r877 - trunk/src/target/kernel/patches
laforge at sita.openmoko.org
laforge at sita.openmoko.org
Tue Feb 6 15:57:49 CET 2007
Author: laforge
Date: 2007-02-06 15:57:47 +0100 (Tue, 06 Feb 2007)
New Revision: 877
Modified:
trunk/src/target/kernel/patches/asoc-neo_backport.patch
trunk/src/target/kernel/patches/asoc.patch
Log:
update to ASoC 0.13.3 and neo_backport_v6.patch from Wolfson
Modified: trunk/src/target/kernel/patches/asoc-neo_backport.patch
===================================================================
--- trunk/src/target/kernel/patches/asoc-neo_backport.patch 2007-02-06 14:47:50 UTC (rev 876)
+++ trunk/src/target/kernel/patches/asoc-neo_backport.patch 2007-02-06 14:57:47 UTC (rev 877)
@@ -1,32 +1,30 @@
-This backports ASoC to our 2.6.17.14 based kernel
-
Index: linux-2.6.17.14/include/sound/soc.h
===================================================================
--- linux-2.6.17.14.orig/include/sound/soc.h
+++ linux-2.6.17.14/include/sound/soc.h
-@@ -342,7 +342,7 @@ struct snd_soc_codec {
- struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
- unsigned int active;
- unsigned int pcm_devs;
+@@ -359,7 +359,7 @@ struct snd_soc_codec {
+ struct list_head dapm_paths;
+ unsigned int dapm_state;
+ unsigned int suspend_dapm_state;
- struct delayed_work delayed_work;
+ struct work_struct delayed_work;
- void *private_data;
- /* codec IO */
-@@ -437,7 +437,7 @@ struct snd_soc_device {
+ /* codec DAI's */
+ struct snd_soc_codec_dai *dai;
+@@ -436,7 +436,7 @@ struct snd_soc_device {
+ struct snd_soc_platform *platform;
struct snd_soc_codec *codec;
struct snd_soc_codec_device *codec_dev;
- void *codec_data;
- struct delayed_work delayed_work;
+ struct work_struct delayed_work;
+ void *codec_data;
};
- /* runtime channel data */
Index: linux-2.6.17.14/sound/soc/codecs/wm8753.c
===================================================================
--- linux-2.6.17.14.orig/sound/soc/codecs/wm8753.c
+++ linux-2.6.17.14/sound/soc/codecs/wm8753.c
-@@ -1423,10 +1423,9 @@ static void wm8753_set_dai_mode(struct s
+@@ -1422,10 +1422,9 @@ static void wm8753_set_dai_mode(struct s
wm8753_dai[1].codec = codec;
}
@@ -39,20 +37,29 @@
wm8753_dapm_event(codec, codec->dapm_state);
}
-@@ -1670,7 +1669,7 @@ static int wm8753_probe(struct platform_
+@@ -1669,7 +1668,7 @@ static int wm8753_probe(struct platform_
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
wm8753_socdev = socdev;
- INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
+ INIT_WORK(&codec->delayed_work, wm8753_work, codec);
- wm8753_workq = create_workqueue("wm8753");
- if (wm8753_workq == NULL) {
- kfree(codec);
+
+ #if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
Index: linux-2.6.17.14/sound/soc/soc-core.c
===================================================================
--- linux-2.6.17.14.orig/sound/soc/soc-core.c
+++ linux-2.6.17.14/sound/soc/soc-core.c
-@@ -273,10 +273,9 @@ out:
+@@ -64,7 +64,7 @@ MODULE_PARM_DESC(pmdown_time, "DAPM stre
+ /*
+ * This function forces any delayed work to be queued and run.
+ */
+-static int run_delayed_work(struct delayed_work *dwork)
++static int run_delayed_work(struct work_struct *dwork)
+ {
+ int ret;
+
+@@ -267,10 +267,9 @@ out:
* This is to ensure there are no pops or clicks in between any music tracks
* due to DAPM power cycling.
*/
@@ -65,7 +72,7 @@
struct snd_soc_codec *codec = socdev->codec;
struct snd_soc_codec_dai *codec_dai;
int i;
-@@ -781,7 +780,7 @@ static int soc_probe(struct platform_dev
+@@ -775,7 +774,7 @@ static int soc_probe(struct platform_dev
}
/* DAPM stream work */
@@ -74,37 +81,20 @@
return 0;
platform_err:
-@@ -1075,15 +1074,7 @@ int snd_soc_new_pcms(struct snd_soc_devi
- return -ENODEV;
- }
-
-- codec->card->dev = device_create(sound_class, codec->card->parent, 0,
-- "card%i", codec->card->number);
-- if (IS_ERR(codec->card->dev)) {
-- snd_card_free(codec->card);
-- printk(KERN_ERR "asoc: could not create sound dev %s\n",
-- codec->name);
-- return -ENODEV;
-- }
--
-+ codec->card->dev = socdev->dev;
- codec->card->private_data = codec;
- strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
-
Index: linux-2.6.17.14/sound/soc/s3c24xx/s3c24xx-i2s.c
===================================================================
--- linux-2.6.17.14.orig/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ linux-2.6.17.14/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -60,13 +60,13 @@ static struct s3c2410_dma_client s3c24xx
- static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_out = {
+ static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = {
.client = &s3c24xx_dma_client_out,
- .channel = DMACH_I2S_OUT,
+ .channel = 2,
.dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO
};
- static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_in = {
+ static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = {
.client = &s3c24xx_dma_client_in,
- .channel = DMACH_I2S_IN,
+ .channel = 1,
@@ -115,19 +105,19 @@
===================================================================
--- linux-2.6.17.14.orig/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ linux-2.6.17.14/sound/soc/s3c24xx/s3c24xx-pcm.c
-@@ -119,9 +119,9 @@ static void s3c24xx_pcm_enqueue(struct s
+@@ -114,9 +114,9 @@ static void s3c24xx_pcm_enqueue(struct s
prtd->dma_pos = pos;
}
--void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
-+void s3c24xx_audio_buffdone(s3c2410_dma_chan_t *channel,
+-static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
++static void s3c24xx_audio_buffdone(s3c2410_dma_chan_t *channel,
void *dev_id, int size,
- enum s3c2410_dma_buffresult result)
+ s3c2410_dma_buffresult_t result)
{
struct snd_pcm_substream *substream = dev_id;
struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-@@ -182,7 +182,8 @@ static int s3c24xx_pcm_hw_params(struct
+@@ -180,7 +180,8 @@ static int s3c24xx_pcm_hw_params(struct
2, S3C2410_DCON_SYNC_PCLK | S3C2410_DCON_HANDSHAKE);
} else {
s3c2410_dma_config(prtd->params->channel,
@@ -137,7 +127,7 @@
s3c2410_dma_devconfig(prtd->params->channel,
S3C2410_DMASRC_HW, 0x3,
-@@ -260,7 +261,7 @@ static int s3c24xx_pcm_trigger(struct sn
+@@ -261,7 +262,7 @@ static int s3c24xx_pcm_trigger(struct sn
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
prtd->state |= ST_RUNNING;
s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
@@ -146,19 +136,6 @@
break;
case SNDRV_PCM_TRIGGER_STOP:
-Index: linux-2.6.17.14/sound/soc/s3c24xx/Kconfig
-===================================================================
---- linux-2.6.17.14.orig/sound/soc/s3c24xx/Kconfig
-+++ linux-2.6.17.14/sound/soc/s3c24xx/Kconfig
-@@ -21,7 +21,7 @@ config SND_S3C24XX_SOC_SMDK2440
-
- config SND_S3C24XX_SOC_SMDK2440_WM8753
- tristate "SoC I2S Audio support for S3C24XX - WM8753"
-- depends on SND_S3C24XX_SOC && MACH_SMDK
-+ depends on SND_S3C24XX_SOC && MACH_GTA01
- select SND_S3C24XX_SOC_I2S
- help
- Say Y if you want to add support for SoC audio on smdk2440
Index: linux-2.6.17.14/drivers/i2c/i2c-core.c
===================================================================
--- linux-2.6.17.14.orig/drivers/i2c/i2c-core.c
Modified: trunk/src/target/kernel/patches/asoc.patch
===================================================================
--- trunk/src/target/kernel/patches/asoc.patch 2007-02-06 14:47:50 UTC (rev 876)
+++ trunk/src/target/kernel/patches/asoc.patch 2007-02-06 14:57:47 UTC (rev 877)
@@ -1,9 +1,7 @@
-This is ASoC v0.13rc3 from http://opensource.wolfsonmicro.com/~lg/asoc/asoc-v0.13rc3.patch
-
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/DAI.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/DAI.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/DAI.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,117 @@
+ASoC currently supports the three main Digital Audio Interfaces (DAI) found on
+SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM.
@@ -125,7 +123,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/clocking.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/clocking.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/clocking.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,47 @@
+Audio Clocking
+==============
@@ -177,7 +175,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/codec.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/codec.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/codec.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,196 @@
+ASoC Codec Driver
+=================
@@ -378,7 +376,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/dapm.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/dapm.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/dapm.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,297 @@
+Dynamic Audio Power Management for Portable Devices
+===================================================
@@ -680,7 +678,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/machine.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/machine.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/machine.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,113 @@
+ASoC Machine Driver
+===================
@@ -799,7 +797,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/overview.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/overview.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/overview.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,83 @@
+ALSA SoC Layer
+==============
@@ -888,7 +886,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/platform.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/platform.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/platform.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,58 @@
+ASoC Platform Driver
+====================
@@ -951,7 +949,7 @@
Index: linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/pops_clicks.txt
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/pops_clicks.txt 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/Documentation/sound/alsa/soc/pops_clicks.txt 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,52 @@
+Audio Pops and Clicks
+=====================
@@ -1007,8 +1005,8 @@
+crosses the zero amplitude line.
