diff options
Diffstat (limited to 'sound')
| -rw-r--r-- | sound/soc/atmel/Kconfig | 9 | ||||
| -rw-r--r-- | sound/soc/atmel/Makefile | 1 | ||||
| -rw-r--r-- | sound/soc/atmel/atmel_ssc_dai.c | 5 | ||||
| -rw-r--r-- | sound/soc/atmel/snd-soc-afeb9260.c | 151 | ||||
| -rw-r--r-- | sound/soc/codecs/ak4671.c | 13 | ||||
| -rw-r--r-- | sound/soc/codecs/alc5623.c | 22 | ||||
| -rw-r--r-- | sound/soc/codecs/alc5632.c | 22 | ||||
| -rw-r--r-- | sound/soc/codecs/arizona.c | 30 |
8 files changed, 33 insertions, 220 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig index 27e3fc4..fb38783 100644 --- a/sound/soc/atmel/Kconfig +++ b/sound/soc/atmel/Kconfig @@ -52,12 +52,3 @@ config SND_AT91_SOC_SAM9X5_WM8731 help Say Y if you want to add support for audio SoC on an at91sam9x5 based board that is using WM8731 codec. - -config SND_AT91_SOC_AFEB9260 - tristate "SoC Audio support for AFEB9260 board" - depends on ARCH_AT91 && ATMEL_SSC && ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC - select SND_ATMEL_SOC_PDC - select SND_ATMEL_SOC_SSC - select SND_SOC_TLV320AIC23_I2C - help - Say Y here to support sound on AFEB9260 board. diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile index 5baabc8..466a821 100644 --- a/sound/soc/atmel/Makefile +++ b/sound/soc/atmel/Makefile @@ -17,4 +17,3 @@ snd-soc-sam9x5-wm8731-objs := sam9x5_wm8731.o obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o obj-$(CONFIG_SND_ATMEL_SOC_WM8904) += snd-atmel-soc-wm8904.o obj-$(CONFIG_SND_AT91_SOC_SAM9X5_WM8731) += snd-soc-sam9x5-wm8731.o -obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index f403f39..b1cc2a4 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -310,7 +310,10 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, * transmit and receive, so if a value has already * been set, it must match this value. */ - if (ssc_p->cmr_div == 0) + if (ssc_p->dir_mask != + (SSC_DIR_MASK_PLAYBACK | SSC_DIR_MASK_CAPTURE)) + ssc_p->cmr_div = div; + else if (ssc_p->cmr_div == 0) ssc_p->cmr_div = div; else if (div != ssc_p->cmr_div) diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c deleted file mode 100644 index 9579799..0000000 --- a/sound/soc/atmel/snd-soc-afeb9260.c +++ /dev/null @@ -1,151 +0,0 @@ -/* - * afeb9260.c -- SoC audio for AFEB9260 - * - * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.org> - * - * 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., 51 Franklin St, Fifth Floor, Boston, MA - * 02110-1301 USA - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/kernel.h> -#include <linux/clk.h> -#include <linux/platform_device.h> - -#include <linux/atmel-ssc.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/mach-types.h> -#include <mach/hardware.h> -#include <linux/gpio.h> - -#include "../codecs/tlv320aic23.h" -#include "atmel-pcm.h" -#include "atmel_ssc_dai.h" - -#define CODEC_CLOCK 12000000 - -static int afeb9260_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; - int err; - - /* Set the codec system clock for DAC and ADC */ - err = - snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN); - - if (err < 0) { - printk(KERN_ERR "can't set codec system clock\n"); - return err; - } - - return err; -} - -static struct snd_soc_ops afeb9260_ops = { - .hw_params = afeb9260_hw_params, -}; - -static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = { - SND_SOC_DAPM_HP("Headphone Jack", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), - SND_SOC_DAPM_MIC("Mic Jack", NULL), -}; - -static const struct snd_soc_dapm_route afeb9260_audio_map[] = { - {"Headphone Jack", NULL, "LHPOUT"}, - {"Headphone Jack", NULL, "RHPOUT"}, - - {"LLINEIN", NULL, "Line In"}, - {"RLINEIN", NULL, "Line In"}, - - {"MICIN", NULL, "Mic Jack"}, -}; - - -/* Digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link afeb9260_dai = { - .name = "TLV320AIC23", - .stream_name = "AIC23", - .cpu_dai_name = "atmel-ssc-dai.0", - .codec_dai_name = "tlv320aic23-hifi", - .platform_name = "atmel_pcm-audio", - .codec_name = "tlv320aic23-codec.0-001a", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_IF | - SND_SOC_DAIFMT_CBM_CFM, - .ops = &afeb9260_ops, -}; - -/* Audio machine driver */ -static struct snd_soc_card snd_soc_machine_afeb9260 = { - .name = "AFEB9260", - .owner = THIS_MODULE, - .dai_link = &afeb9260_dai, - .num_links = 1, - - .dapm_widgets = tlv320aic23_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets), - .dapm_routes = afeb9260_audio_map, - .num_dapm_routes = ARRAY_SIZE(afeb9260_audio_map), -}; - -static struct platform_device *afeb9260_snd_device; - -static int __init afeb9260_soc_init(void) -{ - int err; - struct device *dev; - - if (!(machine_is_afeb9260())) - return -ENODEV; - - - afeb9260_snd_device = platform_device_alloc("soc-audio", -1); - if (!afeb9260_snd_device) { - printk(KERN_ERR "ASoC: Platform device allocation failed\n"); - return -ENOMEM; - } - - platform_set_drvdata(afeb9260_snd_device, &snd_soc_machine_afeb9260); - err = platform_device_add(afeb9260_snd_device); - if (err) - goto err1; - - dev = &afeb9260_snd_device->dev; - - return 0; -err1: - platform_device_put(afeb9260_snd_device); - return err; -} - -static void __exit afeb9260_soc_exit(void) -{ - platform_device_unregister(afeb9260_snd_device); -} - -module_init(afeb9260_soc_init); -module_exit(afeb9260_soc_exit); - -MODULE_AUTHOR("Sergey Lapin <slapin@ossfans.org>"); -MODULE_DESCRIPTION("ALSA SoC for AFEB9260"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 998fa0c..686cacb 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -611,20 +611,7 @@ static struct snd_soc_dai_driver ak4671_dai = { .ops = &ak4671_dai_ops, }; -static int ak4671_probe(struct snd_soc_codec *codec) -{ - return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -} - -static int ak4671_remove(struct snd_soc_codec *codec) -{ - ak4671_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { - .probe = ak4671_probe, - .