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-rw-r--r--sound/soc/atmel/Kconfig9
-rw-r--r--sound/soc/atmel/Makefile1
-rw-r--r--sound/soc/atmel/atmel_ssc_dai.c5
-rw-r--r--sound/soc/atmel/snd-soc-afeb9260.c151
-rw-r--r--sound/soc/codecs/ak4671.c13
-rw-r--r--sound/soc/codecs/alc5623.c22
-rw-r--r--sound/soc/codecs/alc5632.c22
-rw-r--r--sound/soc/codecs/arizona.c30
8 files changed, 33 insertions, 220 deletions
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index 27e3fc4a536b..fb3878312bf8 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 5baabc8bde3a..466a821da98c 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 f403f399808a..b1cc2a4a7fc0 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 9579799ace54..000000000000
--- 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 998fa0c5a0b9..686cacb0e835 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 9d0755aa1d16..bdf8c5ac8ca4 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 85942ca36cbf..d1fdbc266631 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 0c05e7a7945f..19887bfffbf9 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;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-30 21:48:42 +0000