Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (308 commits)
  ALSA: sound/pci/asihpi: check adapter index in hpi_ioctl
  ALSA: aloop - Fix possible IRQ lock inversion
  ALSA: sound/core: merge list_del()/list_add_tail() to list_move_tail()
  ALSA: ctxfi - use list_move() instead of list_del()/list_add() combination
  ALSA: firewire - msleep needs delay.h
  ALSA: firewire-lib, firewire-speakers: handle packet queueing errors
  ALSA: firewire-lib: allocate DMA buffer separately
  ALSA: firewire-lib: use no-info SYT for packets without SYT sample
  ALSA: add LaCie FireWire Speakers/Griffin FireWave Surround driver
  ALSA: hda - Remove an unused variable in patch_realtek.c
  ALSA: hda - pin-adc-mux-dmic auto-configuration of 92HD8X codecs
  ALSA: hda - fix digital mic selection in mixer on 92HD8X codecs
  ALSA: hda - Move default input-src selection to init part
  ALSA: hda - Initialize special cases for input src in init phase
  ALSA: ctxfi - Clear input settings before initialization
  ALSA: ctxfi - Fix SPDIF status retrieval
  ALSA: ctxfi - Fix incorrect SPDIF status bit mask
  ALSA: ctxfi - Fix microphone boost codes/comments
  ALSA: atiixp - Fix wrong time-out checks during ac-link reset
  ALSA: intel8x0m: append 'm' to "r_intel8x0"
  ...

6 files changed, 172 insertions, 17 deletions

diff --git a/sound/soc/ep93xx/Kconfig b/sound/soc/ep93xx/Kconfig
index 57429041189..91a28de9410 100644
--- a/sound/soc/ep93xx/Kconfig
+++ b/sound/soc/ep93xx/Kconfig
@@ -30,3 +30,12 @@ config SND_EP93XX_SOC_SIMONE
 	help
 	  Say Y or M here if you want to add support for AC97 audio on the
 	  Simplemachines Sim.One board.
+
+config SND_EP93XX_SOC_EDB93XX
+	tristate "SoC Audio support for Cirrus Logic EDB93xx boards"
+	depends on SND_EP93XX_SOC && (MACH_EDB9301 || MACH_EDB9302 || MACH_EDB9302A || MACH_EDB9307A || MACH_EDB9315A)
+	select SND_EP93XX_SOC_I2S
+	select SND_SOC_CS4271
+	help
+	  Say Y or M here if you want to add support for I2S audio on the
+	  Cirrus Logic EDB93xx boards.
diff --git a/sound/soc/ep93xx/Makefile b/sound/soc/ep93xx/Makefile
index 8e7977fb6b7..5514146cbdf 100644
--- a/sound/soc/ep93xx/Makefile
+++ b/sound/soc/ep93xx/Makefile
@@ -10,6 +10,8 @@ obj-$(CONFIG_SND_EP93XX_SOC_AC97)		+= snd-soc-ep93xx-ac97.o
 # EP93XX Machine Support
 snd-soc-snappercl15-objs			:= snappercl15.o
 snd-soc-simone-objs				:= simone.o
+snd-soc-edb93xx-objs				:= edb93xx.o
 
