASK, bclk); snd_soc_update_bits(codec, base + ARIZONA_AIF_TX_BCLK_RATE, -- cgit v0.10.2 From 9498822d753d241fc93fbeebc17e668cf3023cf7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 4 Jul 2012 20:07:03 +0100 Subject: ASoC: wm5102: Allow routing through the ASRCs This enables common telephony use cases. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index be74a78..3827fa2 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -275,6 +275,11 @@ ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + static const struct snd_soc_dapm_widget wm5102_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, 0, NULL, 0), @@ -363,6 +368,15 @@ SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, @@ -491,6 +505,11 @@ ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + SND_SOC_DAPM_OUTPUT("HPOUT1L"), SND_SOC_DAPM_OUTPUT("HPOUT1R"), SND_SOC_DAPM_OUTPUT("HPOUT2L"), @@ -539,7 +558,11 @@ SND_SOC_DAPM_OUTPUT("SPKDAT1R"), { name, "LHPF1", "LHPF1" }, \ { name, "LHPF2", "LHPF2" }, \ { name, "LHPF3", "LHPF3" }, \ - { name, "LHPF4", "LHPF4" } + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { { "AIF2 Capture", NULL, "DBVDD2" }, @@ -660,6 +683,11 @@ static const struct snd_soc_dapm_route wm5102_dapm_routes[] = { ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + { "HPOUT1L", NULL, "OUT1L" }, { "HPOUT1R", NULL, "OUT1R" }, -- cgit v0.10.2 From bcbf4a69ee6ca68d62440bc936a3c977c2141a66 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jul 2012 14:30:59 +0100 Subject: ASoC: wm1250-ev1: Flag all supported rates in the DAI Not previously noticed due to normal usage being with CODEC<->CODEC links. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm1250-ev1.c b/sound/soc/codecs/wm1250-ev1.c index e0b51e9..951d7b4 100644 --- a/sound/soc/codecs/wm1250-ev1.c +++ b/sound/soc/codecs/wm1250-ev1.c @@ -121,20 +121,23 @@ static const struct snd_soc_dai_ops wm1250_ev1_ops = { .hw_params = wm1250_ev1_hw_params, }; +#define WM1250_EV1_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_64000) + static struct snd_soc_dai_driver wm1250_ev1_dai = { .name = "wm1250-ev1", .playback = { .stream_name = "Playback", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .capture = { .stream_name = "Capture", .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000, + .rates = WM1250_EV1_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE, }, .ops = &wm1250_ev1_ops, -- cgit v0.10.2 From 5cb9b7482270972421a1f2d4145efc60d7ee1176 Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 6 Jul 2012 16:54:52 +0100 Subject: ASoC: pcm: Clean up logging in soc_new_pcm() Use dev_ style logging throughout soc_new_pcm() Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 48fd15b..7063b8f 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2003,7 +2003,6 @@ static int dpcm_fe_dai_close(struct snd_pcm_substream *fe_substream) /* create a new pcm */ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) { - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; @@ -2042,7 +2041,8 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture, &pcm); } if (ret < 0) { - printk(KERN_ERR "asoc: can't create pcm for codec %s\n", codec->name); + dev_err(rtd->card->dev, "can't create pcm for %s\n", + rtd->dai_link->name); return ret; } dev_dbg(rtd->card->dev, "registered pcm #%d %s\n",num, new_name); @@ -2099,14 +2099,14 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) if (platform->driver->pcm_new) { ret = platform->driver->pcm_new(rtd); if (ret < 0) { - pr_err("asoc: platform pcm constructor failed\n"); + dev_err(platform->dev, "pcm constructor failed\n"); return ret; } } pcm->private_free = platform->driver->pcm_free; out: - printk(KERN_INFO "asoc: %s <-> %s mapping ok\n", codec_dai->name, + dev_info(rtd->card->dev, " %s <-> %s mapping ok\n", codec_dai->name, cpu_dai->name); return ret; } -- cgit v0.10.2 From 3e4536546beb5295e6e0459e745327ebffeade9d Mon Sep 17 00:00:00 2001 From: Simon Wilson Date: Fri, 6 Jul 2012 17:04:17 +0100 Subject: ASoC: twl6040: fix spelling mistake Fix spelling mistake in "High-Performance" option of twl6040 power mode. Signed-off-by: Simon Wilson Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index a36e9fc..0ff1e70 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -553,7 +553,7 @@ static const struct snd_kcontrol_new vibrar_mux_controls = /* Headset power mode */ static const char *twl6040_power_mode_texts[] = { - "Low-Power", "High-Perfomance", + "Low-Power", "High-Performance", }; static const struct soc_enum twl6040_power_mode_enum = -- cgit v0.10.2 From 3ac3f5ca91afc03587b1d2d642f126efc5be37ca Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Fri, 6 Jul 2012 17:07:00 +0100 Subject: ASoC: dpcm: Allow FE to be opened without valid BE routes. Some userspace will open a PCM device and then configure mixers for routing before triggering. This patch allows userspace to do this sequence. Signed-off-by: Liam Girdwood Signed-off-by: Mark Brown diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7063b8f..ef22d0b 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1955,10 +1955,8 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream) fe->dpcm[stream].runtime = fe_substream->runtime; if (dpcm_path_get(fe, stream, &list) <= 0) { - dev_warn(fe->dev, "asoc: %s no valid %s route\n", + dev_dbg(fe->dev, "asoc: %s no valid %s route\n", fe->dai_link->name, stream ? "capture" : "playback"); - mutex_unlock(&fe->card->mutex); - return -EINVAL; } /* calculate valid and active FE <-> BE dpcms */ -- cgit v0.10.2 From fabd03842b77b1eb6c9b08c79be86fa38afbe310 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jul 2012 17:20:06 +0100 Subject: ASoC: dapm: Mark widgets as dirty when a route is added If we add a new route at runtime then we'll need to recheck the connections to the affected widgets. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 912330b..19fda13 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2251,6 +2251,10 @@ static int snd_soc_dapm_add_route(struct snd_soc_dapm_context *dapm, path->connect = 0; return 0; } + + dapm_mark_dirty(wsource, "Route added"); + dapm_mark_dirty(wsink, "Route added"); + return 0; err: -- cgit v0.10.2 From efcc3c61b9b1e4f764e14c48c553e6d477f40512 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jul 2012 17:24:19 +0100 Subject: ASoC: dapm: Allow routes to be deleted at runtime Since we're now relying on DAPM for things like enabling clocks when we reparent the clocks for widgets we need to either use conditional routes (which are expensive) or remove routes at runtime. Add a route removal API to support this use case. Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 05559e5..abe373d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -374,6 +374,8 @@ int snd_soc_dapm_new_widgets(struct snd_soc_dapm_context *dapm); void snd_soc_dapm_free(struct snd_soc_dapm_context *dapm); int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); +int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num); int snd_soc_dapm_weak_routes(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route, int num); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 19fda13..4ba47aa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2264,6 +2264,59 @@ err: return ret; } +static int snd_soc_dapm_del_route(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route) +{ + struct snd_soc_dapm_path *path, *p; + const char *sink; + const char *source; + char prefixed_sink[80]; + char prefixed_source[80]; + + if (route->control) { + dev_err(dapm->dev, + "Removal of routes with controls not supported\n"); + return -EINVAL; + } + + if (dapm->codec && dapm->codec->name_prefix) { + snprintf(prefixed_sink, sizeof(prefixed_sink), "%s %s", + dapm->codec->name_prefix, route->sink); + sink = prefixed_sink; + snprintf(prefixed_source, sizeof(prefixed_source), "%s %s", + dapm->codec->name_prefix, route->source); + source = prefixed_source; + } else { + sink = route->sink; + source = route->source; + } + + path = NULL; + list_for_each_entry(p, &dapm->card->paths, list) { + if (strcmp(p->source->name, source) != 0) + continue; + if (strcmp(p->sink->name, sink) != 0) + continue; + path = p; + break; + } + + if (path) { + dapm_mark_dirty(path->source, "Route removed"); + dapm_mark_dirty(path->sink, "Route removed"); + + list_del(&path->list); + list_del(&path->list_sink); + list_del(&path->list_source); + kfree(path); + } else { + dev_warn(dapm->dev, "Route %s->%s does not exist\n", + source, sink); + } + + return 0; +} + /** * snd_soc_dapm_add_routes - Add routes between DAPM widgets * @dapm: DAPM context @@ -2298,6 +2351,30 @@ int snd_soc_dapm_add_routes(struct snd_soc_dapm_context *dapm, } EXPORT_SYMBOL_GPL(snd_soc_dapm_add_routes); +/** + * snd_soc_dapm_del_routes - Remove routes between DAPM widgets + * @dapm: DAPM context + * @route: audio routes + * @num: number of routes + * + * Removes routes from the DAPM context. + */ +int snd_soc_dapm_del_routes(struct snd_soc_dapm_context *dapm, + const struct snd_soc_dapm_route *route, int num) +{ + int i, ret = 0; + + mutex_lock_nested(&dapm->card->dapm_mutex, SND_SOC_DAPM_CLASS_INIT); + for (i = 0; i < num; i++) { + snd_soc_dapm_del_route(dapm, route); + route++; + } + mutex_unlock(&dapm->card->dapm_mutex); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_dapm_del_routes); + static int snd_soc_dapm_weak_route(struct snd_soc_dapm_context *dapm, const struct snd_soc_dapm_route *route) { -- cgit v0.10.2 From 410837a7a29efa2402f496215244569c988bf0db Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 5 Jul 2012 17:26:59 +0100 Subject: ASoC: arizona: Change DAPM routes for AIF clocks when we change them Signed-off-by: Mark Brown Acked-by: Liam Girdwood diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d0bcca9..901b53e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -588,12 +588,25 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, return 0; } +static const char *arizona_dai_clk_str(int clk_id) +{ + switch (clk_id) { + case ARIZONA_CLK_SYSCLK: + return "SYSCLK"; + case ARIZONA_CLK_ASYNCCLK: + return "ASYNCCLK"; + default: + return "Unknown clock"; + } +} + static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { struct snd_soc_codec *codec = dai->codec; struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); struct arizona_dai_priv *dai_priv = &priv->dai[dai->id - 1]; + struct snd_soc_dapm_route routes[2]; switch (clk_id) { case ARIZONA_CLK_SYSCLK: @@ -603,15 +616,28 @@ static int arizona_dai_set_sysclk(struct snd_soc_dai *dai, return -EINVAL; } - if (clk_id != dai_priv->clk && dai->active) { + if (clk_id == dai_priv->clk) + return 0; + + if (dai->active) { dev_err(codec->dev, "Can't change clock on active DAI %d\n", dai->id); return -EBUSY; } - dai_priv->clk = clk_id; + memset(&routes, 0, sizeof(routes)); + routes[0].sink = dai->driver->capture.stream_name; + routes[1].sink = dai->driver->playback.stream_name; - return 0; + routes[0].source = arizona_dai_clk_str(dai_priv->clk); + routes[1].source = arizona_dai_clk_str(dai_priv->clk); + snd_soc_dapm_del_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + routes[0].source = arizona_dai_clk_str(clk_id); + routes[1].source = arizona_dai_clk_str(clk_id); + snd_soc_dapm_add_routes(&codec->dapm, routes, ARRAY_SIZE(routes)); + + return snd_soc_dapm_sync(&codec->dapm); } const struct snd_soc_dai_ops arizona_dai_ops = { -- cgit v0.10.2 From a837987e7b36a9056cd17c0967efe1ce73a102ff Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Jul 2012 12:16:41 +0100 Subject: ASoC: arizona: Export dai_ops Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 901b53e..0be04b5 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -646,6 +646,7 @@ const struct snd_soc_dai_ops arizona_dai_ops = { .hw_params = arizona_hw_params, .set_sysclk = arizona_dai_set_sysclk, }; +EXPORT_SYMBOL_GPL(arizona_dai_ops); int arizona_init_dai(struct arizona_priv *priv, int id) { -- cgit v0.10.2 From c9c56fd0b766f6f3cd19c83945954ff5b06afc5f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Jul 2012 19:09:01 +0100 Subject: ASoC: arizona: Add IN4 to the mixer tables Some devices have four input structures rather than three. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0be04b5..f3680c3 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -79,6 +79,8 @@ const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "IN2R", "IN3L", "IN3R", + "IN4L", + "IN4R", "AIF1RX1", "AIF1RX2", "AIF1RX3", @@ -138,6 +140,8 @@ int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS] = { 0x13, 0x14, 0x15, + 0x16, + 0x17, 0x20, /* AIF1RX1 */ 0x21, 0x22, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 896711e..b894b64 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -59,7 +59,7 @@ struct arizona_priv { struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; }; -#define ARIZONA_NUM_MIXER_INPUTS 55 +#define ARIZONA_NUM_MIXER_INPUTS 57 extern const unsigned int arizona_mixer_tlv[]; extern const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS]; -- cgit v0.10.2 From bc9dce5853ced3b7a5bc79f1101a4c4b0a752f3e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Jul 2012 19:08:23 +0100 Subject: ASoC: wm5102: Fix cut'n'paste for digital volume registers The analogue PGA shifts were used; this makes no practical difference as the values are the same. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 3827fa2..7a6a11a 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -68,13 +68,13 @@ SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_PGA_VOL_SHIFT, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, 0xbf, 0, digital_tlv), ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), -- cgit v0.10.2 From 774441915de8402103ffe5bf68656f160fefc86f Mon Sep 17 00:00:00 2001 From: Jiri Prchal Date: Mon, 9 Jul 2012 09:48:44 +0200 Subject: ASoC: tlv320aic3x: add deemphasis switch This patch adds missing deemphasis switch. Signed-off-by: Jiri Prchal Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 174de66..7933b8c 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -356,6 +356,9 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), + /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, 0, 119, 0, adc_tlv), -- cgit v0.10.2 From bb1daa803c733462248421dd9beed84fecf1745e Mon Sep 17 00:00:00 2001 From: Jiri Prchal Date: Tue, 10 Jul 2012 14:35:11 +0200 Subject: ASoC: tlv320aic3x: add AGC settings This patch adds AGC target level and times settings for TLV320AIC3x. Enums uses small arrays