Index: linux-2.6.17.14-fic4.test/include/sound/ac97_codec.h
===================================================================
---- linux-2.6.17.14-fic4.test.orig/include/sound/ac97_codec.h 2007-01-31 21:34:15.000000000 +0100
-+++ linux-2.6.17.14-fic4.test/include/sound/ac97_codec.h 2007-01-31 21:36:05.000000000 +0100
+--- linux-2.6.17.14-fic4.test.orig/include/sound/ac97_codec.h 2007-02-06 15:38:06.000000000 +0100
++++ linux-2.6.17.14-fic4.test/include/sound/ac97_codec.h 2007-02-06 15:55:20.000000000 +0100
@@ -407,6 +407,7 @@
struct snd_ac97_bus_ops {
@@ -1020,7 +1018,7 @@
Index: linux-2.6.17.14-fic4.test/include/sound/soc-dapm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/include/sound/soc-dapm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/include/sound/soc-dapm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,286 @@
+/*
+ * linux/sound/soc-dapm.h -- ALSA SoC Dynamic Audio Power Management
@@ -1311,7 +1309,7 @@
Index: linux-2.6.17.14-fic4.test/include/sound/soc.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/include/sound/soc.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/include/sound/soc.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,460 @@
+/*
+ * linux/sound/soc.h -- ALSA SoC Layer
@@ -1336,7 +1334,7 @@
+#include <sound/control.h>
+#include <sound/ac97_codec.h>
+
-+#define SND_SOC_VERSION "0.13.0rc3"
++#define SND_SOC_VERSION "0.13.3"
+
+/*
+ * Convenience kcontrol builders
@@ -1482,7 +1480,6 @@
+/* set runtime hw params */
+int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
+ const struct snd_pcm_hardware *hw);
-+int snd_soc_get_rate(int rate);
+
+/* codec IO */
+#define snd_soc_read(codec, reg) codec->read(codec, reg)
@@ -1540,7 +1537,7 @@
+ unsigned int active:1; /* stream is in use */
+};
+
-+/* ASoC alsa audio ops */
++/* SoC audio ops */
+struct snd_soc_ops {
+ int (*startup)(struct snd_pcm_substream *);
+ void (*shutdown)(struct snd_pcm_substream *);
@@ -1619,6 +1616,7 @@
+ unsigned int id;
+ unsigned char type;
+
++ /* DAI callbacks */
+ int (*probe)(struct platform_device *pdev);
+ void (*remove)(struct platform_device *pdev);
+ int (*suspend)(struct platform_device *pdev,
@@ -1657,7 +1655,6 @@
+ struct snd_ac97 *ac97; /* for ad-hoc ac97 devices */
+ unsigned int active;
+ unsigned int pcm_devs;
-+ struct delayed_work delayed_work;
+ void *private_data;
+
+ /* codec IO */
@@ -1675,6 +1672,7 @@
+ struct list_head dapm_paths;
+ unsigned int dapm_state;
+ unsigned int suspend_dapm_state;
++ struct delayed_work delayed_work;
+
+ /* codec DAI's */
+ struct snd_soc_codec_dai *dai;
@@ -1751,8 +1749,8 @@
+ struct snd_soc_platform *platform;
+ struct snd_soc_codec *codec;
+ struct snd_soc_codec_device *codec_dev;
++ struct delayed_work delayed_work;
+ void *codec_data;
-+ struct delayed_work delayed_work;
+};
+
+/* runtime channel data */
@@ -1775,8 +1773,8 @@
+#endif
Index: linux-2.6.17.14-fic4.test/sound/Kconfig
===================================================================
---- linux-2.6.17.14-fic4.test.orig/sound/Kconfig 2007-01-31 21:34:15.000000000 +0100
-+++ linux-2.6.17.14-fic4.test/sound/Kconfig 2007-01-31 21:36:05.000000000 +0100
+--- linux-2.6.17.14-fic4.test.orig/sound/Kconfig 2007-02-06 15:38:06.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/Kconfig 2007-02-06 15:55:20.000000000 +0100
@@ -74,6 +74,8 @@
source "sound/parisc/Kconfig"
@@ -1789,7 +1787,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/Kconfig 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/Kconfig 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,37 @@
+#
+# SoC audio configuration
@@ -1831,7 +1829,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/Makefile 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/Makefile 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,4 @@
+snd-soc-core-objs := soc-core.o soc-dapm.o
+
@@ -1840,7 +1838,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Kconfig 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Kconfig 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,102 @@
+config SND_SOC_AC97_CODEC
+ tristate "SoC generic AC97 support"
@@ -1947,7 +1945,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Makefile 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/Makefile 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,35 @@
+snd-soc-ac97-objs := ac97.o
+snd-soc-wm8711-objs := wm8711.o
@@ -1987,7 +1985,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,156 @@
+/*
+ * ac97.c -- ALSA Soc AC97 codec support
@@ -2148,7 +2146,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ac97.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,18 @@
+/*
+ * linux/sound/codecs/ac97.h -- ALSA SoC Layer
@@ -2171,7 +2169,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,697 @@
+/*
+ * ak4535.c -- AK4535 ALSA Soc Audio driver
@@ -2873,7 +2871,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/ak4535.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,46 @@
+/*
+ * ak4535.h -- AK4535 Soc Audio driver
@@ -2924,8 +2922,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,747 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,745 @@
+/*
+ * uda1380.c - Philips UDA1380 ALSA SoC audio driver
+ *
@@ -2978,15 +2976,15 @@
+ * uda1380 register cache
+ */
+static const u16 uda1380_reg[UDA1380_CACHEREGNUM] = {
-+ 0x0502, 0x0000, 0x0000, 0x3f3f,
-+ 0x0202, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0xff00, 0x0000, 0x4800,
-+ 0x0000, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0x0000, 0x0000, 0x0000,
-+ 0x0000, 0x8000, 0x0002, 0x0000,
++ 0x0502, 0x0000, 0x0000, 0x3f3f,
++ 0x0202, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0xff00, 0x0000, 0x4800,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x0000, 0x0000, 0x0000,
++ 0x0000, 0x8000, 0x0002, 0x0000,
+};
+
+/*
@@ -3058,10 +3056,11 @@
+
+/* declarations of ALSA reg_elem_REAL controls */
+static const char *uda1380_deemp[] = {"None", "32kHz", "44.1kHz", "48kHz",
-+ "96kHz"};
++ "96kHz"};
+static const char *uda1380_input_sel[] = {"Line", "Mic"};
+static const char *uda1380_output_sel[] = {"Direct", "Mixer"};
-+static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2", "Maximum"};
++static const char *uda1380_spf_mode[] = {"Flat", "Minimum1", "Minimum2",
++ "Maximum"};
+
+static const struct soc_enum uda1380_enum[] = {
+ SOC_ENUM_DOUBLE(UDA1380_DEEMP, 0, 8, 5, uda1380_deemp),
@@ -3173,15 +3172,13 @@
+{
+ int i;
+
-+ for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
++ for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
-+ }
+
+ /* set up audio path interconnects */
-+ for(i = 0; audio_map[i][0] != NULL; i++) {
++ for (i = 0; audio_map[i][0] != NULL; i++)
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
-+ }
+
+ snd_soc_dapm_new_widgets(codec);
+ return 0;
@@ -3198,7 +3195,7 @@
+ iface &= ~(R01_SFORI_MASK | R01_SIM | R01_SFORO_MASK);
+
+ /* FIXME: how to select I2S for DATAO and MSB for DATAI correctly? */
-+ switch(fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
++ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_I2S:
+ iface |= R01_SFORI_I2S | R01_SFORO_I2S;
+ break;
@@ -3209,9 +3206,8 @@
+ iface |= R01_SFORI_MSB | R01_SFORO_I2S;
+ }
+
-+ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM) {
++ if ((fmt & SND_SOC_DAIFMT_MASTER_MASK) == SND_SOC_DAIFMT_CBM_CFM)
+ iface |= R01_SIM;
-+ }
+
+ uda1380_write(codec, UDA1380_IFACE, iface);
+
@@ -3324,7 +3320,7 @@
+ * registers works only when clocked by SYSCLK */
+ u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK);
+ uda1380_write(codec, UDA1380_CLK, ~R00_DAC_CLK & clk);
-+ if(mute)
++ if (mute)
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg | 0x4000);
+ else
+ uda1380_write(codec, UDA1380_DEEMP, mute_reg);
@@ -3337,28 +3333,28 @@
+ int pm = uda1380_read_reg_cache(codec, UDA1380_PM);
+
+ switch (event) {
-+ case SNDRV_CTL_POWER_D0: /* full On */
-+ case SNDRV_CTL_POWER_D1: /* partial On */
-+ case SNDRV_CTL_POWER_D2: /* partial On */
-+ /* enable internal bias */
-+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
-+ break;
-+ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
-+ /* everything off except internal bias */
-+ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
-+ break;
-+ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
-+ /* everything off, inactive */
-+ uda1380_write(codec, UDA1380_PM, 0x0);
-+ break;
++ case SNDRV_CTL_POWER_D0: /* full On */
++ case SNDRV_CTL_POWER_D1: /* partial On */
++ case SNDRV_CTL_POWER_D2: /* partial On */
++ /* enable internal bias */
++ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS | pm);
++ break;
++ case SNDRV_CTL_POWER_D3hot: /* Off, with power */
++ /* everything off except internal bias */
++ uda1380_write(codec, UDA1380_PM, R02_PON_BIAS);
++ break;
++ case SNDRV_CTL_POWER_D3cold: /* Off, without power */
++ /* everything off, inactive */
++ uda1380_write(codec, UDA1380_PM, 0x0);
++ break;
+ }
+ codec->dapm_state = event;
+ return 0;
+}
+
+#define UDA1380_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \
-+ SNDRV_PCM_RATE_48000)
++ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
++ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000)
+
+struct snd_soc_codec_dai uda1380_dai[] = {
+{
@@ -3471,11 +3467,11 @@
+ codec->num_dai = ARRAY_SIZE(uda1380_dai);
