Merge branch 'for-2.6.31' of...

Merge branch 'for-2.6.31' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc

sound/arm/pxa2xx-ac97-lib.c

@@ -65,7 +65,7 @@ static void set_resetgpio_mode(int resetgpio_action)
 		switch (resetgpio_action) {
 		case RESETGPIO_NORMAL_ALTFUNC:
 			if (reset_gpio == 113)
-				mode = 113 | GPIO_OUT | GPIO_DFLT_LOW;
+				mode = 113 | GPIO_ALT_FN_2_OUT;
 			if (reset_gpio == 95)
 				mode = 95 | GPIO_ALT_FN_1_OUT;
 			break;

sound/soc/atmel/Kconfig

@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
           and FRAME signals on the PlayPaq.  Unless you want to play
           with the AT32 as the SSC master, you probably want to say N here,
           as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+	tristate "SoC Audio support for AFEB9260 board"
+	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y here to support sound on AFEB9260 board.

sound/soc/atmel/Makefile

@@ -13,3 +13,4 @@ snd-soc-playpaq-objs := playpaq_wm8510.o
 
 obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
 obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o

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 <sound/soc-dapm.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->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S|
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return err;
+	}
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return 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 audio_map[] = {
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add afeb9260 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up afeb9260 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = afeb9260_tlv320aic23_init,
+	.ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+	.name = "AFEB9260",
+	.platform = &atmel_soc_platform,
+	.dai_link = &afeb9260_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+	.card = &snd_soc_machine_afeb9260,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+	int err;
+	struct device *dev;
+	struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+
+	if (!(machine_is_afeb9260()))
+		return -ENODEV;
+
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		printk(KERN_ERR "ASoC: Failed to request SSC 0\n");
+		err = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+	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, &afeb9260_snd_devdata);
+	afeb9260_snd_devdata.dev = &afeb9260_snd_device->dev;
+	err = platform_device_add(afeb9260_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &afeb9260_snd_device->dev;
+
+	return 0;
+err1:
+	platform_device_del(afeb9260_snd_device);
+	platform_device_put(afeb9260_snd_device);
+err_ssc:
+	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/twl4030.c

@@ -422,36 +422,18 @@ static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control =
 SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum);
 
 /* Left analog microphone selection */
-static const char *twl4030_analoglmic_texts[] =
-		{"Off", "Main mic", "Headset mic", "AUXL", "Carkit mic"};
-
-static const unsigned int twl4030_analoglmic_values[] =
-		{0x0, 0x1, 0x2, 0x4, 0x8};
-
-static const struct soc_enum twl4030_analoglmic_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICL, 0, 0xf,
-			ARRAY_SIZE(twl4030_analoglmic_texts),
-			twl4030_analoglmic_texts,
-			twl4030_analoglmic_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_analoglmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analoglmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_analoglmic_controls[] = {
+	SOC_DAPM_SINGLE("Main mic", TWL4030_REG_ANAMICL, 0, 1, 0),
+	SOC_DAPM_SINGLE("Headset mic", TWL4030_REG_ANAMICL, 1, 1, 0),
+	SOC_DAPM_SINGLE("AUXL", TWL4030_REG_ANAMICL, 2, 1, 0),
+	SOC_DAPM_SINGLE("Carkit mic", TWL4030_REG_ANAMICL, 3, 1, 0),
+};
 
 /* Right analog microphone selection */
-static const char *twl4030_analogrmic_texts[] =
-		{"Off", "Sub mic", "AUXR"};
-
-static const unsigned int twl4030_analogrmic_values[] =
-		{0x0, 0x1, 0x4};
-
-static const struct soc_enum twl4030_analogrmic_enum =
-	SOC_VALUE_ENUM_SINGLE(TWL4030_REG_ANAMICR, 0, 0x5,
-			ARRAY_SIZE(twl4030_analogrmic_texts),
-			twl4030_analogrmic_texts,
-			twl4030_analogrmic_values);
-
-static const struct snd_kcontrol_new twl4030_dapm_analogrmic_control =
-SOC_DAPM_VALUE_ENUM("Route", twl4030_analogrmic_enum);
+static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = {
+	SOC_DAPM_SINGLE("Sub mic", TWL4030_REG_ANAMICR, 0, 1, 0),
+	SOC_DAPM_SINGLE("AUXR", TWL4030_REG_ANAMICR, 1, 1, 0),
+};
 
 /* TX1 L/R Analog/Digital microphone selection */
 static const char *twl4030_micpathtx1_texts[] =
@@ -1138,11 +1120,15 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = {
 		SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD|
 		SND_SOC_DAPM_POST_REG),
 
-	/* Analog input muxes with switch for the capture amplifiers */
-	SND_SOC_DAPM_VALUE_MUX("Analog Left Capture Route",
-		TWL4030_REG_ANAMICL, 4, 0, &twl4030_dapm_analoglmic_control),
-	SND_SOC_DAPM_VALUE_MUX("Analog Right Capture Route",
-		TWL4030_REG_ANAMICR, 4, 0, &twl4030_dapm_analogrmic_control),
+	/* Analog input mixers for the capture amplifiers */
+	SND_SOC_DAPM_MIXER("Analog Left Capture Route",
+		TWL4030_REG_ANAMICL, 4, 0,
+		&twl4030_dapm_analoglmic_controls[0],
+		ARRAY_SIZE(twl4030_dapm_analoglmic_controls)),
+	SND_SOC_DAPM_MIXER("Analog Right Capture Route",
+		TWL4030_REG_ANAMICR, 4, 0,
+		&twl4030_dapm_analogrmic_controls[0],
+		ARRAY_SIZE(twl4030_dapm_analogrmic_controls)),
 