of two channels left and right since it uses different registers. Signed-off-by: Jiri Prchal Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 7933b8c..0d2f8c4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -231,6 +231,25 @@ static const struct soc_enum aic3x_enum[] = { SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const struct soc_enum aic3x_agc_level_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), +}; + +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const struct soc_enum aic3x_agc_attack_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), +}; + +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const struct soc_enum aic3x_agc_decay_enum[] = { + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -355,6 +374,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_agc_level_enum[0]), + SOC_ENUM("Right AGC Target level", aic3x_agc_level_enum[1]), + SOC_ENUM("Left AGC Attack time", aic3x_agc_attack_enum[0]), + SOC_ENUM("Right AGC Attack time", aic3x_agc_attack_enum[1]), + SOC_ENUM("Left AGC Decay time", aic3x_agc_decay_enum[0]), + SOC_ENUM("Right AGC Decay time", aic3x_agc_decay_enum[1]), /* De-emphasis */ SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), -- cgit v0.10.2 From a1f34af0ec35e3131d65e0ae4cec6b048cba3e88 Mon Sep 17 00:00:00 2001 From: Jiri Prchal Date: Tue, 10 Jul 2012 14:36:58 +0200 Subject: ASoC: tlv320aic3x: add input clock selection This patch adds input selection of main codec clock - from what pin. Both registers set same value since codec uses clock divider or pll at one time. Signed-off-by: Jiri Prchal Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 0d2f8c4..b94f81f 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1002,6 +1002,12 @@ static int aic3x_set_dai_sysclk(struct snd_soc_dai *codec_dai, struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, + clk_id << PLLCLK_IN_SHIFT); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, + clk_id << CLKDIV_IN_SHIFT); + aic3x->sysclk = freq; return 0; } diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 5da5eb3..149338b 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -195,6 +195,14 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define PLLCLK_IN_SHIFT 4 +#define CLKDIV_IN_MASK 0xc0 +#define CLKDIV_IN_SHIFT 6 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 /* Software reset register bits */ #define SOFT_RESET 0x80 -- cgit v0.10.2 From 2b4d39fc2a80e271ac8d44fccd02277a4b63c557 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 10 Jul 2012 17:03:46 +0100 Subject: ASoC: arizona: Support variable FLL VCO multipliers Some Arizona chips have a higher frequency for the FLL VCO, support this in the common code. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index f3680c3..5c9caca 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -734,9 +734,9 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Apply the division for our remaining calculations */ Fref /= div; - /* Fvco should be 90-100MHz; don't check the upper bound */ + /* Fvco should be over the targt; don't check the upper bound */ div = 1; - while (Fout * div < 90000000) { + while (Fout * div < 90000000 * fll->vco_mult) { div++; if (div > 7) { arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", @@ -744,7 +744,7 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } } - target = Fout * div; + target = Fout * div / fll->vco_mult; cfg->outdiv = div; arizona_fll_dbg(fll, "Fvco=%dHz\n", target); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index b894b64..59caca8 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -141,6 +141,7 @@ struct arizona_fll { struct arizona *arizona; int id; unsigned int base; + unsigned int vco_mult; struct completion lock; struct completion ok; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 7a6a11a..6537f16 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -853,6 +853,9 @@ static int __devinit wm5102_probe(struct platform_device *pdev) wm5102->core.arizona = arizona; + for (i = 0; i < ARRAY_SIZE(wm5102->fll); i++) + wm5102->fll[i].vco_mult = 1; + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, &wm5102->fll[0]); -- cgit v0.10.2 From f96985e3b3cfcd2d21faca79863fb34533d575aa Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 11 Jul 2012 09:41:23 +0300 Subject: ASoC: STA529: fix an error message GCC complains that "ret" is uninitialized here. Signed-off-by: Dan Carpenter Acked-By: Rajeev Kumar Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index a9f34c7..0c225cd 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -397,8 +397,9 @@ static __devinit int sta529_i2c_probe(struct i2c_client *i2c, sta529->regmap = devm_regmap_init_i2c(i2c, &sta529_regmap); if (IS_ERR(sta529->regmap)) { + ret = PTR_ERR(sta529->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); - return PTR_ERR(sta529->regmap); + return ret; } i2c_set_clientdata(i2c, sta529); -- cgit v0.10.2 From 5c6af635fd77251b753cb1c07a6a6f306ba4e287 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 9 Jul 2012 19:09:41 +0100 Subject: ASoC: wm5110: Add audio CODEC driver The WM5110 is a highly integrated low power audio subsystem for smartphones, tablets and other portable audio devices. It combines an advanced DSP feature set with a flexible, high performance audio hub CODEC. This patch adds the audio CODEC driver for the device. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index bbcb038..9f8e859 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -75,6 +75,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_WM2200 if I2C select SND_SOC_WM5100 if I2C select SND_SOC_WM5102 if MFD_WM5102 + select SND_SOC_WM5110 if MFD_WM5110 select SND_SOC_WM8350 if MFD_WM8350 select SND_SOC_WM8400 if MFD_WM8400 select SND_SOC_WM8510 if SND_SOC_I2C_AND_SPI @@ -134,7 +135,9 @@ config SND_SOC_88PM860X config SND_SOC_ARIZONA tristate default y if SND_SOC_WM5102=y + default y if SND_SOC_WM5110=y default m if SND_SOC_WM5102=m + default m if SND_SOC_WM5110=m config SND_SOC_WM_HUBS tristate @@ -338,6 +341,9 @@ config SND_SOC_WM5100 config SND_SOC_WM5102 tristate +config SND_SOC_WM5110 + tristate + config SND_SOC_WM8350 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 8da3d22..34148bb 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -66,6 +66,7 @@ snd-soc-wm2000-objs := wm2000.o snd-soc-wm2200-objs := wm2200.o snd-soc-wm5100-objs := wm5100.o wm5100-tables.o snd-soc-wm5102-objs := wm5102.o +snd-soc-wm5110-objs := wm5110.o snd-soc-wm8350-objs := wm8350.o snd-soc-wm8400-objs := wm8400.o snd-soc-wm8510-objs := wm8510.