+ codec->reg_cache_size = ARRAY_SIZE(uda1380_reg);
+ codec->reg_cache =
-+ kzalloc(sizeof(u16) * ARRAY_SIZE(uda1380_reg), GFP_KERNEL);
++ kcalloc(ARRAY_SIZE(uda1380_reg), sizeof(u16), GFP_KERNEL);
+ if (codec->reg_cache == NULL)
+ return -ENOMEM;
+ memcpy(codec->reg_cache, uda1380_reg,
-+ sizeof(u16) * ARRAY_SIZE(uda1380_reg));
++ sizeof(u16) * ARRAY_SIZE(uda1380_reg));
+ codec->reg_cache_size = sizeof(u16) * ARRAY_SIZE(uda1380_reg);
+ uda1380_reset(codec);
+
@@ -3503,9 +3499,9 @@
+ uda1380_add_widgets(codec);
+ ret = snd_soc_register_card(socdev);
+ if (ret < 0) {
-+ printk(KERN_ERR "uda1380: failed to register card\n");
++ printk(KERN_ERR "uda1380: failed to register card\n");
+ goto card_err;
-+ }
++ }
+
+ return ret;
+
@@ -3558,13 +3554,13 @@
+ codec->control_data = i2c;
+
+ ret = i2c_attach_client(i2c);
-+ if(ret < 0) {
++ if (ret < 0) {
+ printk(KERN_ERR "failed to attach codec at addr %x\n", addr);
+ goto err;
+ }
+
+ ret = uda1380_init(socdev, setup->dac_clk);
-+ if(ret < 0) {
++ if (ret < 0) {
+ printk(KERN_ERR "failed to initialise UDA1380\n");
+ goto err;
+ }
@@ -3676,8 +3672,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.h 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,84 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/uda1380.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,89 @@
+/*
+ * Audio support for Philips UDA1380
+ *
@@ -3688,21 +3684,24 @@
+ * Copyright (c) 2005 Giorgio Padrin <giorgio at mandarinlogiq.org>
+ */
+
-+#define UDA1380_CLK 0x00
++#ifndef _UDA1380_H
++#define _UDA1380_H
++
++#define UDA1380_CLK 0x00
+#define UDA1380_IFACE 0x01
-+#define UDA1380_PM 0x02
++#define UDA1380_PM 0x02
+#define UDA1380_AMIX 0x03
-+#define UDA1380_HP 0x04
++#define UDA1380_HP 0x04
+#define UDA1380_MVOL 0x10
+#define UDA1380_MIXVOL 0x11
+#define UDA1380_MODE 0x12
+#define UDA1380_DEEMP 0x13
+#define UDA1380_MIXER 0x14
+#define UDA1380_INTSTAT 0x18
-+#define UDA1380_DEC 0x20
-+#define UDA1380_PGA 0x21
-+#define UDA1380_ADC 0x22
-+#define UDA1380_AGC 0x23
++#define UDA1380_DEC 0x20
++#define UDA1380_PGA 0x21
++#define UDA1380_ADC 0x22
++#define UDA1380_AGC 0x23
+#define UDA1380_DECSTAT 0x28
+#define UDA1380_RESET 0x7f
+
@@ -3756,16 +3755,18 @@
+#define UDA1380_DAC_CLK_WSPLL 1
+};
+
-+#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
++#define UDA1380_DAI_DUPLEX 0 /* playback and capture on single DAI */
+#define UDA1380_DAI_PLAYBACK 1 /* playback DAI */
-+#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
++#define UDA1380_DAI_CAPTURE 2 /* capture DAI */
+
+extern struct snd_soc_codec_dai uda1380_dai[3];
+extern struct snd_soc_codec_device soc_codec_dev_uda1380;
++
++#endif /* _UDA1380_H */
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,763 @@
+/*
+ * wm8731.c -- WM8731 ALSA SoC Audio driver
@@ -4068,7 +4069,7 @@
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8731_priv *wm8731 = codec->private_data;
+ u16 iface = wm8731_read_reg_cache(codec, WM8731_IFACE) & 0xfff3;
-+ int i = get_coeff(wm8731->sysclk, snd_soc_get_rate(params_rate(params)));
++ int i = get_coeff(wm8731->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
@@ -4533,7 +4534,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8731.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,44 @@
+/*
+ * wm8731.h -- WM8731 Soc Audio driver
@@ -4582,8 +4583,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,1045 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,1055 @@
+/*
+ * wm8750.c -- WM8750 ALSA SoC audio driver
+ *
@@ -4637,8 +4638,6 @@
+#define warn(format, arg...) \
+ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
+
-+static struct workqueue_struct *wm8750_workq = NULL;
-+
+/* codec private data */
+struct wm8750_priv {
+ unsigned int sysclk;
@@ -5373,8 +5372,7 @@
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D0;
-+ queue_delayed_work(wm8750_workq, &codec->delayed_work,
-+ msecs_to_jiffies(1000));
++ schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+ }
+
+ return 0;
@@ -5418,8 +5416,7 @@
+ /* charge output caps */
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
-+ queue_delayed_work(wm8750_workq, &codec->delayed_work,
-+ msecs_to_jiffies(1000));
++ schedule_delayed_work(&codec->delayed_work, msecs_to_jiffies(1000));
+
+ /* set the update bits */
+ reg = wm8750_read_reg_cache(codec, WM8750_LDAC);
@@ -5576,11 +5573,7 @@
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8750_socdev = socdev;
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8750_work);
-+ wm8750_workq = create_workqueue("wm8750");
-+ if (wm8750_workq == NULL) {
-+ kfree(codec);
-+ return -ENOMEM;
-+ }
++
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
@@ -5596,6 +5589,25 @@
+ return ret;
+}
+
++/*
++ * This function forces any delayed work to be queued and run.
++ */
++static int run_delayed_work(struct delayed_work *dwork)
++{
++ int ret;
++
++ /* cancel any work waiting to be queued. */
++ ret = cancel_delayed_work(dwork);
++
++ /* if there was any work waiting then we run it now and
++ * wait for it's completion */
++ if (ret) {
++ schedule_delayed_work(dwork, 0);
++ flush_scheduled_work();
++ }
++ return ret;
++}
++
+/* power down chip */
+static int wm8750_remove(struct platform_device *pdev)
+{
@@ -5604,8 +5616,7 @@
+
+ if (codec->control_data)
+ wm8750_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+ if (wm8750_workq)
-+ destroy_workqueue(wm8750_workq);
++ run_delayed_work(&codec->delayed_work);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
@@ -5632,7 +5643,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8750.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,67 @@
+/*
+ * Copyright 2005 Openedhand Ltd.
@@ -5704,8 +5715,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,1725 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,1738 @@
+/*
+ * wm8753.c -- WM8753 ALSA Soc Audio driver
+ *
@@ -5786,7 +5797,6 @@
+module_param(caps_charge, int, 0);
+MODULE_PARM_DESC(caps_charge, "WM8753 cap charge time (msecs)");
+
-+static struct workqueue_struct *wm8753_workq = NULL;
+static void wm8753_set_dai_mode(struct snd_soc_codec *codec,
+ unsigned int mode);
+
@@ -6386,12 +6396,11 @@
+{
+ int i;
+
-+ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) {
++ for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-+ }
+
+ /* set up the WM8753 audio map */
-+ for(i = 0; audio_map[i][0] != NULL; i++) {
++ for (i = 0; audio_map[i][0] != NULL; i++) {
+ snd_soc_dapm_connect_input(codec, audio_map[i][0],
+ audio_map[i][1], audio_map[i][2]);
+ }
@@ -6407,13 +6416,12 @@
+ u32 k:24;
+};
+
-+static struct _pll_div pll_div;
-+
+/* The size in bits of the pll divide multiplied by 10
+ * to allow rounding later */
+#define FIXED_PLL_SIZE ((1 << 22) * 10)
+
-+static void pll_factors(unsigned int target, unsigned int source)
++static void pll_factors(struct _pll_div *pll_div, unsigned int target,
++ unsigned int source)
+{
+ unsigned long long Kpart;
+ unsigned int K, Ndiv, Nmod;
@@ -6421,16 +6429,16 @@
+ Ndiv = target / source;
+ if (Ndiv < 6) {
+ source >>= 1;
-+ pll_div.div2 = 1;
++ pll_div->div2 = 1;
+ Ndiv = target / source;
+ } else
-+ pll_div.div2 = 0;
++ pll_div->div2 = 0;
+
+ if ((Ndiv < 6) || (Ndiv > 12))
+ printk(KERN_WARNING
+ "WM8753 N value outwith recommended range! N = %d\n",Ndiv);
+
-+ pll_div.n = Ndiv;
++ pll_div->n = Ndiv;
+ Nmod = target % source;
+ Kpart = FIXED_PLL_SIZE * (long long)Nmod;
+
@@ -6445,7 +6453,7 @@
+ /* Move down to proper range now rounding is done */
+ K /= 10;
+
-+ pll_div.k = K;
++ pll_div->k = K;
+}
+
+static int wm8753_set_dai_pll(struct snd_soc_codec_dai *codec_dai,
@@ -6476,8 +6484,9 @@
+ } else {
+
+ u16 value = 0;
++ struct _pll_div pll_div;
+
-+ pll_factors(freq_out * 8, freq_in);
++ pll_factors(&pll_div, freq_out * 8, freq_in);
+
+ /* set up N and K PLL divisor ratios */
+ /* bits 8:5 = PLL_N, bits 3:0 = PLL_K[21:18] */
@@ -6496,7 +6505,7 @@
+ wm8753_write(codec, WM8753_PLL1CTL1 + offset, 0x0027 |
+ (pll_div.div2 << 3));
+ wm8753_write(codec, WM8753_CLOCK, reg | enable);
-+ }
++ }
+ return 0;
+}
+
@@ -6590,6 +6599,7 @@
+ wm8753->pcmclk = freq;
+ return 0;
+ }
++ break;
+ }
+ return -EINVAL;
+}
@@ -7170,7 +7180,7 @@
+ if (codec->suspend_dapm_state == SNDRV_CTL_POWER_D0) {
+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D0;
-+ queue_delayed_work(wm8753_workq, &codec->delayed_work,
++ schedule_delayed_work(&codec->delayed_work,
+ msecs_to_jiffies(caps_charge));
+ }
+
@@ -7216,8 +7226,8 @@
+ /* charge output caps */
+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D2);
+ codec->dapm_state = SNDRV_CTL_POWER_D3hot;
-+ queue_delayed_work(wm8753_workq,
-+ &codec->delayed_work, msecs_to_jiffies(caps_charge));
++ schedule_delayed_work(&codec->delayed_work,
++ msecs_to_jiffies(caps_charge));
+
+ /* set the update bits */
+ reg = wm8753_read_reg_cache(codec, WM8753_LDAC);
@@ -7379,11 +7389,7 @@
+ INIT_LIST_HEAD(&codec->dapm_paths);
+ wm8753_socdev = socdev;
+ INIT_DELAYED_WORK(&codec->delayed_work, wm8753_work);
-+ wm8753_workq = create_workqueue("wm8753");
-+ if (wm8753_workq == NULL) {
-+ kfree(codec);
-+ return -ENOMEM;
-+ }
++
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
+ if (setup->i2c_address) {
+ normal_i2c[0] = setup->i2c_address;
@@ -7398,6 +7404,25 @@
+ return ret;
+}
+
++/*
++ * This function forces any delayed work to be queued and run.