 obj-$(CONFIG_SND_EP93XX_SOC_SNAPPERCL15)	+= snd-soc-snappercl15.o
 obj-$(CONFIG_SND_EP93XX_SOC_SIMONE)		+= snd-soc-simone.o
+obj-$(CONFIG_SND_EP93XX_SOC_EDB93XX)		+= snd-soc-edb93xx.o
diff --git a/sound/soc/ep93xx/edb93xx.c b/sound/soc/ep93xx/edb93xx.c
new file mode 100644
index 00000000000..d3aa15119d2
--- /dev/null
+++ b/sound/soc/ep93xx/edb93xx.c
@@ -0,0 +1,142 @@
+/*
+ * SoC audio for EDB93xx
+ *
+ * Copyright (c) 2010 Alexander Sverdlin <subaparts@yandex.ru>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * as published by the Free Software Foundation; either version 2
+ * of the License, or (at your option) any later version.
+ *
+ * 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.
+ *
+ * This driver support CS4271 codec being master or slave, working
+ * in control port mode, connected either via SPI or I2C.
+ * The data format accepted is I2S or left-justified.
+ * DAPM support not implemented.
+ */
+
+#include <linux/platform_device.h>
+#include <linux/gpio.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include "ep93xx-pcm.h"
+
+#define edb93xx_has_audio() (machine_is_edb9301() ||	\
+			     machine_is_edb9302() ||	\
+			     machine_is_edb9302a() ||	\
+			     machine_is_edb9307a() ||	\
+			     machine_is_edb9315a())
+
+static int edb93xx_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;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int err;
+	unsigned int mclk_rate;
+	unsigned int rate = params_rate(params);
+
+	/*
+	 * According to CS4271 datasheet we use MCLK/LRCK=256 for
+	 * rates below 50kHz and 128 for higher sample rates
+	 */
+	if (rate < 50000)
+		mclk_rate = rate * 64 * 4;
+	else
+		mclk_rate = rate * 64 * 2;
+
+	err = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (err)
+		return err;
+
+	err = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBS_CFS);
+	if (err)
+		return err;
+
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, mclk_rate,
+				     SND_SOC_CLOCK_IN);
+	if (err)
+		return err;
+
+	return snd_soc_dai_set_sysclk(cpu_dai, 0, mclk_rate,
+				      SND_SOC_CLOCK_OUT);
+}
+
+static struct snd_soc_ops edb93xx_ops = {
+	.hw_params	= edb93xx_hw_params,
+};
+
+static struct snd_soc_dai_link edb93xx_dai = {
+	.name		= "CS4271",
+	.stream_name	= "CS4271 HiFi",
+	.platform_name	= "ep93xx-pcm-audio",
+	.cpu_dai_name	= "ep93xx-i2s",
+	.codec_name	= "spi0.0",
+	.codec_dai_name	= "cs4271-hifi",
+	.ops		= &edb93xx_ops,
+};
+
+static struct snd_soc_card snd_soc_edb93xx = {
+	.name		= "EDB93XX",
+	.dai_link	= &edb93xx_dai,
+	.num_links	= 1,
+};
+
+static struct platform_device *edb93xx_snd_device;
+
+static int __init edb93xx_init(void)
+{
+	int ret;
+
+	if (!edb93xx_has_audio())
+		return -ENODEV;
+
+	ret = ep93xx_i2s_acquire(EP93XX_SYSCON_DEVCFG_I2SONAC97,
+				 EP93XX_SYSCON_I2SCLKDIV_ORIDE |
+				 EP93XX_SYSCON_I2SCLKDIV_SPOL);
+	if (ret)
+		return ret;
+
+	edb93xx_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!edb93xx_snd_device) {
+		ret = -ENOMEM;
+		goto free_i2s;
+	}
+
+	platform_set_drvdata(edb93xx_snd_device, &snd_soc_edb93xx);
+	ret = platform_device_add(edb93xx_snd_device);
+	if (ret)
+		goto device_put;
+
+	return 0;
+
+device_put:
+	platform_device_put(edb93xx_snd_device);
+free_i2s:
+	ep93xx_i2s_release();
+	return ret;
+}
+module_init(edb93xx_init);
+
+static void __exit edb93xx_exit(void)
+{
+	platform_device_unregister(edb93xx_snd_device);
+	ep93xx_i2s_release();
+}
+module_exit(edb93xx_exit);
+
+MODULE_AUTHOR("Alexander Sverdlin <subaparts@yandex.ru>");
+MODULE_DESCRIPTION("ALSA SoC EDB93xx");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ep93xx/ep93xx-ac97.c b/sound/soc/ep93xx/ep93xx-ac97.c
index 68a0bae1208..104e95cda0a 100644
--- a/sound/soc/ep93xx/ep93xx-ac97.c
+++ b/sound/soc/ep93xx/ep93xx-ac97.c
@@ -253,7 +253,6 @@ static int ep93xx_ac97_trigger(struct snd_pcm_substream *substream,
 	struct ep93xx_ac97_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned v = 0;
 