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_WM2000) += snd-soc-wm2000.o obj-$(CONFIG_SND_SOC_WM2200) += snd-soc-wm2200.o obj-$(CONFIG_SND_SOC_WM5100) += snd-soc-wm5100.o obj-$(CONFIG_SND_SOC_WM5102) += snd-soc-wm5102.o +obj-$(CONFIG_SND_SOC_WM5110) += snd-soc-wm5110.o obj-$(CONFIG_SND_SOC_WM8350) += snd-soc-wm8350.o obj-$(CONFIG_SND_SOC_WM8400) += snd-soc-wm8400.o obj-$(CONFIG_SND_SOC_WM8510) += snd-soc-wm8510.o diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c new file mode 100644 index 0000000..8033f70 --- /dev/null +++ b/sound/soc/codecs/wm5110.c @@ -0,0 +1,950 @@ +/* + * wm5110.c -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include "arizona.h" +#include "wm5110.h" + +struct wm5110_priv { + struct arizona_priv core; + struct arizona_fll fll[2]; +}; + +static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); +static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); +static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); +static DECLARE_TLV_DB_SCALE(noise_tlv, 0, 600, 0); + +static const struct snd_kcontrol_new wm5110_snd_controls[] = { +SOC_SINGLE("IN1 High Performance Switch", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN2 High Performance Switch", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN3 High Performance Switch", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3_OSR_SHIFT, 1, 0), +SOC_SINGLE("IN4 High Performance Switch", ARIZONA_IN4L_CONTROL, + ARIZONA_IN4_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R_RANGE_TLV("IN1 Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1R_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN2 Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2R_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("IN3 Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3R_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), + +SOC_DOUBLE_R("IN1 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN2 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN3 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("IN4 Digital Switch", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("IN1 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, ARIZONA_IN1L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN2 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, ARIZONA_IN2L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN3 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, ARIZONA_IN3L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("IN4 Digital Volume", ARIZONA_ADC_DIGITAL_VOLUME_4L, + ARIZONA_ADC_DIGITAL_VOLUME_4R, ARIZONA_IN4L_DIG_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +ARIZONA_MIXER_CONTROLS("EQ1", ARIZONA_EQ1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B3 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B3 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B3 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B3 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B3_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B4 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B4_GAIN_SHIFT, + 24, 0, eq_tlv), +SOC_SINGLE_TLV("EQ4 B5 Volume", ARIZONA_EQ4_2, ARIZONA_EQ4_B5_GAIN_SHIFT, + 24, 0, eq_tlv), + +ARIZONA_MIXER_CONTROLS("DRC1L", ARIZONA_DRC1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC1R", ARIZONA_DRC1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2L", ARIZONA_DRC2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("DRC2R", ARIZONA_DRC2RMIX_INPUT_1_SOURCE), + +SND_SOC_BYTES_MASK("DRC1", ARIZONA_DRC1_CTRL1, 5, + ARIZONA_DRC1R_ENA | ARIZONA_DRC1L_ENA), +SND_SOC_BYTES_MASK("DRC2", ARIZONA_DRC2_CTRL1, 5, + ARIZONA_DRC2R_ENA | ARIZONA_DRC2L_ENA), + +ARIZONA_MIXER_CONTROLS("LHPF1", ARIZONA_HPLP1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF2", ARIZONA_HPLP2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF3", ARIZONA_HPLP3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("LHPF4", ARIZONA_HPLP4MIX_INPUT_1_SOURCE), + +SOC_ENUM("LHPF1 Mode", arizona_lhpf1_mode), +SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), +SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), +SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), + +ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), + +SOC_SINGLE_TLV("Noise Generator Volume", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_GAIN_SHIFT, 0x16, 0, noise_tlv), + +ARIZONA_MIXER_CONTROLS("HPOUT1L", ARIZONA_OUT1LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT1R", ARIZONA_OUT1RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2L", ARIZONA_OUT2LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("HPOUT2R", ARIZONA_OUT2RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("EPOUT", ARIZONA_OUT3LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTL", ARIZONA_OUT4LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), + +SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 1, 0), +SOC_SINGLE("OUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 1, 0), +SOC_SINGLE("EPOUT High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 1, 0), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, + ARIZONA_OUT5_OSR_SHIFT, 1, 0), +SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, + ARIZONA_OUT6_OSR_SHIFT, 1, 0), + +SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("OUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_MUTE_SHIFT, 1, 1), +SOC_SINGLE("EPOUT Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("Speaker Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_MUTE_SHIFT, 1, 1), +SOC_DOUBLE_R("SPKDAT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_MUTE_SHIFT, 1, 1), + +SOC_DOUBLE_R_TLV("HPOUT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("OUT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, ARIZONA_OUT2L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_SINGLE_TLV("EPOUT Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_OUT3L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("Speaker Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, ARIZONA_OUT4L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, + 0xbf, 0, digital_tlv), +SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, + ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, + 0xbf, 0, digital_tlv), + +SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUTPUT_PATH_CONFIG_1R, + ARIZONA_OUT1L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_DOUBLE_R_RANGE_TLV("OUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUTPUT_PATH_CONFIG_2R, + ARIZONA_OUT2L_PGA_VOL_SHIFT, + 0x34, 0x40, 0, ana_tlv), +SOC_SINGLE_RANGE_TLV("EPOUT Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), + +SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, + ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), +SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, + ARIZONA_SPK2R_MUTE_SHIFT, 1, 1), + +ARIZONA_MIXER_CONTROLS("AIF1TX1", ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX2", ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX3", ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX4", ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX5", ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX6", ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX7", ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF1TX8", ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF2TX1", ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF2TX2", ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE), + +ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), +ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), +}; + +ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(EQ4, ARIZONA_EQ4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(DRC1L, ARIZONA_DRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC1R, ARIZONA_DRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2L, ARIZONA_DRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(DRC2R, ARIZONA_DRC2RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(LHPF1, ARIZONA_HPLP1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF2, ARIZONA_HPLP2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF3, ARIZONA_HPLP3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(LHPF4, ARIZONA_HPLP4MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(Mic, ARIZONA_MICMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(Noise, ARIZONA_NOISEMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(PWM1, ARIZONA_PWM1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(PWM2, ARIZONA_PWM2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(OUT1L, ARIZONA_OUT1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT1R, ARIZONA_OUT1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2L, ARIZONA_OUT2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT2R, ARIZONA_OUT2RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(OUT3, ARIZONA_OUT3LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTL, ARIZONA_OUT4LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKOUTR, ARIZONA_OUT4RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1L, ARIZONA_OUT5LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT1R, ARIZONA_OUT5RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2L, ARIZONA_OUT6LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(SPKDAT2R, ARIZONA_OUT6RMIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF1TX1, ARIZONA_AIF1TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX2, ARIZONA_AIF1TX2MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX3, ARIZONA_AIF1TX3MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX4, ARIZONA_AIF1TX4MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX5, ARIZONA_AIF1TX5MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX6, ARIZONA_AIF1TX6MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX7, ARIZONA_AIF1TX7MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF1TX8, ARIZONA_AIF1TX8MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF2TX1, ARIZONA_AIF2TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF2TX2, ARIZONA_AIF2TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(AIF3TX1, ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(AIF3TX2, ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE); + +ARIZONA_MIXER_ENUMS(ASRC1L, ARIZONA_ASRC1LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC1R, ARIZONA_ASRC1RMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2L, ARIZONA_ASRC2LMIX_INPUT_1_SOURCE); +ARIZONA_MIXER_ENUMS(ASRC2R, ARIZONA_ASRC2RMIX_INPUT_1_SOURCE); + +static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, + ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD2", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("DBVDD3", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("CPVDD", 20), +SND_SOC_DAPM_REGULATOR_SUPPLY("MICVDD", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDL", 0), +SND_SOC_DAPM_REGULATOR_SUPPLY("SPKVDDR", 0), + +SND_SOC_DAPM_SIGGEN("TONE"), +SND_SOC_DAPM_SIGGEN("NOISE"), + +SND_SOC_DAPM_INPUT("IN1L"), +SND_SOC_DAPM_INPUT("IN1R"), +SND_SOC_DAPM_INPUT("IN2L"), +SND_SOC_DAPM_INPUT("IN2R"), +SND_SOC_DAPM_INPUT("IN3L"), +SND_SOC_DAPM_INPUT("IN3R"), +SND_SOC_DAPM_INPUT("IN4L"), +SND_SOC_DAPM_INPUT("IN4R"), + +SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT, + 0, NULL, 0, arizona_in_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS2", ARIZONA_MIC_BIAS_CTRL_2, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("MICBIAS3", ARIZONA_MIC_BIAS_CTRL_3, + ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Noise Generator", ARIZONA_COMFORT_NOISE_GENERATOR, + ARIZONA_NOISE_GEN_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Tone Generator 1", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("Tone Generator 2", ARIZONA_TONE_GENERATOR_1, + ARIZONA_TONE2_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("Mic Mute Mixer", ARIZONA_MIC_NOISE_MIX_CONTROL_1, + ARIZONA_MICMUTE_MIX_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("EQ1", ARIZONA_EQ1_1, ARIZONA_EQ1_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ2", ARIZONA_EQ2_1, ARIZONA_EQ2_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ3", ARIZONA_EQ3_1, ARIZONA_EQ3_ENA_SHIFT, 0, NULL, 0), +SND_SOC_DAPM_PGA("EQ4", ARIZONA_EQ4_1, ARIZONA_EQ4_ENA_SHIFT, 0, NULL, 0), + +SND_SOC_DAPM_PGA("DRC1L", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC1R", ARIZONA_DRC1_CTRL1, ARIZONA_DRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2L", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("DRC2R", ARIZONA_DRC2_CTRL1, ARIZONA_DRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("LHPF1", ARIZONA_HPLPF1_1, ARIZONA_LHPF1_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF2", ARIZONA_HPLPF2_1, ARIZONA_LHPF2_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF3", ARIZONA_HPLPF3_1, ARIZONA_LHPF3_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("LHPF4", ARIZONA_HPLPF4_1, ARIZONA_LHPF4_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_PGA("PWM1 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM1_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_PGA("PWM2 Driver", ARIZONA_PWM_DRIVE_1, ARIZONA_PWM2_ENA_SHIFT, + 0, NULL, 0), + +SND_SOC_DAPM_PGA("ASRC1L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC1R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC1R_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2L", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2L_ENA_SHIFT, 0, + NULL, 0), +SND_SOC_DAPM_PGA("ASRC2R", ARIZONA_ASRC_ENABLE, ARIZONA_ASRC2R_ENA_SHIFT, 0, + NULL, 0), + +SND_SOC_DAPM_AIF_OUT("AIF1TX1", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX2", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX3", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX4", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX5", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX6", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX7", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF1TX8", NULL, 0, + ARIZONA_AIF1_TX_ENABLES, ARIZONA_AIF1TX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF1RX1", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX2", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX2_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX3", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX3_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX4", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX4_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX5", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX5_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX6", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX6_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX7", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX7_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF1RX8", NULL, 0, + ARIZONA_AIF1_RX_ENABLES, ARIZONA_AIF1RX8_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF2TX1", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF2TX2", NULL, 0, + ARIZONA_AIF2_TX_ENABLES, ARIZONA_AIF2TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF2RX1", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF2RX2", NULL, 