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, .controls = ak4671_snd_controls, .num_controls = ARRAY_SIZE(ak4671_snd_controls), diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index 9d0755a..bdf8c5a 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -866,7 +866,6 @@ static int alc5623_suspend(struct snd_soc_codec *codec) { struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); regcache_cache_only(alc5623->regmap, true); return 0; @@ -887,15 +886,6 @@ static int alc5623_resume(struct snd_soc_codec *codec) return ret; } - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - /* charge alc5623 caps */ - if (codec->dapm.suspend_bias_level == SND_SOC_BIAS_ON) { - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - codec->dapm.bias_level = SND_SOC_BIAS_ON; - alc5623_set_bias_level(codec, codec->dapm.bias_level); - } - return 0; } @@ -906,9 +896,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) alc5623_reset(codec); - /* power on device */ - alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - if (alc5623->add_ctrl) { snd_soc_write(codec, ALC5623_ADD_CTRL_REG, alc5623->add_ctrl); @@ -964,19 +951,12 @@ static int alc5623_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5623_remove(struct snd_soc_codec *codec) -{ - alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .probe = alc5623_probe, - .remove = alc5623_remove, .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, + .suspend_bias_off = true, }; static const struct regmap_config alc5623_regmap = { diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index 85942ca..d1fdbc2 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1038,23 +1038,15 @@ static struct snd_soc_dai_driver alc5632_dai = { }; #ifdef CONFIG_PM -static int alc5632_suspend(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static int alc5632_resume(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); regcache_sync(alc5632->regmap); - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; } #else -#define alc5632_suspend NULL #define alc5632_resume NULL #endif @@ -1062,9 +1054,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) { struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); - /* power on device */ - alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - switch (alc5632->id) { case 0x5c: snd_soc_add_codec_controls(codec, alc5632_vol_snd_controls, @@ -1077,19 +1066,12 @@ static int alc5632_probe(struct snd_soc_codec *codec) return 0; } -/* power down chip */ -static int alc5632_remove(struct snd_soc_codec *codec) -{ - alc5632_set_bias_level(codec, SND_SOC_BIAS_OFF); - return 0; -} - static struct snd_soc_codec_driver soc_codec_device_alc5632 = { .probe = alc5632_probe, - .remove = alc5632_remove, - .suspend = alc5632_suspend, .resume = alc5632_resume, .set_bias_level = alc5632_set_bias_level, + .suspend_bias_off = true, + .controls = alc5632_snd_controls, .num_controls = ARRAY_SIZE(alc5632_snd_controls), .dapm_widgets = alc5632_dapm_widgets, diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c05e7a..19887bf 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -61,6 +61,11 @@ #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 +#define ARIZONA_FMT_DSP_MODE_A 0 +#define ARIZONA_FMT_DSP_MODE_B 1 +#define ARIZONA_FMT_I2S_MODE 2 +#define ARIZONA_FMT_LEFT_JUSTIFIED_MODE 3 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -648,7 +653,7 @@ SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { - "1.536MHz", "3.072MHz", "6.144MHz", + "1.536MHz", "3.072MHz", "6.144MHz", "768kHz", }; const struct soc_enum arizona_in_dmic_osr[] = { @@ -946,10 +951,26 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_A: - mode = 0; + mode = ARIZONA_FMT_DSP_MODE_A; + break; + case SND_SOC_DAIFMT_DSP_B: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "DSP_B not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_DSP_MODE_B; break; case SND_SOC_DAIFMT_I2S: - mode = 2; + mode = ARIZONA_FMT_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) + != SND_SOC_DAIFMT_CBM_CFM) { + arizona_aif_err(dai, "LEFT_J not valid in slave mode\n"); + return -EINVAL; + } + mode = ARIZONA_FMT_LEFT_JUSTIFIED_MODE; break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", @@ -1298,7 +1319,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, /* Force multiple of 2 channels for I2S mode */ val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); - if ((channels & 1) && (val & ARIZONA_AIF1_FMT_MASK)) { + val &= ARIZONA_AIF1_FMT_MASK; + if ((channels & 1) && (val == ARIZONA_FMT_I2S_MODE)) { arizona_aif_dbg(dai, "Forcing stereo mode\n"); bclk_target /= channels; bclk_target *= channels + 1; |