From 601722157b3f6be73623644eeae6f14940f0bd8f Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Mon, 4 Jun 2012 10:41:03 +0800 Subject: ARM: MMP: add pxa910-ssp into ssp_id_table add pxa910-ssp into ssp_id_table, and fix pxa-ssp compiling issue under mach-mmp architect. Signed-off-by: Qiao Zhou Acked-by: Haojian Zhuang Signed-off-by: Mark Brown diff --git a/arch/arm/plat-pxa/ssp.c b/arch/arm/plat-pxa/ssp.c index 58b7980..584c9bf 100644 --- a/arch/arm/plat-pxa/ssp.c +++ b/arch/arm/plat-pxa/ssp.c @@ -193,6 +193,7 @@ static const struct platform_device_id ssp_id_table[] = { { "pxa25x-nssp", PXA25x_NSSP }, { "pxa27x-ssp", PXA27x_SSP }, { "pxa168-ssp", PXA168_SSP }, + { "pxa910-ssp", PXA910_SSP }, { }, }; diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index 44835fb..f9fe15e 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -161,6 +161,7 @@ enum pxa_ssp_type { PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ PXA27x_SSP, PXA168_SSP, + PXA910_SSP, CE4100_SSP, }; diff --git a/include/linux/spi/pxa2xx_spi.h b/include/linux/spi/pxa2xx_spi.h index d3e1075..c73d144 100644 --- a/include/linux/spi/pxa2xx_spi.h +++ b/include/linux/spi/pxa2xx_spi.h @@ -43,7 +43,7 @@ struct pxa2xx_spi_chip { void (*cs_control)(u32 command); }; -#ifdef CONFIG_ARCH_PXA +#if defined(CONFIG_ARCH_PXA) || defined(CONFIG_ARCH_MMP) #include #include -- cgit v0.10.2 From 972a55b62d592cfcd6d73577df8a52f1251ea9a7 Mon Sep 17 00:00:00 2001 From: Qiao Zhou Date: Mon, 4 Jun 2012 10:41:04 +0800 Subject: ASoC: fix pxa-ssp compiling issue under mach-mmp pxa-ssp.c uses API like cpu_is_pxa3xx(), cpu_is_pxa2xx(), which is defined under arch-pxa architecture, and drivers under mach-mmp can't find it. so just use ssp->type to replace that API. Signed-off-by: Qiao Zhou Acked-by: Haojian Zhuang Signed-off-by: Mark Brown diff --git a/include/linux/pxa2xx_ssp.h b/include/linux/pxa2xx_ssp.h index f9fe15e..f366320 100644 --- a/include/linux/pxa2xx_ssp.h +++ b/include/linux/pxa2xx_ssp.h @@ -160,6 +160,7 @@ enum pxa_ssp_type { PXA25x_SSP, /* pxa 210, 250, 255, 26x */ PXA25x_NSSP, /* pxa 255, 26x (including ASSP) */ PXA27x_SSP, + PXA3xx_SSP, PXA168_SSP, PXA910_SSP, CE4100_SSP, diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 1c2aa7f..4da5fc5 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -33,7 +33,6 @@ #include #include -#include #include "../../arm/pxa2xx-pcm.h" #include "pxa-ssp.h" @@ -194,7 +193,7 @@ static void pxa_ssp_set_scr(struct ssp_device *ssp, u32 div) { u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) { + if (ssp->type == PXA25x_SSP) { sscr0 &= ~0x0000ff00; sscr0 |= ((div - 2)/2) << 8; /* 2..512 */ } else { @@ -212,7 +211,7 @@ static u32 pxa_ssp_get_scr(struct ssp_device *ssp) u32 sscr0 = pxa_ssp_read_reg(ssp, SSCR0); u32 div; - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) + if (ssp->type == PXA25x_SSP) div = ((sscr0 >> 8) & 0xff) * 2 + 2; else div = ((sscr0 >> 8) & 0xfff) + 1; @@ -242,7 +241,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_PLL: /* Internal PLL is fixed */ - if (cpu_is_pxa25x()) + if (ssp->type == PXA25x_SSP) priv->sysclk = 1843200; else priv->sysclk = 13000000; @@ -266,11 +265,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_disable(ssp->clk); val = pxa_ssp_read_reg(ssp, SSCR0) | sscr0; pxa_ssp_write_reg(ssp, SSCR0, val); - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) clk_enable(ssp->clk); return 0; @@ -294,24 +293,20 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, case PXA_SSP_AUDIO_DIV_SCDB: val = pxa_ssp_read_reg(ssp, SSACD); val &= ~SSACD_SCDB; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val &= ~SSACD_SCDX8; -#endif switch (div) { case PXA_SSP_CLK_SCDB_1: val |= SSACD_SCDB; break; case PXA_SSP_CLK_SCDB_4: break; -#if defined(CONFIG_PXA3xx) case PXA_SSP_CLK_SCDB_8: - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) val |= SSACD_SCDX8; else return -EINVAL; break; -#endif default: return -EINVAL; } @@ -337,10 +332,8 