++ */
++static int run_delayed_work(struct delayed_work *dwork)
++{
++ int ret;
++
++ /* cancel any work waiting to be queued. */
++ ret = cancel_delayed_work(dwork);
++
++ /* if there was any work waiting then we run it now and
++ * wait for it's completion */
++ if (ret) {
++ schedule_delayed_work(dwork, 0);
++ flush_scheduled_work();
++ }
++ return ret;
++}
++
+/* power down chip */
+static int wm8753_remove(struct platform_device *pdev)
+{
@@ -7406,8 +7431,7 @@
+
+ if (codec->control_data)
+ wm8753_dapm_event(codec, SNDRV_CTL_POWER_D3cold);
-+ if (wm8753_workq)
-+ destroy_workqueue(wm8753_workq);
++ run_delayed_work(&codec->delayed_work);
+ snd_soc_free_pcms(socdev);
+ snd_soc_dapm_free(socdev);
+#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
@@ -7434,7 +7458,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8753.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,126 @@
+/*
+ * wm8753.h -- audio driver for WM8753
@@ -7459,62 +7483,62 @@
+#define WM8753_ADC 0x02
+#define WM8753_PCM 0x03
+#define WM8753_HIFI 0x04
-+#define WM8753_IOCTL 0x05
-+#define WM8753_SRATE1 0x06
-+#define WM8753_SRATE2 0x07
++#define WM8753_IOCTL 0x05
++#define WM8753_SRATE1 0x06
++#define WM8753_SRATE2 0x07
+#define WM8753_LDAC 0x08
+#define WM8753_RDAC 0x09
+#define WM8753_BASS 0x0a
-+#define WM8753_TREBLE 0x0b
++#define WM8753_TREBLE 0x0b
+#define WM8753_ALC1 0x0c
+#define WM8753_ALC2 0x0d
+#define WM8753_ALC3 0x0e
-+#define WM8753_NGATE 0x0f
++#define WM8753_NGATE 0x0f
+#define WM8753_LADC 0x10
+#define WM8753_RADC 0x11
-+#define WM8753_ADCTL1 0x12
++#define WM8753_ADCTL1 0x12
+#define WM8753_3D 0x13
+#define WM8753_PWR1 0x14
+#define WM8753_PWR2 0x15
+#define WM8753_PWR3 0x16
+#define WM8753_PWR4 0x17
+#define WM8753_ID 0x18
-+#define WM8753_INTPOL 0x19
-+#define WM8753_INTEN 0x1a
-+#define WM8753_GPIO1 0x1b
-+#define WM8753_GPIO2 0x1c
-+#define WM8753_RESET 0x1f
-+#define WM8753_RECMIX1 0x20
-+#define WM8753_RECMIX2 0x21
-+#define WM8753_LOUTM1 0x22
-+#define WM8753_LOUTM2 0x23
-+#define WM8753_ROUTM1 0x24
-+#define WM8753_ROUTM2 0x25
-+#define WM8753_MOUTM1 0x26
-+#define WM8753_MOUTM2 0x27
-+#define WM8753_LOUT1V 0x28
-+#define WM8753_ROUT1V 0x29
-+#define WM8753_LOUT2V 0x2a
-+#define WM8753_ROUT2V 0x2b
-+#define WM8753_MOUTV 0x2c
-+#define WM8753_OUTCTL 0x2d
-+#define WM8753_ADCIN 0x2e
-+#define WM8753_INCTL1 0x2f
-+#define WM8753_INCTL2 0x30
-+#define WM8753_LINVOL 0x31
-+#define WM8753_RINVOL 0x32
-+#define WM8753_MICBIAS 0x33
-+#define WM8753_CLOCK 0x34
-+#define WM8753_PLL1CTL1 0x35
-+#define WM8753_PLL1CTL2 0x36
-+#define WM8753_PLL1CTL3 0x37
-+#define WM8753_PLL1CTL4 0x38
-+#define WM8753_PLL2CTL1 0x39
-+#define WM8753_PLL2CTL2 0x3a
-+#define WM8753_PLL2CTL3 0x3b
-+#define WM8753_PLL2CTL4 0x3c
-+#define WM8753_BIASCTL 0x3d
-+#define WM8753_ADCTL2 0x3f
++#define WM8753_INTPOL 0x19
++#define WM8753_INTEN 0x1a
++#define WM8753_GPIO1 0x1b
++#define WM8753_GPIO2 0x1c
++#define WM8753_RESET 0x1f
++#define WM8753_RECMIX1 0x20
++#define WM8753_RECMIX2 0x21
++#define WM8753_LOUTM1 0x22
++#define WM8753_LOUTM2 0x23
++#define WM8753_ROUTM1 0x24
++#define WM8753_ROUTM2 0x25
++#define WM8753_MOUTM1 0x26
++#define WM8753_MOUTM2 0x27
++#define WM8753_LOUT1V 0x28
++#define WM8753_ROUT1V 0x29
++#define WM8753_LOUT2V 0x2a
++#define WM8753_ROUT2V 0x2b
++#define WM8753_MOUTV 0x2c
++#define WM8753_OUTCTL 0x2d
++#define WM8753_ADCIN 0x2e
++#define WM8753_INCTL1 0x2f
++#define WM8753_INCTL2 0x30
++#define WM8753_LINVOL 0x31
++#define WM8753_RINVOL 0x32
++#define WM8753_MICBIAS 0x33
++#define WM8753_CLOCK 0x34
++#define WM8753_PLL1CTL1 0x35
++#define WM8753_PLL1CTL2 0x36
++#define WM8753_PLL1CTL3 0x37
++#define WM8753_PLL1CTL4 0x38
++#define WM8753_PLL2CTL1 0x39
++#define WM8753_PLL2CTL2 0x3a
++#define WM8753_PLL2CTL3 0x3b
++#define WM8753_PLL2CTL4 0x3c
++#define WM8753_BIASCTL 0x3d
++#define WM8753_ADCTL2 0x3f
+
+struct wm8753_setup_data {
+ unsigned short i2c_address;
@@ -7524,7 +7548,7 @@
+#define WM8753_PLL2 1
+
+/* clock inputs */
-+#define WM8753_MCLK 0
++#define WM8753_MCLK 0
+#define WM8753_PCMCLK 1
+
+/* clock divider id's */
@@ -7556,7 +7580,7 @@
+#define WM8753_VXCLK_DIV_16 (4 << 6)
+
+#define WM8753_DAI_HIFI 0
-+#define WM8753_DAI_VOICE 1
++#define WM8753_DAI_VOICE 1
+
+extern struct snd_soc_codec_dai wm8753_dai[2];
+extern struct snd_soc_codec_device soc_codec_dev_wm8753;
@@ -7565,7 +7589,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,603 @@
+/*
+ * wm8772.c -- WM8772 ALSA Soc Audio driver
@@ -8173,7 +8197,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8772.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,46 @@
+/*
+ * wm8772.h -- audio driver for WM8772
@@ -8224,7 +8248,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,971 @@
+/*
+ * wm8971.c -- WM8971 ALSA SoC Audio driver
@@ -9200,7 +9224,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8971.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,63 @@
+/*
+ * wm8971.h -- audio driver for WM8971
@@ -9268,7 +9292,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,873 @@
+/*
+ * wm8974.c -- WM8974 ALSA Soc Audio driver
@@ -10146,7 +10170,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8974.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,104 @@
+/*
+ * wm8974.h -- WM8974 Soc Audio driver
@@ -10255,7 +10279,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,771 @@
+/*
+ * wm9712.c -- ALSA Soc WM9712 codec support
@@ -11031,7 +11055,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9712.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,14 @@
+/*
+ * wm9712.h -- WM9712 Soc Audio driver
@@ -11050,7 +11074,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,1220 @@
+/*
+ * wm9713.c -- ALSA Soc WM9713 codec support
@@ -12275,7 +12299,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm9713.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,51 @@
+/*
+ * wm9713.h -- WM9713 Soc Audio driver
@@ -12331,8 +12355,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Kconfig 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,134 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Kconfig 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,143 @@
+menu "SoC Audio for the Intel PXA2xx"
+
+config SND_PXA2xx_SOC
@@ -12466,12 +12490,21 @@
+ Say Y if you want to add support for SoC audio on the
+ HTC Magician.
+
++config SND_PXA2xx_SOC_AMESOM_TLV320
++ tristate "SoC SSP Audio support for AMESOM - TLV320AIC24k"
++ depends on SND_PXA2xx_SOC && MACH_AMESOM
++ select SND_PXA2xx_SOC_I2S
++ select SND_PXA2xx_SOC_SSP
++ help
++ Say Y if you want to add support for SoC audio on Amesom
++ with the tlv320.
++
+endmenu
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Makefile 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,37 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/Makefile 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,39 @@
+# PXA Platform Support
+snd-soc-pxa2xx-objs := pxa2xx-pcm.o
+snd-soc-pxa2xx-ac97-objs := pxa2xx-ac97.o
@@ -12496,6 +12529,7 @@
+snd-soc-tosa-objs := tosa.o
+snd-soc-spitz-objs := spitz.o
+snd-soc-magician-objs := magician.o
++snd-soc-amesom-tlv320-objs := amesom_tlv320.o
+
+obj-$(CONFIG_SND_PXA2xx_SOC_CORGI) += snd-soc-corgi.o
+obj-$(CONFIG_SND_PXA2xx_SOC_MAINSTONE_WM8731) += snd-soc-mainstone-wm8731.o
@@ -12509,10 +12543,11 @@
+obj-$(CONFIG_SND_PXA2xx_SOC_TOSA) += snd-soc-tosa.o
+obj-$(CONFIG_SND_PXA2xx_SOC_SPITZ) += snd-soc-spitz.o
+obj-$(CONFIG_SND_PXA2xx_SOC_MAGICIAN) += snd-soc-magician.o
++obj-$(CONFIG_SND_PXA2xx_SOC_AMESOM_TLV320) += snd-soc-amesom-tlv320.o
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/corgi.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/corgi.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/corgi.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,383 @@
+/*
+ * corgi.c -- SoC audio for Corgi
@@ -12900,7 +12935,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,127 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -13032,7 +13067,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_baseband.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_baseband.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_baseband.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,212 @@
+/*
+ * mainstone_baseband.c
@@ -13249,7 +13284,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_bluetooth.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_bluetooth.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_bluetooth.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,371 @@
+/*
+ * mainstone_bluetooth.c
@@ -13625,7 +13660,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8731.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8731.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8731.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,203 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -13833,7 +13868,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8753.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8753.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8753.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,547 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -14385,7 +14420,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8974.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8974.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm8974.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,104 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -14494,7 +14529,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9712.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9712.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9712.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,172 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -14671,7 +14706,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9713.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9713.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/mainstone_wm9713.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,318 @@
+/*
+ * mainstone.c -- SoC audio for Mainstone
@@ -14994,7 +15029,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/poodle.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/poodle.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/poodle.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,352 @@
+/*
+ * poodle.c -- SoC audio for Poodle
@@ -15351,7 +15386,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,431 @@
+/*
+ * linux/sound/pxa2xx-ac97.c -- AC97 support for the Intel PXA2xx chip.
@@ -15787,7 +15822,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,318 @@
+/*
+ * pxa2xx-i2s.c -- ALSA Soc Audio Layer
@@ -16110,7 +16145,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,372 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.c -- ALSA PCM interface for the Intel PXA2xx chip
@@ -16487,7 +16522,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,34 @@
+/*
+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
@@ -16526,8 +16561,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,671 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,666 @@
+/*
+ * pxa2xx-ssp.c -- ALSA Soc Audio Layer
+ *
@@ -16544,8 +16579,6 @@
+ * 12th Aug 2005 Initial version.