-
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
diff --git a/sound/soc/ep93xx/ep93xx-i2s.c b/sound/soc/ep93xx/ep93xx-i2s.c
index fff579a1c13..042f4e93746 100644
--- a/sound/soc/ep93xx/ep93xx-i2s.c
+++ b/sound/soc/ep93xx/ep93xx-i2s.c
@@ -242,7 +242,7 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 {
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 	unsigned word_len, div, sdiv, lrdiv;
-	int found = 0, err;
+	int err;
 
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
@@ -275,15 +275,14 @@ static int ep93xx_i2s_hw_params(struct snd_pcm_substream *substream,
 	 * the codec uses.
 	 */
 	div = clk_get_rate(info->mclk) / params_rate(params);
-	for (sdiv = 2; sdiv <= 4; sdiv += 2)
-		for (lrdiv = 64; lrdiv <= 128; lrdiv <<= 1)
-			if (sdiv * lrdiv == div) {
-				found = 1;
-				goto out;
-			}
-out:
-	if (!found)
-		return -EINVAL;
+	sdiv = 4;
+	if (div > (256 + 512) / 2) {
+		lrdiv = 128;
+	} else {
+		lrdiv = 64;
+		if (div < (128 + 256) / 2)
+			sdiv = 2;
+	}
 
 	err = clk_set_rate(info->sclk, clk_get_rate(info->mclk) / sdiv);
 	if (err)
@@ -314,10 +313,12 @@ static int ep93xx_i2s_suspend(struct snd_soc_dai *dai)
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_disable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 
 static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
@@ -325,10 +326,12 @@ static int ep93xx_i2s_resume(struct snd_soc_dai *dai)
 	struct ep93xx_i2s_info *info = snd_soc_dai_get_drvdata(dai);
 
 	if (!dai->active)
-		return;
+		return 0;
 
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_PLAYBACK);
 	ep93xx_i2s_enable(info, SNDRV_PCM_STREAM_CAPTURE);
+
+	return 0;
 }
 #else
 #define ep93xx_i2s_suspend	NULL
@@ -352,13 +355,13 @@ static struct snd_soc_dai_driver ep93xx_i2s_dai = {
 	.playback	= {
 		.channels_min	= 2,
 		.channels_max	= 2,
-		.rates		= SNDRV_PCM_RATE_8000_96000,
+		.rates		= SNDRV_PCM_RATE_8000_192000,
 		.formats	= EP93XX_I2S_FORMATS,
 	},
 	.capture	= {
 		 .channels_min	= 2,
 		 .channels_max	= 2,
-		 .rates		= SNDRV_PCM_RATE_8000_96000,
+		 .rates		= SNDRV_PCM_RATE_8000_192000,
 		 .formats	= EP93XX_I2S_FORMATS,
 	},
 	.ops		= &ep93xx_i2s_dai_ops,
diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c
index 06670776f64..a456e491155 100644
--- a/sound/soc/ep93xx/ep93xx-pcm.c
+++ b/sound/soc/ep93xx/ep93xx-pcm.c
@@ -35,9 +35,9 @@ static const struct snd_pcm_hardware ep93xx_pcm_hardware = {
 				   SNDRV_PCM_INFO_INTERLEAVED	|
 				   SNDRV_PCM_INFO_BLOCK_TRANSFER),
 				   
-	.rates			= SNDRV_PCM_RATE_8000_96000,
+	.rates			= SNDRV_PCM_RATE_8000_192000,
 	.rate_min		= SNDRV_PCM_RATE_8000,
-	.rate_max		= SNDRV_PCM_RATE_96000,
+	.rate_max		= SNDRV_PCM_RATE_192000,
 	
 	.formats		= (SNDRV_PCM_FMTBIT_S16_LE |
 				   SNDRV_PCM_FMTBIT_S24_LE |