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, struct ssp_device *ssp = priv->ssp; u32 ssacd = pxa_ssp_read_reg(ssp, SSACD) & ~0x70; -#if defined(CONFIG_PXA3xx) - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) pxa_ssp_write_reg(ssp, SSACDD, 0); -#endif switch (freq_out) { case 5622000: @@ -365,11 +358,10 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, break; default: -#ifdef CONFIG_PXA3xx /* PXA3xx has a clock ditherer which can be used to generate * a wider range of frequencies - calculate a value for it. */ - if (cpu_is_pxa3xx()) { + if (ssp->type == PXA3xx_SSP) { u32 val; u64 tmp = 19968; tmp *= 1000000; @@ -386,7 +378,6 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, val, freq_out); break; } -#endif return -EINVAL; } @@ -590,10 +581,8 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -#ifdef CONFIG_PXA3xx - if (cpu_is_pxa3xx()) + if (ssp->type == PXA3xx_SSP) sscr0 |= SSCR0_FPCKE; -#endif sscr0 |= SSCR0_DataSize(16); break; case SNDRV_PCM_FORMAT_S24_LE: @@ -618,9 +607,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, * trying and failing a lot; some of the registers * needed for that mode are only available on PXA3xx. */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) + if (ssp->type != PXA3xx_SSP) return -EINVAL; sspsp |= SSPSP_SFRMWDTH(width * 2); @@ -628,9 +615,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sspsp |= SSPSP_EDMYSTOP(3); sspsp |= SSPSP_DMYSTOP(3); sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif } else { /* The frame width is the width the LRCLK is * asserted for; the delay is expressed in -- cgit v0.10.2 From 433897f7408b556f7dfbb98c94deea02e634d2a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:38:12 +0800 Subject: ASoC: wm8904: Fix GPIO and MICBIAS initialisation for regmap conversion We no longer have a flat ASoC cache so can't peer directly into the array any more but should instead use the register I/O functions to update the cache. Signed-off-by: Mark Brown Cc: stable@vger.kernel.org (v3.4) diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 65d525d..4e190b5 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2084,7 +2084,6 @@ 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; - u16 *reg_cache = codec->reg_cache; int ret, i; codec->cache_sync = 1; @@ -2180,14 +2179,18 @@ static int wm8904_probe(struct snd_soc_codec *codec) if (!pdata->gpio_cfg[i]) continue; - reg_cache[WM8904_GPIO_CONTROL_1 + i] - = pdata->gpio_cfg[i] & 0xffff; + 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++) - reg_cache[WM8904_MIC_BIAS_CONTROL_0 + i] - = pdata->mic_cfg[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 -- cgit v0.10.2 From c1b88ee2bbb82c56ac24c70850004de9a43915d5 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 9 Jun 2012 11:27:17 +0800 Subject: ASoC: wm8904: Fix cache only management We should be using the regmap API consistently for all the cache only configuration and we should be going cache only before we power down the supplies. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 4e190b5..812acd8 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1863,6 +1863,7 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, return ret; } + regcache_cache_only(wm8904->regmap, false); regcache_sync(wm8904->regmap); /* Enable bias */ @@ -1899,14 +1900,8 @@ static int wm8904_set_bias_level(struct snd_soc_codec *codec, snd_soc_update_bits(codec, WM8904_BIAS_CONTROL_0, WM8904_BIAS_ENA, 0); -#ifdef CONFIG_REGULATOR - /* Post 2.6.34 we will be able to get a callback when - * the regulators are disabled which we can use but - * for now just assume that the power will be cut if - * the regulator API is in use. - */ - codec->cache_sync = 1; -#endif + regcache_cache_only(wm8904->regmap, true); + regcache_mark_dirty(wm8904->regmap); regulator_bulk_disable(ARRAY_SIZE(wm8904->supplies), wm8904->supplies); @@ -2086,7 +2081,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) struct wm8904_pdata *pdata = wm8904->pdata; int