+ *
+ * TODO:
-+ * o The SSP driver _mostly_ works, however is in need of testing and
-+ * someone with time to complete it.
+ * o Test network mode for > 16bit sample size
+ */
+
@@ -16568,15 +16601,14 @@
+#include "pxa2xx-pcm.h"
+#include "pxa2xx-ssp.h"
+
-+#define PXA_SSP_DEBUG 1
++#define PXA_SSP_DEBUG 0
+
-+/*
-+ * The following should be defined in pxa-regs.h
-+ */
-+#define SSCR0_ACS (1 << 30) /* Audio Clock Select */
-+#define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass (SSCR0[ACS] must be set) */
-+#define SSACD_ACPS(x) (x << 4) /* Audio clock PLL select */
-+#define SSACD_ACDS(x) (x << 0) /* Audio clock divider select */
++#if PXA_SSP_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
+
+/*
+ * SSP audio private data
@@ -16768,9 +16800,14 @@
+{
+ int port = cpu_dai->id + 1;
+ u32 sscr0 = SSCR0_P(port) &
-+ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS);
++ ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ADC);
+
++ dbg("pxa2xx_ssp_set_dai_sysclk id: %d, clk_id %d, freq %d",
++ cpu_dai->id, clk_id, freq);
++
+ switch (clk_id) {
++ case PXA2XX_SSP_CLK_NET_PLL:
++ sscr0 |= SSCR0_MOD;
+ case PXA2XX_SSP_CLK_PLL:
+ /* Internal PLL is fixed on pxa25x and pxa27x */
+#ifdef CONFIG_PXA27x
@@ -16790,7 +16827,7 @@
+ case PXA2XX_SSP_CLK_AUDIO:
+ ssp_clk[cpu_dai->id].sysclk = 0;
+ SSCR0_P(port) |= SSCR0_SerClkDiv(1);
-+ sscr0 |= SSCR0_ACS;
++ sscr0 |= SSCR0_ADC;
+ break;
+ default:
+ return -ENODEV;
@@ -16918,24 +16955,16 @@
+ /* FIXME: this is what wince uses for msb */
+ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB) {
+ SSCR0_P(port) = SSCR0_EDSS | SSCR0_TISSP | SSCR0_DataSize(16);
-+
-+// SSCR1_P(port) = SSCR1_RxTresh(8) | SSCR1_TxTresh(8); /* doesn't seem to be needed */
-+ return 0;
++ goto master;
+ }
+
+ /* check for I2S emulation mode - handle it separately */
-+ if (((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) ||
-+ ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_MSB)) {
++ if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_I2S) {
+ /* 8.4.11 */
+
+ /* Only SSCR0[NCS] or SSCR0[ECS] bit fields settings are optional */
+ SSCR0_P(port) = SSCR0_EDSS | SSCR0_PSP | SSCR0_DataSize(16);
+
-+ /* SSCR1 = 0x203C3C03 */
-+ /* SSCR1[SCLKDIR] and SSCR1[SFRMDIR] must be cleared (master only ???),
-+ * all other bit fields settings are optional. */
-+ //SSCR1_P(port) &= ~(SSCR1_SCLKDIR | SSCR1_SFRMDIR);
-+
+ /* set FIFO thresholds */
+ SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1);
+
@@ -16943,13 +16972,14 @@
+ /* all bit fields must be cleared except: FSRT = 1 and
+ * SFRMWDTH = 16, DMYSTART=0,1) */
+ SSPSP_P(port) = SSPSP_FSRT | SSPSP_SFRMWDTH(16) | SSPSP_DMYSTRT(0);
-+ return 0;
++ goto master;
+ }
+
+ SSCR0_P(port) |= SSCR0_PSP;
+ SSCR1_P(port) = SSCR1_RxTresh(14) | SSCR1_TxTresh(1) |
+ SSCR1_TRAIL | SSCR1_RWOT;
+
++master:
+ switch(fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ case SND_SOC_DAIFMT_CBM_CFM:
+ SSCR1_P(port) |= (SSCR1_SCLKDIR | SSCR1_SFRMDIR);
@@ -17012,6 +17042,8 @@
+ dma += 2; /* stereo DMA offset is 2, mono is 0 */
+ cpu_dai->dma_data = ssp_dma_params[cpu_dai->id][dma];
+
++ dbg("pxa2xx_ssp_hw_params: dma %d", dma);
++
+ /* we can only change the settings if the port is not in use */
+ if (SSCR0_P(port) & SSCR0_SSE)
+ return 0;
@@ -17034,12 +17066,11 @@
+ break;
+ }
+
-+#if PXA_SSP_DEBUG
-+ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x SSACD %x\n",
++ dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x SSACD 0x%08x",
+ SSCR0_P(port), SSCR1_P(port),
+ SSTO_P(port), SSPSP_P(port),
+ SSSR_P(port), SSACD_P(port));
-+#endif
++
+ return 0;
+}
+
@@ -17087,13 +17118,12 @@
+ default:
+ ret = -EINVAL;
+ }
-+#if PXA_SSP_DEBUG
-+ printk("trig cmd %d\n", cmd);
-+ printk("SSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x\n",
++
++ dbg("SSCR0 0x%08x SSCR1 0x%08x SSTO 0x%08x SSPSP 0x%08x SSSR 0x%08x",
+ SSCR0_P(port), SSCR1_P(port),
+ SSTO_P(port), SSPSP_P(port),
+ SSSR_P(port));
-+#endif
++
+ return ret;
+}
+
@@ -17202,7 +17232,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/spitz.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/spitz.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/spitz.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,394 @@
+/*
+ * spitz.c -- SoC audio for Sharp SL-Cxx00 models Spitz, Borzoi and Akita
@@ -17601,7 +17631,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/tosa.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/tosa.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/tosa.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,289 @@
+/*
+ * tosa.c -- SoC audio for Tosa
@@ -17895,7 +17925,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/soc-dapm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/soc-dapm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/soc-dapm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,1329 @@
+/*
+ * soc-dapm.c -- ALSA SoC Dynamic Audio Power Management
@@ -19229,8 +19259,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/soc-core.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/soc-core.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,1617 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/soc-core.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,1587 @@
+/*
+ * soc-core.c -- ALSA SoC Audio Layer
+ *
@@ -19274,7 +19304,7 @@
+#include <sound/initval.h>
+
+/* debug */
-+#define SOC_DEBUG 1
++#define SOC_DEBUG 0
+#if SOC_DEBUG
+#define dbg(format, arg...) printk(format, ## arg)
+#else
@@ -19285,13 +19315,6 @@
+static DEFINE_MUTEX(io_mutex);
+static DECLARE_WAIT_QUEUE_HEAD(soc_pm_waitq);
+
-+/* supported sample rates */
-+/* ATTENTION: these values depend on the definition in pcm.h! */
-+static const unsigned int rates[] = {
-+ 5512, 8000, 11025, 16000, 22050, 32000, 44100,
-+ 48000, 64000, 88200, 96000, 176400, 192000
-+};
-+
+/*
+ * This is a timeout to do a DAPM powerdown after a stream is closed().
+ * It can be used to eliminate pops between different playback streams, e.g.
@@ -20111,6 +20134,7 @@
+ kfree(rtd);
+ return ret;
+ }
++
+ pcm->private_data = rtd;
+ soc_pcm_ops.mmap = socdev->platform->pcm_ops->mmap;
+ soc_pcm_ops.pointer = socdev->platform->pcm_ops->pointer;
@@ -20269,22 +20293,6 @@
+EXPORT_SYMBOL_GPL(snd_soc_test_bits);
+
+/**
-+ * snd_soc_get_rate - get int sample rate
-+ * @hwpcmrate: the hardware pcm rate
-+ *
-+ * Returns the audio rate integaer value, else 0.
