Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623',...

Merge remote-tracking branches 'asoc/topic/ak4671', 'asoc/topic/alc5623', 'asoc/topic/alc5632', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next

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.

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

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)

sound/soc/atmel/snd-soc-afeb9260.c

-/*
- * 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");
-

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),

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 = {

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,

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;