ret, i; - codec->cache_sync = 1; codec->control_data = wm8904->regmap; switch (wm8904->devtype) { @@ -2149,6 +2143,7 @@ static int wm8904_probe(struct snd_soc_codec *codec) goto err_enable; } + regcache_cache_only(wm8904->regmap, true); /* 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); -- cgit v0.10.2 From 90ba6859ce914feff3632eeb7227b9f99c030ca9 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:15:50 +0800 Subject: ASoC: wm8996: Remove spurious regulator_bulk_free() We're using demv_regulator_bulk_get() so don't need to manually free and this is in the CODEC driver not the I2C driver anyway. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 8af422e..379bd1e 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3051,7 +3051,6 @@ static int wm8996_remove(struct snd_soc_codec *codec) for (i = 0; i < ARRAY_SIZE(wm8996->supplies); i++) regulator_unregister_notifier(wm8996->supplies[i].consumer, &wm8996->disable_nb[i]); - regulator_bulk_free(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); return 0; } -- cgit v0.10.2 From db1334098392e1278a113e629a7ae58a7453f83f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:23:13 +0800 Subject: ASoC: wm8996: Move reset before the initial regulator disable If we don't have control over the LDO but do have control over the other regulators then we may end up trying to write to a powered off device. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 379bd1e..64d9cf7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -3205,14 +3205,14 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "revision %c\n", (reg & WM8996_CHIP_REV_MASK) + 'A'); - regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); - ret = wm8996_reset(wm8996); if (ret < 0) { dev_err(&i2c->dev, "Failed to issue reset\n"); goto err_regmap; } + regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); + wm8996_init_gpio(wm8996); ret = snd_soc_register_codec(&i2c->dev, -- cgit v0.10.2 From 9c699e0a9566cfb6245284ce4857d7a06b02f3b1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 11 Jun 2012 18:20:02 +0800 Subject: ASoC: wm8996: Mark the CODEC as cache only when powering off on boot Otherwise we might try to write to a powered off device. Signed-off-by: Mark Brown diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 64d9cf7..dc9b42b 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2837,8 +2837,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) } } - regcache_cache_only(codec->control_data, true); - /* Apply platform data settings */ snd_soc_update_bits(codec, WM8996_LINE_INPUT_CONTROL, WM8996_INL_MODE_MASK | WM8996_INR_MODE_MASK, @@ -3211,6 +3209,7 @@ static __devinit int wm8996_i2c_probe(struct i2c_client *i2c, 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 3419ae781f1592b3d367107db6500090495490cd Mon Sep 17 00:00:00 2001 From: Stephen Warren Date: Mon, 11 Jun 2012 11:27:19 -0600 Subject: ASoC: tegra+wm8903: turn of mic detect when card is removed If mic detect is left enabled and the WM8903 detects a status change, the WM8903 driver will make a callback against the free()d jack, which will cause a crash. This problem can be triggered by fully initializing an audio card, then removing and re-inserting the machine driver module. 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..3b6da91 100644 --- a/sound/soc/tegra/tegra_wm8903.c +++ b/sound/soc/tegra/tegra_wm8903.c @@ -346,6 +346,17 @@ static int tegra_wm8903_init(struct snd_soc_pcm_runtime *rtd) return 0; } +static int tegra_wm8903_remove(struct snd_soc_card *card) +{ + struct snd_soc_pcm_runtime *rtd = &(card->rtd[0]); + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_codec *codec = codec_dai->codec; + + wm8903_mic_detect(codec, NULL, 0, 0); + + return 0; +} + static struct snd_soc_dai_link tegra_wm8903_dai = { .name = "WM8903", .stream_name = "WM8903 PCM", @@ -363,6 +374,8 @@ static struct snd_soc_card snd_soc_tegra_wm8903 = { .dai_link = &tegra_wm8903_dai, .num_links = 1, + .remove = tegra_wm8903_remove, + .controls = tegra_wm8903_controls, .num_controls = ARRAY_SIZE(tegra_wm8903_controls), .dapm_widgets = tegra_wm8903_dapm_widgets, -- cgit v0.10.2 From 7fb75db139965c1955bf6bbde62a357f9843286d Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sun, 17 Jun 2012 13:44:27 +0200 Subject: ALSA: snd-usb: fix sync pipe check Fix a bogus sanity check for sync pipe in pcm.c. This flaw was introduced during the streaming logic refactorization. While at it, improve the error messages that are generated in such cases. Signed-off-by: Daniel Mack Reported-and-tested-by: Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index cdf8b76..fc7ce7c 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,17 +354,21 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) (get_endpoint(alts, 1)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && get_endpoint(alts, 1)->bSynchAddress != 0 && !implicit_fb)) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. bmAttributes %02x, bLength %d, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + get_endpoint(alts, 1)->bmAttributes, + get_endpoint(alts, 1)->bLength, + get_endpoint(alts, 1)->bSynchAddress); return -EINVAL; } ep = get_endpoint(alts, 1)->bEndpointAddress; - if (get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && + if (!implicit_fb && + get_endpoint(alts, 0)->bLength >= USB_DT_ENDPOINT_AUDIO_SIZE && (( is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress | USB_DIR_IN)) || - (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)) || - ( is_playback && !implicit_fb))) { - snd_printk(KERN_ERR "%d:%d:%d : invalid synch pipe\n", - dev->devnum, fmt->iface, fmt->altsetting); + (!is_playback && ep != (unsigned int)(get_endpoint(alts, 0)->bSynchAddress & ~USB_DIR_IN)))) { + snd_printk(KERN_ERR "%d:%d:%d : invalid sync pipe. is_playback %d, ep %02x, bSynchAddress %02x\n", + dev->devnum, fmt->iface, fmt->altsetting, + is_playback, ep, get_endpoint(alts, 0)->bSynchAddress); return -EINVAL; } -- cgit v0.10.2 From afe25967ecf66b38d94d374f0fcb5f4add458a4c Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 16 Jun 2012 16:58:04 +0200 Subject: ALSA: snd-usb: make snd_usb_substream_capture_trigger static Signed-off-by: Daniel Mack Signed-off-by: Takashi Iwai diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index fc7ce7c..54607f8 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -1151,7 +1151,8 @@ static int snd_usb_substream_playback_trigger(struct snd_pcm_substream *substrea return -EINVAL; } -int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, int cmd) +static int snd_usb_substream_capture_trigger(struct snd_pcm_substream *substream, + int cmd) { int err; struct snd_usb_substream *subs = substream->runtime->private_data; -- cgit v0.10.2 From b4a91cf05c33d4ab5b2b3738a257a3fe49b462bd Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Fri, 15 Jun 2012 19:36:23 -0700 Subject: ALSA: hda - Handle open while transitioning to D3. This addresses an issue encountered when a pcm is opened while transitioning to low power state (codec->power_on == 1 && codec->power_transition == -1). Add snd_pcm_power_up_d3wait to hda_codec. This function is used to power up from azx_open as opposed to snd_hda_power_up used from codec_exec_verb. When powering up from azx_open, wait for pending power downs to complete, avoiding the power up continuing in parallel with the power down on the work queue. The specific issue seen was with the CS4210 codec, it powers off the ADC and DAC nid in its suspend handler. If it is re-opened before the ~100ms power down process completes, the ADC and DAC nid are initialized while powered down and audio is lost until another suspend/resume cycle. Signed-off-by: Dylan Reid Signed-off-by: Takashi Iwai diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 41ca803..7504e62 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4393,20 +4393,19 @@ void snd_hda_update_power_acct(struct hda_codec *codec) codec->power_jiffies += delta; } -/** - * snd_hda_power_up - Power-up the codec - * @codec: HD-audio codec - * - * Increment the power-up counter and power up the hardware really when - * not turned on yet. - */ -void snd_hda_power_up(struct hda_codec *codec) +/* Transition to powered up, if wait_power_down then wait for a pending + * transition to D3 to complete. A pending D3 transition is indicated + * with power_transition == -1. */ +static void __snd_hda_power_up(struct hda_codec *codec, bool wait_power_down) { struct hda_bus *bus = codec->bus; spin_lock(&codec->power_lock); codec->power_count++; - if (codec->power_on || codec->power_transition > 0) { + /* Return if power_on or transitioning to power_on, unless currently + * powering down. */ + if ((codec->power_on || codec->power_transition > 0) && + !(wait_power_down && codec->power_transition < 0)) { spin_unlock(&codec->power_lock); return; } @@ -4430,8 +4429,37 @@ void snd_hda_power_up(struct hda_codec *codec) codec->power_transition = 0; spin_unlock(&codec->power_lock); } + +/** + * snd_hda_power_up - Power-up the codec + * @codec: HD-audio codec + * + * Increment the power-up counter and power up the hardware really when + * not turned on yet. + */ +void snd_hda_power_up(struct hda_codec *codec) +{ + __snd_hda_power_up(codec, false); +} EXPORT_SYMBOL_HDA(snd_hda_power_up); +/** + * snd_hda_power_up_d3wait - Power-up the codec after waiting for any pending + * D3 transition to complete. This differs from snd_hda_power_up() when + * power_transition == -1. snd_hda_power_up sees this case as a nop, + * snd_hda_power_up_d3wait waits for the D3 transition to complete then powers + * back up. + * @codec: HD-audio codec + * + * Cancel any power down operation hapenning on the work queue, then power up. + */ +void snd_hda_power_up_d3wait(struct hda_codec *codec) +{ + /* This will cancel and wait for pending power_work to complete. */ + __snd_hda_power_up(codec, true); +} +EXPORT_SYMBOL_HDA(snd_hda_power_up_d3wait); + #define power_save(codec) \ ((codec)->bus->power_save ? *(codec)->bus->power_save : 0) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 4fc3960..2fdaadb 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1056,10 +1056,12 @@ const char *snd_hda_get_jack_location(u32 cfg); */ #ifdef CONFIG_SND_HDA_POWER_SAVE void snd_hda_power_up(struct hda_codec *codec); +void snd_hda_power_up_d3wait(struct hda_codec *codec); void snd_hda_power_down(struct hda_codec *codec); void snd_hda_update_power_acct(struct hda_codec *codec); #else static inline void snd_hda_power_up(struct hda_codec *codec) {} +static inline void snd_hda_power_up_d3wait(struct hda_codec *codec) {} static inline void snd_hda_power_down(struct hda_codec *codec) {} #endif diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0276382..7757536 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -1766,7 +1766,7 @@ static int azx_pcm_open(struct snd_pcm_substream *substream) buff_step); snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, buff_step); - snd_hda_power_up(apcm->codec); + snd_hda_power_up_d3wait(apcm->codec); err = hinfo->ops.open(hinfo, apcm->codec, substream); if (err < 0) { azx_release_device(azx_dev); -- cgit v0.10.2 From 0b1d8e09089b69ac2e8be203fb28cd07cfe035b2 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Sat, 16 Jun 2012 16:58:36 +0200 Subject: ALSA: 6fire: use NULL instead of 0 for pointer assignment Signed-off-by: Daniel Mack Cc: Torsten Schenk Signed-off-by: Takashi Iwai diff --git a/sound/usb/6fire/firmware.c b/sound/usb/6fire/firmware.c index 6f9715a..56ad923 100644 --- a/sound/usb/6fire/firmware.c +++ b/sound/usb/6fire/firmware.c @@ -209,7 +209,7 @@ static int usb6fire_fw_ezusb_upload( int ret; u8 data; struct usb_device *device = interface_to_usbdev(intf); - const struct firmware *fw = 0; + const struct firmware *fw = NULL; struct ihex_record *rec = kmalloc(sizeof(struct ihex_record), GFP_KERNEL); -- cgit v0.10.2