-+ */
-+int snd_soc_get_rate(int hwpcmrate)
-+{
-+ int rate = ffs(hwpcmrate) - 1;
-+
-+ if (rate > ARRAY_SIZE(rates))
-+ return 0;
-+ return rates[rate];
-+}
-+EXPORT_SYMBOL_GPL(snd_soc_get_rate);
-+
-+/**
+ * snd_soc_new_pcms - create new sound card and pcms
+ * @socdev: the SoC audio device
+ *
@@ -20309,15 +20317,7 @@
+ return -ENODEV;
+ }
+
-+ codec->card->dev = device_create(sound_class, codec->card->parent, 0,
-+ "card%i", codec->card->number);
-+ if (IS_ERR(codec->card->dev)) {
-+ snd_card_free(codec->card);
-+ printk(KERN_ERR "asoc: could not create sound dev %s\n",
-+ codec->name);
-+ return -ENODEV;
-+ }
-+
++ codec->card->dev = socdev->dev;
+ codec->card->private_data = codec;
+ strncpy(codec->card->driver, codec->name, sizeof(codec->card->driver));
+
@@ -20851,7 +20851,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/Kconfig 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/Kconfig 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,32 @@
+menu "SoC Audio for the Atmel AT91"
+
@@ -20888,7 +20888,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/Makefile 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/Makefile 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,11 @@
+# AT91 Platform Support
+snd-soc-at91-objs := at91-pcm.o
@@ -20904,7 +20904,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,591 @@
+/*
+ * imx-ssi.c -- SSI driver for Freescale IMX
@@ -21500,7 +21500,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/Kconfig 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/Kconfig 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,31 @@
+menu "SoC Audio for the Freescale i.MX"
+
@@ -21536,7 +21536,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/Makefile 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/Makefile 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,18 @@
+# i.MX Platform Support
+snd-soc-imx21-objs := imx21-pcm.o
@@ -21558,21 +21558,21 @@
+
Index: linux-2.6.17.14-fic4.test/sound/Makefile
===================================================================
---- linux-2.6.17.14-fic4.test.orig/sound/Makefile 2007-01-31 21:34:16.000000000 +0100
-+++ linux-2.6.17.14-fic4.test/sound/Makefile 2007-01-31 21:36:19.000000000 +0100
+--- linux-2.6.17.14-fic4.test.orig/sound/Makefile 2007-02-06 15:38:07.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/Makefile 2007-02-06 15:55:34.000000000 +0100
@@ -4,7 +4,7 @@
obj-$(CONFIG_SOUND) += soundcore.o
obj-$(CONFIG_SOUND_PRIME) += oss/
obj-$(CONFIG_DMASOUND) += oss/
-obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/
-+obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/
++obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ asoc/
ifeq ($(CONFIG_SND),y)
obj-y += last.o
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,715 @@
+/*
+ * wm8711.c -- WM8711 ALSA SoC Audio driver
@@ -21832,7 +21832,7 @@
+ struct snd_soc_codec *codec = socdev->codec;
+ struct wm8711_priv *wm8711 = codec->private_data;
+ u16 iface = wm8711_read_reg_cache(codec, WM8711_IFACE) & 0xfffc;
-+ int i = get_coeff(wm8711->sysclk, snd_soc_get_rate(params_rate(params)));
++ int i = get_coeff(wm8711->sysclk, params_rate(params));
+ u16 srate = (coeff_div[i].sr << 2) |
+ (coeff_div[i].bosr << 1) | coeff_div[i].usb;
+
@@ -22292,7 +22292,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8711.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,42 @@
+/*
+ * wm8711.h -- WM8711 Soc Audio driver
@@ -22339,7 +22339,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,923 @@
+/*
+ * wm8980.c -- WM8980 ALSA Soc Audio driver
@@ -23267,7 +23267,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8980.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,116 @@
+/*
+ * wm8980.h -- WM8980 Soc Audio driver
@@ -23388,7 +23388,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/eti_b1_wm8731.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/eti_b1_wm8731.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/eti_b1_wm8731.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,375 @@
+/*
+ * eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
@@ -23768,7 +23768,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,860 @@
+/*
+ * wm8510.c -- WM8510 ALSA Soc Audio driver
@@ -24633,7 +24633,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8510.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,103 @@
+/*
+ * wm8510.h -- WM8510 Soc Audio driver
@@ -24741,7 +24741,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ac97.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ac97.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ac97.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,222 @@
+/*
+ * imx-ssi.c -- SSI driver for Freescale IMX
@@ -24967,8 +24967,8 @@
+MODULE_LICENSE("GPL");
Index: linux-2.6.17.14-fic4.test/include/linux/i2c-id.h
===================================================================
---- linux-2.6.17.14-fic4.test.orig/include/linux/i2c-id.h 2007-01-31 21:34:15.000000000 +0100
-+++ linux-2.6.17.14-fic4.test/include/linux/i2c-id.h 2007-01-31 21:36:44.000000000 +0100
+--- linux-2.6.17.14-fic4.test.orig/include/linux/i2c-id.h 2007-02-06 15:38:06.000000000 +0100
++++ linux-2.6.17.14-fic4.test/include/linux/i2c-id.h 2007-02-06 15:56:03.000000000 +0100
@@ -112,6 +112,8 @@
#define I2C_DRIVERID_X1205 82 /* Xicor/Intersil X1205 RTC */
#define I2C_DRIVERID_PCF8563 83 /* Philips PCF8563 RTC */
@@ -24981,7 +24981,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,885 @@
+/*
+ * wm8976.c -- WM8976 ALSA Soc Audio driver
@@ -25871,7 +25871,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8976.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,112 @@
+/*
+ * wm8976.h -- WM8976 Soc Audio driver
@@ -25988,7 +25988,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,454 @@
+/*
+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
@@ -26447,7 +26447,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx21-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,237 @@
+/*
+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
@@ -26689,7 +26689,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,417 @@
+/*
+ * linux/sound/arm/mxc-pcm.c -- ALSA SoC interface for the Freescale i.MX CPU's
@@ -27111,7 +27111,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx31-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,241 @@
+/*
+ * mxc-pcm.h :- ASoC platform header for Freescale i.MX
@@ -27357,8 +27357,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/magician.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/magician.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,577 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/magician.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,563 @@
+/*
+ * SoC audio for HTC Magician
+ *
@@ -27433,7 +27433,7 @@
+ snd_soc_dapm_set_endpoint(codec, "Speaker", 0);
+
+ /* set up jack connection */
-+ switch(magician_jack_func) {
++ switch (magician_jack_func) {
+ case MAGICIAN_HP:
+ /* enable and unmute hp jack, disable mic bias */
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
@@ -27450,16 +27450,20 @@
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
+ break;
+ }
-+/*
++#if 0
++ /* fixme pH5, can we detect and config the correct Mic type ? */
+ switch(magician_in_sel) {
+ case MAGICIAN_IN_MIC:
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++ break;
+ case MAGICIAN_IN_MIC_EXT:
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 1);
++ break;
+ case MAGICIAN_IN_BT_HS:
+ snd_soc_dapm_set_endpoint(codec, "Mic Jack", 0);
++ break;
+ }
-+*/
++#endif
+ snd_soc_dapm_sync_endpoints(codec);
+}
+
@@ -27478,7 +27482,7 @@
+ * Magician uses SSP port for playback.
+ */
+static int magician_playback_hw_params(struct snd_pcm_substream *substream,
-+ struct snd_pcm_hw_params *params)
++ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
@@ -27490,15 +27494,15 @@
+ * Rate = SSPSCLK / (word size(16))
+ * SSPSCLK = (ACPS / ACDS) / SSPSCLKDIV(div4 or div1)
+ */
-+ switch(params_rate(params)) {
++ switch (params_rate(params)) {
+ case 8000:
+ acps = 32842000;
-+ acds = PXA2XX_SSP_CLK_AUDIO_DIV_32; /* wrong - 32 bits/sample */
++ acds = PXA2XX_SSP_CLK_AUDIO_DIV_32; /* wrong - 32 bits/sample */
+ div4 = PXA2XX_SSP_CLK_SCDB_4;
+ break;
+ case 11025:
+ acps = 5622000;
-+ acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; /* 16 bits/sample, 1 slot */
++ acds = PXA2XX_SSP_CLK_AUDIO_DIV_8; /* 16 bits/sample, 1 slot */
+ div4 = PXA2XX_SSP_CLK_SCDB_4;
+ break;
+ case 22050:
@@ -27520,29 +27524,31 @@
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
-+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
-+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set audio clock as clock source */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_AUDIO, 0,
-+ SND_SOC_CLOCK_OUT);
++ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock ACDS divider */
-+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
++ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
++ PXA2XX_SSP_AUDIO_DIV_ACDS, acds);
+ if (ret < 0)
+ return ret;
+
+ /* set the SSP audio system clock SCDB divider4 */
-+ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
++ ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai,
++ PXA2XX_SSP_AUDIO_DIV_SCDB, div4);
+ if (ret < 0)
+ return ret;
+
@@ -27551,8 +27557,6 @@
+ if (ret < 0)
+ return ret;
+
-+ /* pH5 -- may need to set other SSP port config */
-+
+ return 0;
+}
+
@@ -27570,13 +27574,7 @@
+ SACR0 |= SACR0_BCKD;
+ SACR0 |= SACR0_ENB;
+ pxa_set_cken(CKEN8_I2S, 1);
-+#if 0
-+ printk ("enabled I2S clock\n");
-+ printk("SSP port enabled now\nSSCR0 %x SSCR1 %x SSTO %x SSPSP %x SSSR %x SSACD %x\n",
-+ SSCR0_P(1), SSCR1_P(1),
-+ SSTO_P(1), SSPSP_P(1),
-+ SSSR_P(1), SSACD_P(1));
-+#endif
++
+ return 0;
+}
+
@@ -27586,7 +27584,6 @@
+ SACR0 &= ~SACR0_ENB;
+ SACR0 &= ~SACR0_BCKD;
+ pxa_set_cken(CKEN8_I2S, 0);
-+ printk ("disabled I2S clock\n");
+ return 0;
+}
+
@@ -27594,7 +27591,7 @@
+ * Magician uses I2S for capture.
+ */
+static int magician_capture_hw_params(struct snd_pcm_substream *substream,
-+ struct snd_pcm_hw_params *params)
++ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
@@ -27603,24 +27600,22 @@
+
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai,
-+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set cpu DAI configuration */
+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
-+ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
++ SND_SOC_DAIFMT_MSB | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+ if (ret < 0)
+ return ret;
+
+ /* set the I2S system clock as output */
+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_I2S_SYSCLK, 0,
-+ SND_SOC_CLOCK_OUT);
++ SND_SOC_CLOCK_OUT);
+ if (ret < 0)
+ return ret;
+
-+ /* pH5 may need to set other I2S config */
-+
+ return 0;
+}
+
@@ -27648,13 +27643,15 @@
+ .hw_free = magician_playback_hw_free,
+};
+
-+static int magician_get_jack(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_get_jack(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_jack_func;
+ return 0;
+}
+
-+static int magician_set_jack(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_set_jack(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
@@ -27666,15 +27663,17 @@
+ return 1;
+}
+
-+static int magician_get_spk(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_get_spk(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_spk_func;
+ return 0;
+}
+
-+static int magician_set_spk(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_set_spk(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
-+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_spk_func == ucontrol->value.integer.value[0])
+ return 0;
@@ -27684,29 +27683,35 @@
+ return 1;
+}
+
-+static int magician_get_input(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_get_input(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
+ ucontrol->value.integer.value[0] = magician_in_sel;
+ return 0;
+}
+
-+static int magician_set_input(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t * ucontrol)
++static int magician_set_input(snd_kcontrol_t * kcontrol,
++ snd_ctl_elem_value_t * ucontrol)
+{
-+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
++ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
+ if (magician_in_sel == ucontrol->value.integer.value[0])
+ return 0;
+
+ magician_in_sel = ucontrol->value.integer.value[0];
-+// magician_ext_control(codec);