0, + ARIZONA_AIF2_RX_ENABLES, ARIZONA_AIF2RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_OUT("AIF3TX1", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_OUT("AIF3TX2", NULL, 0, + ARIZONA_AIF3_TX_ENABLES, ARIZONA_AIF3TX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX1_ENA_SHIFT, 0), +SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, + ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), + +SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT2R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT5R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT5R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), +SND_SOC_DAPM_PGA_E("OUT6R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT6R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + +ARIZONA_MIXER_WIDGETS(EQ1, "EQ1"), +ARIZONA_MIXER_WIDGETS(EQ2, "EQ2"), +ARIZONA_MIXER_WIDGETS(EQ3, "EQ3"), +ARIZONA_MIXER_WIDGETS(EQ4, "EQ4"), + +ARIZONA_MIXER_WIDGETS(DRC1L, "DRC1L"), +ARIZONA_MIXER_WIDGETS(DRC1R, "DRC1R"), +ARIZONA_MIXER_WIDGETS(DRC2L, "DRC2L"), +ARIZONA_MIXER_WIDGETS(DRC2R, "DRC2R"), + +ARIZONA_MIXER_WIDGETS(LHPF1, "LHPF1"), +ARIZONA_MIXER_WIDGETS(LHPF2, "LHPF2"), +ARIZONA_MIXER_WIDGETS(LHPF3, "LHPF3"), +ARIZONA_MIXER_WIDGETS(LHPF4, "LHPF4"), + +ARIZONA_MIXER_WIDGETS(Mic, "Mic"), +ARIZONA_MIXER_WIDGETS(Noise, "Noise"), + +ARIZONA_MIXER_WIDGETS(PWM1, "PWM1"), +ARIZONA_MIXER_WIDGETS(PWM2, "PWM2"), + +ARIZONA_MIXER_WIDGETS(OUT1L, "HPOUT1L"), +ARIZONA_MIXER_WIDGETS(OUT1R, "HPOUT1R"), +ARIZONA_MIXER_WIDGETS(OUT2L, "HPOUT2L"), +ARIZONA_MIXER_WIDGETS(OUT2R, "HPOUT2R"), +ARIZONA_MIXER_WIDGETS(OUT3, "EPOUT"), +ARIZONA_MIXER_WIDGETS(SPKOUTL, "SPKOUTL"), +ARIZONA_MIXER_WIDGETS(SPKOUTR, "SPKOUTR"), +ARIZONA_MIXER_WIDGETS(SPKDAT1L, "SPKDAT1L"), +ARIZONA_MIXER_WIDGETS(SPKDAT1R, "SPKDAT1R"), +ARIZONA_MIXER_WIDGETS(SPKDAT2L, "SPKDAT2L"), +ARIZONA_MIXER_WIDGETS(SPKDAT2R, "SPKDAT2R"), + +ARIZONA_MIXER_WIDGETS(AIF1TX1, "AIF1TX1"), +ARIZONA_MIXER_WIDGETS(AIF1TX2, "AIF1TX2"), +ARIZONA_MIXER_WIDGETS(AIF1TX3, "AIF1TX3"), +ARIZONA_MIXER_WIDGETS(AIF1TX4, "AIF1TX4"), +ARIZONA_MIXER_WIDGETS(AIF1TX5, "AIF1TX5"), +ARIZONA_MIXER_WIDGETS(AIF1TX6, "AIF1TX6"), +ARIZONA_MIXER_WIDGETS(AIF1TX7, "AIF1TX7"), +ARIZONA_MIXER_WIDGETS(AIF1TX8, "AIF1TX8"), + +ARIZONA_MIXER_WIDGETS(AIF2TX1, "AIF2TX1"), +ARIZONA_MIXER_WIDGETS(AIF2TX2, "AIF2TX2"), + +ARIZONA_MIXER_WIDGETS(AIF3TX1, "AIF3TX1"), +ARIZONA_MIXER_WIDGETS(AIF3TX2, "AIF3TX2"), + +ARIZONA_MIXER_WIDGETS(ASRC1L, "ASRC1L"), +ARIZONA_MIXER_WIDGETS(ASRC1R, "ASRC1R"), +ARIZONA_MIXER_WIDGETS(ASRC2L, "ASRC2L"), +ARIZONA_MIXER_WIDGETS(ASRC2R, "ASRC2R"), + +SND_SOC_DAPM_OUTPUT("HPOUT1L"), +SND_SOC_DAPM_OUTPUT("HPOUT1R"), +SND_SOC_DAPM_OUTPUT("HPOUT2L"), +SND_SOC_DAPM_OUTPUT("HPOUT2R"), +SND_SOC_DAPM_OUTPUT("EPOUTN"), +SND_SOC_DAPM_OUTPUT("EPOUTP"), +SND_SOC_DAPM_OUTPUT("SPKOUTLN"), +SND_SOC_DAPM_OUTPUT("SPKOUTLP"), +SND_SOC_DAPM_OUTPUT("SPKOUTRN"), +SND_SOC_DAPM_OUTPUT("SPKOUTRP"), +SND_SOC_DAPM_OUTPUT("SPKDAT1L"), +SND_SOC_DAPM_OUTPUT("SPKDAT1R"), +SND_SOC_DAPM_OUTPUT("SPKDAT2L"), +SND_SOC_DAPM_OUTPUT("SPKDAT2R"), +}; + +#define ARIZONA_MIXER_INPUT_ROUTES(name) \ + { name, "Noise Generator", "Noise Generator" }, \ + { name, "Tone Generator 1", "Tone Generator 1" }, \ + { name, "Tone Generator 2", "Tone Generator 2" }, \ + { name, "IN1L", "IN1L PGA" }, \ + { name, "IN1R", "IN1R PGA" }, \ + { name, "IN2L", "IN2L PGA" }, \ + { name, "IN2R", "IN2R PGA" }, \ + { name, "IN3L", "IN3L PGA" }, \ + { name, "IN3R", "IN3R PGA" }, \ + { name, "IN4L", "IN4L PGA" }, \ + { name, "IN4R", "IN4R PGA" }, \ + { name, "Mic Mute Mixer", "Mic Mute Mixer" }, \ + { name, "AIF1RX1", "AIF1RX1" }, \ + { name, "AIF1RX2", "AIF1RX2" }, \ + { name, "AIF1RX3", "AIF1RX3" }, \ + { name, "AIF1RX4", "AIF1RX4" }, \ + { name, "AIF1RX5", "AIF1RX5" }, \ + { name, "AIF1RX6", "AIF1RX6" }, \ + { name, "AIF1RX7", "AIF1RX7" }, \ + { name, "AIF1RX8", "AIF1RX8" }, \ + { name, "AIF2RX1", "AIF2RX1" }, \ + { name, "AIF2RX2", "AIF2RX2" }, \ + { name, "AIF3RX1", "AIF3RX1" }, \ + { name, "AIF3RX2", "AIF3RX2" }, \ + { name, "EQ1", "EQ1" }, \ + { name, "EQ2", "EQ2" }, \ + { name, "EQ3", "EQ3" }, \ + { name, "EQ4", "EQ4" }, \ + { name, "DRC1L", "DRC1L" }, \ + { name, "DRC1R", "DRC1R" }, \ + { name, "DRC2L", "DRC2L" }, \ + { name, "DRC2R", "DRC2R" }, \ + { name, "LHPF1", "LHPF1" }, \ + { name, "LHPF2", "LHPF2" }, \ + { name, "LHPF3", "LHPF3" }, \ + { name, "LHPF4", "LHPF4" }, \ + { name, "ASRC1L", "ASRC1L" }, \ + { name, "ASRC1R", "ASRC1R" }, \ + { name, "ASRC2L", "ASRC2L" }, \ + { name, "ASRC2R", "ASRC2R" } + +static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { + { "AIF2 Capture", NULL, "DBVDD2" }, + { "AIF2 Playback", NULL, "DBVDD2" }, + + { "AIF3 Capture", NULL, "DBVDD3" }, + { "AIF3 Playback", NULL, "DBVDD3" }, + + { "OUT1L", NULL, "CPVDD" }, + { "OUT1R", NULL, "CPVDD" }, + { "OUT2L", NULL, "CPVDD" }, + { "OUT2R", NULL, "CPVDD" }, + { "OUT3L", NULL, "CPVDD" }, + + { "OUT4L", NULL, "SPKVDDL" }, + { "OUT4R", NULL, "SPKVDDR" }, + + { "OUT1L", NULL, "SYSCLK" }, + { "OUT1R", NULL, "SYSCLK" }, + { "OUT2L", NULL, "SYSCLK" }, + { "OUT2R", NULL, "SYSCLK" }, + { "OUT3L", NULL, "SYSCLK" }, + { "OUT4L", NULL, "SYSCLK" }, + { "OUT4R", NULL, "SYSCLK" }, + { "OUT5L", NULL, "SYSCLK" }, + { "OUT5R", NULL, "SYSCLK" }, + { "OUT6L", NULL, "SYSCLK" }, + { "OUT6R", NULL, "SYSCLK" }, + + { "MICBIAS1", NULL, "MICVDD" }, + { "MICBIAS2", NULL, "MICVDD" }, + { "MICBIAS3", NULL, "MICVDD" }, + + { "Noise Generator", NULL, "NOISE" }, + { "Tone Generator 1", NULL, "TONE" }, + { "Tone Generator 2", NULL, "TONE" }, + + { "Mic Mute Mixer", NULL, "Noise Mixer" }, + { "Mic Mute Mixer", NULL, "Mic Mixer" }, + + { "AIF1 Capture", NULL, "AIF1TX1" }, + { "AIF1 Capture", NULL, "AIF1TX2" }, + { "AIF1 Capture", NULL, "AIF1TX3" }, + { "AIF1 Capture", NULL, "AIF1TX4" }, + { "AIF1 Capture", NULL, "AIF1TX5" }, + { "AIF1 Capture", NULL, "AIF1TX6" }, + { "AIF1 Capture", NULL, "AIF1TX7" }, + { "AIF1 Capture", NULL, "AIF1TX8" }, + + { "AIF1RX1", NULL, "AIF1 Playback" }, + { "AIF1RX2", NULL, "AIF1 Playback" }, + { "AIF1RX3", NULL, "AIF1 Playback" }, + { "AIF1RX4", NULL, "AIF1 Playback" }, + { "AIF1RX5", NULL, "AIF1 Playback" }, + { "AIF1RX6", NULL, "AIF1 Playback" }, + { "AIF1RX7", NULL, "AIF1 Playback" }, + { "AIF1RX8", NULL, "AIF1 Playback" }, + + { "AIF2 Capture", NULL, "AIF2TX1" }, + { "AIF2 Capture", NULL, "AIF2TX2" }, + + { "AIF2RX1", NULL, "AIF2 Playback" }, + { "AIF2RX2", NULL, "AIF2 Playback" }, + + { "AIF3 Capture", NULL, "AIF3TX1" }, + { "AIF3 Capture", NULL, "AIF3TX2" }, + + { "AIF3RX1", NULL, "AIF3 Playback" }, + { "AIF3RX2", NULL, "AIF3 Playback" }, + + { "AIF1 Playback", NULL, "SYSCLK" }, + { "AIF2 Playback", NULL, "SYSCLK" }, + { "AIF3 Playback", NULL, "SYSCLK" }, + + { "AIF1 Capture", NULL, "SYSCLK" }, + { "AIF2 Capture", NULL, "SYSCLK" }, + { "AIF3 Capture", NULL, "SYSCLK" }, + + ARIZONA_MIXER_ROUTES("OUT1L", "HPOUT1L"), + ARIZONA_MIXER_ROUTES("OUT1R", "HPOUT1R"), + ARIZONA_MIXER_ROUTES("OUT2L", "HPOUT2L"), + ARIZONA_MIXER_ROUTES("OUT2R", "HPOUT2R"), + ARIZONA_MIXER_ROUTES("OUT3L", "EPOUT"), + + ARIZONA_MIXER_ROUTES("OUT4L", "SPKOUTL"), + ARIZONA_MIXER_ROUTES("OUT4R", "SPKOUTR"), + ARIZONA_MIXER_ROUTES("OUT5L", "SPKDAT1L"), + ARIZONA_MIXER_ROUTES("OUT5R", "SPKDAT1R"), + ARIZONA_MIXER_ROUTES("OUT6L", "SPKDAT2L"), + ARIZONA_MIXER_ROUTES("OUT6R", "SPKDAT2R"), + + ARIZONA_MIXER_ROUTES("PWM1 