 	SND_SOC_DAPM_PGA("ADC Physical Left",
 		TWL4030_REG_AVADC_CTL, 3, 0, NULL, 0),

sound/soc/pxa/Kconfig

@@ -89,13 +89,13 @@ config SND_PXA2XX_SOC_E800
 	  Toshiba e800 PDA
 
 config SND_PXA2XX_SOC_EM_X270
-	tristate "SoC Audio support for CompuLab EM-x270"
+	tristate "SoC Audio support for CompuLab EM-x270, eXeda and CM-X300"
 	depends on SND_PXA2XX_SOC && MACH_EM_X270
 	select SND_PXA2XX_SOC_AC97
 	select SND_SOC_WM9712
 	help
 	  Say Y if you want to add support for SoC audio on
-	  CompuLab EM-x270.
+	  CompuLab EM-x270, eXeda and CM-X300 machines.
 
 config SND_PXA2XX_SOC_PALM27X
 	bool "SoC Audio support for Palm T|X, T5 and LifeDrive"

sound/soc/pxa/em-x270.c

 /*
- * em-x270.c  --  SoC audio for EM-X270
+ * SoC audio driver for EM-X270, eXeda and CM-X300
  *
- * Copyright 2007 CompuLab, Ltd.
+ * Copyright 2007, 2009 CompuLab, Ltd.
  *
  * Author: Mike Rapoport <mike@compulab.co.il>
  *
@@ -68,7 +68,8 @@ static int __init em_x270_init(void)
 {
 	int ret;
 
-	if (!machine_is_em_x270())
+	if (!(machine_is_em_x270() || machine_is_exeda()
+	      || machine_is_cm_x300()))
 		return -ENODEV;
 
 	em_x270_snd_device = platform_device_alloc("soc-audio", -1);
@@ -95,5 +96,5 @@ module_exit(em_x270_exit);
 
 /* Module information */
 MODULE_AUTHOR("Mike Rapoport");
-MODULE_DESCRIPTION("ALSA SoC EM-X270");
+MODULE_DESCRIPTION("ALSA SoC EM-X270, eXeda and CM-X300");
 MODULE_LICENSE("GPL");

sound/soc/pxa/pxa2xx-i2s.c

@@ -329,6 +329,7 @@ struct snd_soc_dai pxa_i2s_dai = {
 		.rates = PXA2XX_I2S_RATES,
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
 	.ops = &pxa_i2s_dai_ops,
+	.symmetric_rates = 1,
 };
 
 EXPORT_SYMBOL_GPL(pxa_i2s_dai);

sound/soc/soc-core.c

@@ -992,6 +992,9 @@ static int soc_remove(struct platform_device *pdev)
 	struct snd_soc_platform *platform = card->platform;
 	struct snd_soc_codec_device *codec_dev = socdev->codec_dev;
 
+	if (!card->instantiated)
+		return 0;
+
 	run_delayed_work(&card->delayed_work);
 
 	if (platform->remove)
@@ -2387,6 +2390,39 @@ void snd_soc_unregister_platform(struct snd_soc_platform *platform)
 }
 EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
 
+static u64 codec_format_map[] = {
+	SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE,
+	SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_U16_BE,
+	SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE,
+	SNDRV_PCM_FMTBIT_U24_LE | SNDRV_PCM_FMTBIT_U24_BE,
+	SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S32_BE,
+	SNDRV_PCM_FMTBIT_U32_LE | SNDRV_PCM_FMTBIT_U32_BE,
+	SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+	SNDRV_PCM_FMTBIT_U24_3LE | SNDRV_PCM_FMTBIT_U24_3BE,
+	SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE,
+	SNDRV_PCM_FMTBIT_U20_3LE | SNDRV_PCM_FMTBIT_U20_3BE,
+	SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE,
+	SNDRV_PCM_FMTBIT_U18_3LE | SNDRV_PCM_FMTBIT_U18_3BE,
+	SNDRV_PCM_FMTBIT_FLOAT_LE | SNDRV_PCM_FMTBIT_FLOAT_BE,
+	SNDRV_PCM_FMTBIT_FLOAT64_LE | SNDRV_PCM_FMTBIT_FLOAT64_BE,
+	SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE
+	| SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+};
+
+/* Fix up the DAI formats for endianness: codecs don't actually see
+ * the endianness of the data but we're using the CPU format
+ * definitions which do need to include endianness so we ensure that
+ * codec DAIs always have both big and little endian variants set.
+ */
+static void fixup_codec_formats(struct snd_soc_pcm_stream *stream)
+{
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(codec_format_map); i++)
+		if (stream->formats & codec_format_map[i])
+			stream->formats |= codec_format_map[i];
+}
+
 /**
  * snd_soc_register_codec - Register a codec with the ASoC core
  *
@@ -2394,6 +2430,8 @@ EXPORT_SYMBOL_GPL(snd_soc_unregister_platform);
  */
 int snd_soc_register_codec(struct snd_soc_codec *codec)
 {
+	int i;
+
 	if (!codec->name)
 		return -EINVAL;
 
@@ -2403,6 +2441,11 @@ int snd_soc_register_codec(struct snd_soc_codec *codec)
 
 	INIT_LIST_HEAD(&codec->list);
 
+	for (i = 0; i < codec->num_dai; i++) {
+		fixup_codec_formats(&codec->dai[i].playback);
+		fixup_codec_formats(&codec->dai[i].capture);
+	}
+
 	mutex_lock(&client_mutex);
 	list_add(&codec->list, &codec_list);
 	snd_soc_instantiate_cards();