-+ switch(magician_in_sel) {
++
++ switch (magician_in_sel) {
+ case MAGICIAN_MIC:
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL0);
-+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL1);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_IN_SEL0);
++ magician_egpio_enable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_IN_SEL1);
+ break;
+ case MAGICIAN_MIC_EXT:
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL0);
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_IN_SEL1);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_IN_SEL0);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_IN_SEL1);
+ }
+
+ return 1;
@@ -27715,27 +27720,33 @@
+static int magician_spk_power(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
-+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
++ magician_egpio_enable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_SPK_POWER);
+ else
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_SPK_POWER);
+ return 0;
+}
+
+static int magician_hp_power(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
-+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++ magician_egpio_enable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_EP_POWER);
+ else
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_EP_POWER);
+ return 0;
+}
+
-+static int magician_mic_bias(struct snd_soc_dapm_widget* w, int event)
++static int magician_mic_bias(struct snd_soc_dapm_widget *w, int event)
+{
+ if (SND_SOC_DAPM_EVENT_ON(event))
-+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
++ magician_egpio_enable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_MIC_POWER);
+ else
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_MIC_POWER);
++ magician_egpio_disable(&magician_cpld,
++ EGPIO_NR_MAGICIAN_MIC_POWER);
+ return 0;
+}
+
@@ -27747,15 +27758,15 @@
+};
+
+/* magician machine audio_map */
-+static const char* audio_map[][3] = {
++static const char *audio_map[][3] = {
+
+ /* headphone connected to VOUTLHP, VOUTRHP */
+ {"Headphone Jack", NULL, "VOUTLHP"},
+ {"Headphone Jack", NULL, "VOUTRHP"},
+
+ /* ext speaker connected to VOUTL, VOUTR */
-+ {"Speaker", NULL , "VOUTL"},
-+ {"Speaker", NULL , "VOUTR"},
++ {"Speaker", NULL, "VOUTL"},
++ {"Speaker", NULL, "VOUTR"},
+
+ /* mic is connected to VINM */
+ {"VINM", NULL, "Mic Jack"},
@@ -27767,9 +27778,9 @@
+ {NULL, NULL, NULL},
+};
+
-+static const char *jack_function[] = {"Off", "Headset", "Headphone"};
-+static const char *spk_function[] = {"On", "Off"};
-+static const char *input_select[] = {"Internal Mic", "External Mic"};
++static const char *jack_function[] = { "Off", "Headset", "Headphone" };
++static const char *spk_function[] = { "On", "Off" };
++static const char *input_select[] = { "Internal Mic", "External Mic" };
+static const struct soc_enum magician_enum[] = {
+ SOC_ENUM_SINGLE_EXT(4, jack_function),
+ SOC_ENUM_SINGLE_EXT(2, spk_function),
@@ -27777,9 +27788,12 @@
+};
+
+static const struct snd_kcontrol_new uda1380_magician_controls[] = {
-+ SOC_ENUM_EXT("Jack Function", magician_enum[0], magician_get_jack, magician_set_jack),
-+ SOC_ENUM_EXT("Speaker Function", magician_enum[1], magician_get_spk, magician_set_spk),
-+ SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input, magician_set_input),
++ SOC_ENUM_EXT("Jack Function", magician_enum[0], magician_get_jack,
++ magician_set_jack),
++ SOC_ENUM_EXT("Speaker Function", magician_enum[1], magician_get_spk,
++ magician_set_spk),
++ SOC_ENUM_EXT("Input Select", magician_enum[2], magician_get_input,
++ magician_set_input),
+};
+
+/*
@@ -27789,8 +27803,6 @@
+{
+ int i, err;
+
-+printk("magician_uda1380_init\n");
-+
+ /* NC codec pins */
+ snd_soc_dapm_set_endpoint(codec, "VOUTLHP", 0);
+ snd_soc_dapm_set_endpoint(codec, "VOUTRHP", 0);
@@ -27798,18 +27810,20 @@
+ /* Add magician specific controls */
+ for (i = 0; i < ARRAY_SIZE(uda1380_magician_controls); i++) {
+ if ((err = snd_ctl_add(codec->card,
-+ snd_soc_cnew(&uda1380_magician_controls[i], codec, NULL))) < 0)
++ snd_soc_cnew(&uda1380_magician_controls[i],
++ codec, NULL))) < 0)
+ return err;
+ }
+
+ /* Add magician specific widgets */
-+ for(i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
++ for (i = 0; i < ARRAY_SIZE(uda1380_dapm_widgets); i++) {
+ snd_soc_dapm_new_control(codec, &uda1380_dapm_widgets[i]);
+ }
+
+ /* Set up magician specific audio path interconnects */
-+ for(i = 0; audio_map[i][0] != NULL; i++) {
-+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ for (i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0],
++ audio_map[i][1], audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
@@ -27858,24 +27872,6 @@
+
+static struct platform_device *magician_snd_device;
+
-+static irqreturn_t magician_ep_plug_isr (int isr, void *data)
-+{
-+ //struct snd_soc_codec *codec = data;
-+ int ep_state = magician_ep_state(&magician_cpld);
-+
-+ printk("earphone: %d\n", ep_state);
-+
-+ if (ep_state) {
-+ //snd_soc_dapm_set_endpoint(codec, "Headphone Jack", 1);
-+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
-+ } else {
-+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
-+ }
-+
-+ return IRQ_HANDLED;
-+}
-+
-+
+static int __init magician_init(void)
+{
+ int ret;
@@ -27883,19 +27879,13 @@
+ if (!machine_is_magician())
+ return -ENODEV;
+
-+printk("snd-soc-magician init\n");
-+
+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_POWER);
+
-+// we probably need to have the clock running here:
-+
++ /* we may need to have the clock running here - pH5 */
+ magician_egpio_enable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
+ udelay(5);
+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_CODEC_RESET);
+
-+ /* correct place? we'll need it to talk to the uda1380 */
-+ request_module("i2c-pxa");
-+
+ magician_snd_device = platform_device_alloc("soc-audio", -1);
+ if (!magician_snd_device)
+ return -ENOMEM;
@@ -27906,9 +27896,6 @@
+
+ if (ret)
+ platform_device_put(magician_snd_device);
-+ request_irq(IRQ_MAGICIAN_EP, magician_ep_plug_isr, IRQF_SAMPLE_RANDOM,
-+ "EP plug", NULL);
-+ /*&magician_snd_devdata->codec_dev->codec->parent(snd_soc_device)->codec*/
+
+ pxa_gpio_mode(GPIO53_SSP1SYSCLK_MD);
+ pxa_gpio_mode(GPIO26_SSP1RX_MD);
@@ -27922,7 +27909,6 @@
+static void __exit magician_exit(void)
+{
+ platform_device_unregister(magician_snd_device);
-+ free_irq(IRQ_MAGICIAN_EP, NULL);
+
+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_SPK_POWER);
+ magician_egpio_disable(&magician_cpld, EGPIO_NR_MAGICIAN_EP_POWER);
@@ -27939,7 +27925,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-ac97.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-ac97.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-ac97.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,270 @@
+/*
+ * ltemplate-ac97.c -- AC97 support for the xxx chip.
@@ -28214,7 +28200,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,784 @@
+/*
+ * template-codec.c -- Template Codec Audio driver
@@ -29003,7 +28989,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-i2s.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-i2s.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-i2s.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,223 @@
+/*
+ * This program is free software; you can redistribute it and/or modify it
@@ -29231,7 +29217,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,166 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
@@ -29402,7 +29388,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,19 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
@@ -29426,7 +29412,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-codec.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,21 @@
+/*
+ * This program is free software; you can redistribute it and/or modify
@@ -29452,7 +29438,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/templates/template-machine.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-machine.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/templates/template-machine.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,161 @@
+/*
+ * This program is free software; you can redistribute it and/or modify it
@@ -29618,7 +29604,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,720 @@
+/*
+ * at91-i2s.c -- ALSA SoC I2S Audio Layer Platform driver
@@ -30343,7 +30329,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,432 @@
+/*
+ * at91-pcm.c -- ALSA PCM interface for the Atmel AT91 SoC
@@ -30780,7 +30766,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,72 @@
+/*
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
@@ -30857,8 +30843,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.h 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,42 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ssp.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,43 @@
+/*
+ * linux/sound/arm/pxa2xx-ssp.h
+ *
@@ -30880,6 +30866,7 @@
+#define PXA2XX_SSP_CLK_EXT 1
+#define PXA2XX_SSP_CLK_NET 2
+#define PXA2XX_SSP_CLK_AUDIO 3
++#define PXA2XX_SSP_CLK_NET_PLL 4
+
+/* SSP audio dividers */
+#define PXA2XX_SSP_AUDIO_DIV_ACDS 0
@@ -30904,7 +30891,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-ac97.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,22 @@
+/*
+ * linux/sound/arm/pxa2xx-ac97.h
@@ -30931,7 +30918,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/pxa2xx-i2s.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,20 @@
+/*
+ * linux/sound/arm/pxa2xx-i2s.h
@@ -30956,8 +30943,8 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Kconfig
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Kconfig 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,30 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Kconfig 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,31 @@
+menu "SoC Audio for the Samsung S3C24XX"
+
+config SND_S3C24XX_SOC
@@ -30979,10 +30966,11 @@
+ help
+ Say Y if you want to add support for SoC audio on SMDK2440
+
-+config SND_S3C24XX_SOC_SMDK2440_WM8753
-+ tristate "SoC I2S Audio support for S3C24XX - WM8753"
-+ depends on SND_S3C24XX_SOC && MACH_SMDK
++config SND_S3C24XX_SOC_NEO1973_WM8753
++ tristate "SoC I2S Audio support for NEO1973 - WM8753"
++ depends on SND_S3C24XX_SOC && MACH_GTA01
+ select SND_S3C24XX_SOC_I2S
++ select SND_SOC_WM8753
+ help
+ Say Y if you want to add support for SoC audio on smdk2440
+ with the WM8753.
@@ -30991,7 +30979,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Makefile
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Makefile 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/Makefile 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,13 @@
+# S3c24XX Platform Support
+snd-soc-s3c24xx-objs := s3c24xx-pcm.o
@@ -31002,15 +30990,15 @@
+
+# S3C24XX Machine Support
+snd-soc-smdk2440-objs := smdk2440.o
-+snd-soc-smdk2440-wm8753-objs := smdk2440_wm8753.o
++snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
+
+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440) += snd-soc-smdk2440.o
-+obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2440_WM8753) += snd-soc-smdk2440-wm8753.o
++obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,484 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,439 @@
+/*
+ * s3c24xx-i2s.c -- ALSA Soc Audio Layer
+ *
@@ -31071,13 +31059,13 @@
+ .name = "I2S PCM Stereo in"
+};
+
-+static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_out = {
++static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_out = {
+ .client = &s3c24xx_dma_client_out,
+ .channel = DMACH_I2S_OUT,
+ .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO
+};
+
-+static s3c24xx_pcm_dma_params_t s3c24xx_i2s_pcm_stereo_in = {
++static struct s3c24xx_pcm_dma_params s3c24xx_i2s_pcm_stereo_in = {
+ .client = &s3c24xx_dma_client_in,
+ .channel = DMACH_I2S_IN,
+ .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO
@@ -31085,15 +31073,15 @@
+
+struct s3c24xx_i2s_info {
+ void __iomem *regs;
-+ struct clk *iis_clk;
++ struct clk *iis_clk;
+};
+static struct s3c24xx_i2s_info s3c24xx_i2s;
+
+static void s3c24xx_snd_txctrl(int on)
+{
-+ unsigned long iisfcon;
-+ unsigned long iiscon;
-+ unsigned long iismod;
++ u32 iisfcon;
++ u32 iiscon;
++ u32 iismod;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31118,7 +31106,7 @@
+ * Samsung supplied kernel, this should allow the DMA
+ * engine and FIFOs to reset. If this isn't allowed, the
+ * DMA engine will simply freeze randomly.