Driver", "PWM1"), + ARIZONA_MIXER_ROUTES("PWM2 Driver", "PWM2"), + + ARIZONA_MIXER_ROUTES("AIF1TX1", "AIF1TX1"), + ARIZONA_MIXER_ROUTES("AIF1TX2", "AIF1TX2"), + ARIZONA_MIXER_ROUTES("AIF1TX3", "AIF1TX3"), + ARIZONA_MIXER_ROUTES("AIF1TX4", "AIF1TX4"), + ARIZONA_MIXER_ROUTES("AIF1TX5", "AIF1TX5"), + ARIZONA_MIXER_ROUTES("AIF1TX6", "AIF1TX6"), + ARIZONA_MIXER_ROUTES("AIF1TX7", "AIF1TX7"), + ARIZONA_MIXER_ROUTES("AIF1TX8", "AIF1TX8"), + + ARIZONA_MIXER_ROUTES("AIF2TX1", "AIF2TX1"), + ARIZONA_MIXER_ROUTES("AIF2TX2", "AIF2TX2"), + + ARIZONA_MIXER_ROUTES("AIF3TX1", "AIF3TX1"), + ARIZONA_MIXER_ROUTES("AIF3TX2", "AIF3TX2"), + + ARIZONA_MIXER_ROUTES("EQ1", "EQ1"), + ARIZONA_MIXER_ROUTES("EQ2", "EQ2"), + ARIZONA_MIXER_ROUTES("EQ3", "EQ3"), + ARIZONA_MIXER_ROUTES("EQ4", "EQ4"), + + ARIZONA_MIXER_ROUTES("DRC1L", "DRC1L"), + ARIZONA_MIXER_ROUTES("DRC1R", "DRC1R"), + ARIZONA_MIXER_ROUTES("DRC2L", "DRC2L"), + ARIZONA_MIXER_ROUTES("DRC2R", "DRC2R"), + + ARIZONA_MIXER_ROUTES("LHPF1", "LHPF1"), + ARIZONA_MIXER_ROUTES("LHPF2", "LHPF2"), + ARIZONA_MIXER_ROUTES("LHPF3", "LHPF3"), + ARIZONA_MIXER_ROUTES("LHPF4", "LHPF4"), + + ARIZONA_MIXER_ROUTES("ASRC1L", "ASRC1L"), + ARIZONA_MIXER_ROUTES("ASRC1R", "ASRC1R"), + ARIZONA_MIXER_ROUTES("ASRC2L", "ASRC2L"), + ARIZONA_MIXER_ROUTES("ASRC2R", "ASRC2R"), + + { "HPOUT1L", NULL, "OUT1L" }, + { "HPOUT1R", NULL, "OUT1R" }, + + { "HPOUT2L", NULL, "OUT2L" }, + { "HPOUT2R", NULL, "OUT2R" }, + + { "EPOUTN", NULL, "OUT3L" }, + { "EPOUTP", NULL, "OUT3L" }, + + { "SPKOUTLN", NULL, "OUT4L" }, + { "SPKOUTLP", NULL, "OUT4L" }, + + { "SPKOUTRN", NULL, "OUT4R" }, + { "SPKOUTRP", NULL, "OUT4R" }, + + { "SPKDAT1L", NULL, "OUT5L" }, + { "SPKDAT1R", NULL, "OUT5R" }, + + { "SPKDAT2L", NULL, "OUT6L" }, + { "SPKDAT2R", NULL, "OUT6R" }, +}; + +static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, + unsigned int Fref, unsigned int Fout) +{ + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + + switch (fll_id) { + case WM5110_FLL1: + return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout); + case WM5110_FLL2: + return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout); + default: + return -EINVAL; + } +} + +#define WM5110_RATES SNDRV_PCM_RATE_8000_192000 + +#define WM5110_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver wm5110_dai[] = { + { + .name = "wm5110-aif1", + .id = 1, + .base = ARIZONA_AIF1_BCLK_CTRL, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 8, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif2", + .id = 2, + .base = ARIZONA_AIF2_BCLK_CTRL, + .playback = { + .stream_name = "AIF2 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, + { + .name = "wm5110-aif3", + .id = 3, + .base = ARIZONA_AIF3_BCLK_CTRL, + .playback = { + .stream_name = "AIF3 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .capture = { + .stream_name = "AIF3 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = WM5110_RATES, + .formats = WM5110_FORMATS, + }, + .ops = &arizona_dai_ops, + .symmetric_rates = 1, + }, +}; + +static int wm5110_codec_probe(struct snd_soc_codec *codec) +{ + struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); + + codec->control_data = priv->core.arizona->regmap; + return snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); +} + +#define WM5110_DIG_VU 0x0200 + +static unsigned int wm5110_digital_vu[] = { + ARIZONA_ADC_DIGITAL_VOLUME_1L, + ARIZONA_ADC_DIGITAL_VOLUME_1R, + ARIZONA_ADC_DIGITAL_VOLUME_2L, + ARIZONA_ADC_DIGITAL_VOLUME_2R, + ARIZONA_ADC_DIGITAL_VOLUME_3L, + ARIZONA_ADC_DIGITAL_VOLUME_3R, + + ARIZONA_DAC_DIGITAL_VOLUME_1L, + ARIZONA_DAC_DIGITAL_VOLUME_1R, + ARIZONA_DAC_DIGITAL_VOLUME_2L, + ARIZONA_DAC_DIGITAL_VOLUME_2R, + ARIZONA_DAC_DIGITAL_VOLUME_3L, + ARIZONA_DAC_DIGITAL_VOLUME_3R, + ARIZONA_DAC_DIGITAL_VOLUME_4L, + ARIZONA_DAC_DIGITAL_VOLUME_4R, + ARIZONA_DAC_DIGITAL_VOLUME_5L, + ARIZONA_DAC_DIGITAL_VOLUME_5R, +}; + +static struct snd_soc_codec_driver soc_codec_dev_wm5110 = { + .probe = wm5110_codec_probe, + + .idle_bias_off = true, + + .set_sysclk = arizona_set_sysclk, + .set_pll = wm5110_set_fll, + + .controls = wm5110_snd_controls, + .num_controls = ARRAY_SIZE(wm5110_snd_controls), + .dapm_widgets = wm5110_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(wm5110_dapm_widgets), + .dapm_routes = wm5110_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(wm5110_dapm_routes), +}; + +static int __devinit wm5110_probe(struct platform_device *pdev) +{ + struct arizona *arizona = dev_get_drvdata(pdev->dev.parent); + struct wm5110_priv *wm5110; + int i; + + wm5110 = devm_kzalloc(&pdev->dev, sizeof(struct wm5110_priv), + GFP_KERNEL); + if (wm5110 == NULL) + return -ENOMEM; + platform_set_drvdata(pdev, wm5110); + + wm5110->core.arizona = arizona; + + for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) + wm5110->fll[i].vco_mult = 3; + + arizona_init_fll(arizona, 1, ARIZONA_FLL1_CONTROL_1 - 1, + ARIZONA_IRQ_FLL1_LOCK, ARIZONA_IRQ_FLL1_CLOCK_OK, + &wm5110->fll[0]); + arizona_init_fll(arizona, 2, ARIZONA_FLL2_CONTROL_1 - 1, + ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, + &wm5110->fll[1]); + + for (i = 0; i < ARRAY_SIZE(wm5110_dai); i++) + arizona_init_dai(&wm5110->core, i); + + /* Latch volume update bits */ + for (i = 0; i < ARRAY_SIZE(wm5110_digital_vu); i++) + regmap_update_bits(arizona->regmap, wm5110_digital_vu[i], + WM5110_DIG_VU, WM5110_DIG_VU); + + pm_runtime_enable(&pdev->dev); + pm_runtime_idle(&pdev->dev); + + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wm5110, + wm5110_dai, ARRAY_SIZE(wm5110_dai)); +} + +static int __devexit wm5110_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + pm_runtime_disable(&pdev->dev); + + return 0; +} + +static struct platform_driver wm5110_codec_driver = { + .driver = { + .name = "wm5110-codec", + .owner = THIS_MODULE, + }, + .probe = wm5110_probe, + .remove = __devexit_p(wm5110_remove), +}; + +module_platform_driver(wm5110_codec_driver); + +MODULE_DESCRIPTION("ASoC WM5110 driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:wm5110-codec"); diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h new file mode 100644 index 0000000..75e9351 --- /dev/null +++ b/sound/soc/codecs/wm5110.h @@ -0,0 +1,21 @@ +/* + * wm5110.h -- WM5110 ALSA SoC Audio driver + * + * Copyright 2012 Wolfson Microelectronics plc + * + * Author: Mark Brown + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _WM5110_H +#define _WM5110_H + +#include "arizona.h" + +#define WM5110_FLL1 1 +#define WM5110_FLL2 2 + +#endif -- cgit v0.10.2 From b761c0ca2e964a240d74e50da9e27dc0b3be0649 Mon Sep 17 00:00:00 2001 From: Matthias Kaehlcke Date: Wed, 11 Jul 2012 17:36:34 +0200 Subject: ASoC: Free memory in the error paths of soc_of_parse_audio_routing() Release the memory of the routing table before leaving the function upon errors in the device tree Signed-off-by: Matthias Kaehlcke Signed-off-by: Mark Brown diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 64b464c..f219b2f 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4136,6 +4136,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, 2 * i, ret); + kfree(routes); return -EINVAL; } ret = of_property_read_string_index(np, propname, @@ -4144,6 +4145,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, dev_err(card->dev, "Property '%s' index %d could not be read: %d\n", propname, (2 * i) + 1, ret); + kfree(routes); return -EINVAL; } } -- cgit v0.10.2 From e4dd76788c7e5b27165890d712c8c4f6f0abd645 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jul 2012 19:03:48 +0100 Subject: ASoC: wm8962: Redo early init of the part on resume Ensure robust startup of the part by going through the reset procedure prior to resyncing the full register cache, avoiding potential intermittent faults in some designs. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index beb709b..eaf6586 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3725,6 +3725,9 @@ static int wm8962_runtime_resume(struct device *dev) } regcache_cache_only(wm8962->regmap, false); + + wm8962_reset(wm8962); + regcache_sync(wm8962->regmap); regmap_update_bits(wm8962->regmap, WM8962_ANTI_POP, -- cgit v0.10.2 From 98b3cf1290d2d6fbc926dc410d3713c8244e7604 Mon Sep 17 00:00:00 2001 From: Marek Belisko Date: Thu, 12 Jul 2012 23:00:16 +0200 Subject: ASoC: dapm: Fix compilation warning MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit Fix following: sound/soc/soc-dapm.c: In function ‘dapm_clock_event’: sound/soc/soc-dapm.c:1021:1: warning: control reaches end of non-void function [-Wreturn-type] Signed-off-by: Marek Belisko Signed-off-by: Mark Brown diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4ba47aa..f7a13f7 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1019,6 +1019,7 @@ int dapm_clock_event(struct snd_soc_dapm_widget *w, return 0; } #endif + return 0; } EXPORT_SYMBOL_GPL(dapm_clock_event); -- cgit v0.10.2 From 0eed8a18696af4e6cf0315f935a730521b54725e Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 13 Jul 2012 19:22:44 +0200 Subject: ASoC: Convert S3C24XX I2S driver to gpiolib API The s3c2410_gpio* calls are obsolete and have been scheduled for removal since several kernel releases. Remove them and use common gpiolib API. This patch is a prerequisite for removal of the obsolete S3C24XX SoC GPIO definitions. Tested on Micro2440-SDK. Cc: Ben Dooks Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index c4aa4d4..0aae3a3 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -23,7 +23,6 @@ #include #include -#include #include #include @@ -391,12 +390,9 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) } clk_enable(s3c24xx_i2s.iis_clk); - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); writel(S3C2410_IISCON_IISEN, s3c24xx_i2s.regs + S3C2410_IISCON); -- cgit v0.10.2 From 601787c232306e0bb84fff9fc7c2be5a5c7b87a0 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Fri, 13 Jul 2012 19:22:45 +0200 Subject: ASoC: Convert S3C2412 I2S driver to gpiolib API The s3c2410_gpio* calls are obsolete and have been scheduled for removal since several kernel releases. Remove them and use common gpiolib API. This patch is a prerequisite for removal of the obsolete S3C24XX SoC GPIO definitions. Compile tested only. Cc: Ben Dooks Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index 79fbeea..ac7701b 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -25,7 +25,6 @@ #include #include -#include #include #include "dma.h" @@ -83,12 +82,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.iis_cclk = s3c2412_i2s.iis_pclk; - /* Configure the I2S pins in correct mode */ - s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); - s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); - s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); - s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + /* Configure the I2S pins (GPE0...GPE4) in correct mode */ + s3c_gpio_cfgall_range(S3C2410_GPE(0), 5, S3C_GPIO_SFN(2), + S3C_GPIO_PULL_NONE); return 0; } -- cgit v0.10.2 From b4046d013b5b9a7cab835def403f7f421cdf0cb6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jul 2012 19:11:30 +0100 Subject: ASoC: wm8994: Update micdet for irqdomain conversion The conversion of the core driver to irqdomains means that we don't need and irq_base to have working interrupts so use wm8994_request_irq() to deal with looking up the interrupt number for the micdet IRQ. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 7bb8752..6576338 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3695,9 +3695,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; - else if (wm8994->pdata && wm8994->pdata->irq_base) - wm8994->micdet_irq = wm8994->pdata->irq_base + - WM8994_IRQ_MIC1_DET; pm_runtime_enable(codec->dev); pm_runtime_idle(codec->dev); @@ -3836,6 +3833,10 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) dev_warn(codec->dev, "Failed to request Mic detect IRQ: %d\n", ret); + } else { + wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_MIC1_DET, + wm8958_mic_irq, "Mic detect", + wm8994); } } -- cgit v0.10.2 From 31a2239a5a77c48b12c54210aa250ce76c8f9535 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 18 Jul 2012 19:16:06 +0100 Subject: ASoC: littlemill: Add userspace control of the WM1250 I/O Signed-off-by: Mark Brown diff --git a/sound/soc/samsung/littlemill.c b/sound/soc/samsung/littlemill.c index c82c646..ee52c8a0 100644 --- a/sound/soc/samsung/littlemill.c +++ b/sound/soc/samsung/littlemill.c @@ -211,6 +211,11 @@ static int bbclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const struct snd_kcontrol_new controls[] = { + SOC_DAPM_PIN_SWITCH("WM1250 Input"), + SOC_DAPM_PIN_SWITCH("WM1250 Output"), +}; + static struct snd_soc_dapm_widget widgets[] = { SND_SOC_DAPM_HP("Headphone", NULL), @@ -282,6 +287,8 @@ static struct snd_soc_card littlemill = { .set_bias_level = littlemill_set_bias_level, .set_bias_level_post = littlemill_set_bias_level_post, + .controls = controls, + .num_controls = ARRAY_SIZE(controls), .dapm_widgets = widgets, .num_dapm_widgets = ARRAY_SIZE(widgets), .dapm_routes = audio_paths, -- cgit v0.10.2 From 409b78cc17a4a3d07a541037575da648ced99437 Mon Sep 17 00:00:00 2001 From: Torben Hohn Date: Wed, 18 Jul 2012 15:01:17 +0200 Subject: ASoC imx-audmux: add MX31_AUDMUX_PORT7_SSI_PINS_7 define The MX31 Audmux also has 7 Ports. This patch adds the missing define, and makes the debugfs code iterate over that port too. Signed-off-by: Torben Hohn Signed-off-by: Mark Brown diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 0803274..e7c800e 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -156,7 +156,7 @@ static void __init audmux_debugfs_init(void) return; } - for (i = 0; i < MX31_AUDMUX_PORT6_SSI_PINS_6 + 1; i++) { + for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { snprintf(buf, sizeof(buf), "ssi%d", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) diff --git a/sound/soc/fsl/imx-audmux.h b/sound/soc/fsl/imx-audmux.h index 04ebbab..b8ff44b 100644 --- a/sound/soc/fsl/imx-audmux.h +++ b/sound/soc/fsl/imx-audmux.h @@ -14,6 +14,7 @@ #define MX31_AUDMUX_PORT4_SSI_PINS_4 3 #define MX31_AUDMUX_PORT5_SSI_PINS_5 4 #define MX31_AUDMUX_PORT6_SSI_PINS_6 5 +#define MX31_AUDMUX_PORT7_SSI_PINS_7 6 #define MX51_AUDMUX_PORT1_SSI0 0 #define MX51_AUDMUX_PORT2_SSI1 1 -- cgit v0.10.2