-+ */
++ */
+
+ iisfcon &= ~S3C2410_IISFCON_TXENABLE;
+ iisfcon &= ~S3C2410_IISFCON_TXDMA;
@@ -31132,14 +31120,13 @@
+ }
+
+ DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+
+}
+
+static void s3c24xx_snd_rxctrl(int on)
+{
-+ unsigned long iisfcon;
-+ unsigned long iiscon;
-+ unsigned long iismod;
++ u32 iisfcon;
++ u32 iiscon;
++ u32 iismod;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31159,12 +31146,12 @@
+ writel(iisfcon, s3c24xx_i2s.regs + S3C2410_IISFCON);
+ writel(iiscon, s3c24xx_i2s.regs + S3C2410_IISCON);
+ } else {
-+ /* note, we have to disable the FIFOs otherwise bad things
-+ * seem to happen when the DMA stops. According to the
-+ * Samsung supplied kernel, this should allow the DMA
-+ * engine and FIFOs to reset. If this isn't allowed, the
-+ * DMA engine will simply freeze randomly.
-+ */
++ /* note, we have to disable the FIFOs otherwise bad things
++ * seem to happen when the DMA stops. According to the
++ * Samsung supplied kernel, this should allow the DMA
++ * engine and FIFOs to reset. If this isn't allowed, the
++ * DMA engine will simply freeze randomly.
++ */
+
+ iisfcon &= ~S3C2410_IISFCON_RXENABLE;
+ iisfcon &= ~S3C2410_IISFCON_RXDMA;
@@ -31178,19 +31165,16 @@
+ }
+
+ DBG("w: IISCON: %lx IISMOD: %lx IISFCON: %lx\n", iiscon, iismod, iisfcon);
-+
+}
+
-+/* s3c24xx_snd_lrsync
-+ *
++/*
+ * Wait for the LR signal to allow synchronisation to the L/R clock
-+ * from the DAC. May only be needed for slave mode.
-+*/
-+
-+static int s3c24xx_snd_lrsync()
++ * from the codec. May only be needed for slave mode.
++ */
++static int s3c24xx_snd_lrsync(void)
+{
-+ unsigned long iiscon;
-+ int timeout = 10000;
++ u32 iiscon;
++ long timeout = jiffies + msecs_to_jiffies(5);
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31199,39 +31183,30 @@
+ if (iiscon & S3C2410_IISCON_LRINDEX)
+ break;
+
-+ if (--timeout < 0)
++ if (timeout < jiffies)
+ return -ETIMEDOUT;
+ }
+
+ return 0;
+}
+
-+/* s3c24xx_snd_is_clkmaster()
-+ *
++/*
+ * Check whether CPU is the master or slave
-+*/
-+
-+static inline int s3c24xx_snd_is_clkmaster()
++ */
++static inline int s3c24xx_snd_is_clkmaster(void)
+{
+ DBG("Entered %s\n", __FUNCTION__);
+
+ return (readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & S3C2410_IISMOD_SLAVE) ? 0:1;
+}
+
-+static int s3c24xx_i2s_startup(struct snd_pcm_substream *substream)
-+{
-+ DBG("Entered %s\n", __FUNCTION__);
-+
-+ return 0;
-+}
-+
+/*
+ * Set S3C24xx I2S DAI format
+ */
+static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai,
+ unsigned int fmt)
+{
-+ unsigned long iismod;
++ u32 iismod;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31267,7 +31242,7 @@
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+ unsigned long iismod;
++ u32 iismod;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31296,7 +31271,6 @@
+static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
-+ unsigned long flags;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31325,43 +31299,20 @@
+ break;
+ default:
+ ret = -EINVAL;
++ break;
+ }
+
+exit_err:
+ return ret;
+}
+
-+static void s3c24xx_i2s_shutdown(struct snd_pcm_substream *substream)
-+{
-+ unsigned long iismod, iiscon, iisfcon;
-+
-+ DBG("Entered %s\n", __FUNCTION__);
-+
-+}
-+
-+#ifdef CONFIG_PM
-+static int s3c24xx_i2s_suspend(struct platform_device *dev,
-+ struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+static int s3c24xx_i2s_resume(struct platform_device *pdev,
-+ struct snd_soc_cpu_dai *dai)
-+{
-+}
-+
-+#else
-+#define s3c24xx_i2s_suspend NULL
-+#define s3c24xx_i2s_resume NULL
-+#endif
-+
+/*
+ * Set S3C24xx Clock source
+ */
+static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai,
+ int clk_id, unsigned int freq, int dir)
+{
-+ unsigned long iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
++ u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD);
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31387,7 +31338,7 @@
+static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai,
+ int div_id, int div)
+{
-+ unsigned long reg;
++ u32 reg;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31412,20 +31363,17 @@
+ return 0;
+}
+
-+
-+/* To avoid duplicating clock code, allow machine driver to
++/*
++ * To avoid duplicating clock code, allow machine driver to
+ * get the clockrate from here.
+ */
-+
-+unsigned long s3c24xx_i2s_get_clockrate(void)
++u32 s3c24xx_i2s_get_clockrate(void)
+{
+ return clk_get_rate(s3c24xx_i2s.iis_clk);
+}
-+
+EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate);
+
-+static unsigned int s3c24xx_i2s_probe(struct platform_device *pdev,
-+ struct snd_soc_cpu_dai *dai)
++static int s3c24xx_i2s_probe(struct platform_device *pdev)
+{
+ DBG("Entered %s\n", __FUNCTION__);
+
@@ -31433,7 +31381,7 @@
+ if (s3c24xx_i2s.regs == NULL)
+ return -ENXIO;
+
-+ s3c24xx_i2s.iis_clk=clk_get(pdev, "iis");
++ s3c24xx_i2s.iis_clk=clk_get(&pdev->dev, "iis");
+ if (s3c24xx_i2s.iis_clk == NULL) {
+ DBG("failed to get iis_clock\n");
+ return -ENODEV;
@@ -31465,8 +31413,6 @@
+ .id = 0,
+ .type = SND_SOC_DAI_I2S,
+ .probe = s3c24xx_i2s_probe,
-+ .suspend = s3c24xx_i2s_suspend,
-+ .resume = s3c24xx_i2s_resume,
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 2,
@@ -31478,8 +31424,6 @@
+ .rates = S3C24XX_I2S_RATES,
+ .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE,},
+ .ops = {
-+ .startup = s3c24xx_i2s_startup,
-+ .shutdown = s3c24xx_i2s_shutdown,
+ .trigger = s3c24xx_i2s_trigger,
+ .hw_params = s3c24xx_i2s_hw_params,},
+ .dai_ops = {
@@ -31488,7 +31432,6 @@
+ .set_sysclk = s3c24xx_i2s_set_sysclk,
+ },
+};
-+
+EXPORT_SYMBOL_GPL(s3c24xx_i2s_dai);
+
+/* Module information */
@@ -31498,7 +31441,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-i2s.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,35 @@
+/*
+ * s3c24xx-i2s.c -- ALSA Soc Audio Layer
@@ -31532,14 +31475,14 @@
+#define S3C24XX_PRESCALE(a,b) \
+ (((a - 1) << S3C2410_IISPSR_INTSHIFT) | ((b - 1) << S3C2410_IISPSR_EXTSHFIT))
+
-+unsigned long s3c24xx_i2s_get_clockrate(void);
++u32 s3c24xx_i2s_get_clockrate(void);
+
+#endif /*S3C24XXI2S_H_*/
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,484 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,467 @@
+/*
+ * s3c24xx-pcm.c -- ALSA Soc Audio Layer
+ *
@@ -31588,18 +31531,14 @@
+#endif
+
+static const struct snd_pcm_hardware s3c24xx_pcm_hardware = {
-+ .info = ( SNDRV_PCM_INFO_INTERLEAVED |
++ .info = SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP |
-+ SNDRV_PCM_INFO_MMAP_VALID |
-+ 0),
-+ .formats = ( SNDRV_PCM_FMTBIT_S16_LE |
++ SNDRV_PCM_INFO_MMAP_VALID,
++ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_U16_LE |
+ SNDRV_PCM_FMTBIT_U8 |
-+ SNDRV_PCM_FMTBIT_S8 ),
-+ .rates = SNDRV_PCM_RATE_44100,
-+ .rate_min = 8000,
-+ .rate_max = 48000,
++ SNDRV_PCM_FMTBIT_S8,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 128*1024,
@@ -31619,7 +31558,7 @@
+ dma_addr_t dma_start;
+ dma_addr_t dma_pos;
+ dma_addr_t dma_end;
-+ s3c24xx_pcm_dma_params_t *params;
++ struct s3c24xx_pcm_dma_params *params;
+};
+
+/* s3c24xx_pcm_enqueue
@@ -31627,7 +31566,6 @@
+ * place a dma buffer onto the queue for the dma system
+ * to handle.
+*/
-+
+static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
@@ -31661,7 +31599,7 @@
+ prtd->dma_pos = pos;
+}
+
-+void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
++static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,
+ void *dev_id, int size,
+ enum s3c2410_dma_buffresult result)
+{
@@ -31691,13 +31629,17 @@
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+ s3c24xx_pcm_dma_params_t *dma = rtd->dai->cpu_dai->dma_data;
++ struct s3c24xx_pcm_dma_params *dma = rtd->dai->cpu_dai->dma_data;
+ unsigned long totbytes = params_buffer_bytes(params);
-+ unsigned long flags;
+ int ret=0;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
++ /* return if this is a bufferless transfer e.g.
++ * codec <--> BT codec or GSM modem -- lg FIXME */
++ if (!dma)
++ return 0;
++
+ /* prepare DMA */
+ prtd->params = dma;
+
@@ -31714,7 +31656,6 @@
+
+ /* channel needs configuring for mem=>device, increment memory addr,
+ * sync to pclk, half-word transfers to the IIS-FIFO. */
-+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ s3c2410_dma_devconfig(prtd->params->channel,
+ S3C2410_DMASRC_MEM, S3C2410_DISRCC_INC |
@@ -31738,14 +31679,14 @@
+
+ runtime->dma_bytes = totbytes;
+
-+ spin_lock_irqsave(&prtd->lock, flags);
++ spin_lock_irq(&prtd->lock);
+ prtd->dma_loaded = 0;
+ prtd->dma_limit = runtime->hw.periods_min;
+ prtd->dma_period = params_period_bytes(params);
+ prtd->dma_start = runtime->dma_addr;
+ prtd->dma_pos = prtd->dma_start;
+ prtd->dma_end = prtd->dma_start + totbytes;
-+ spin_unlock_irqrestore(&prtd->lock, flags);
++ spin_unlock_irq(&prtd->lock);
+
+ return 0;
+}
@@ -31756,7 +31697,7 @@
+
+ DBG("Entered %s\n", __FUNCTION__);
+
-+ // TODO - do we need to ensure DMA flushed
++ /* TODO - do we need to ensure DMA flushed */
+ snd_pcm_set_runtime_buffer(substream, NULL);
+
+ if(prtd->params) {
@@ -31764,18 +31705,22 @@
+ prtd->params = NULL;
+ }
+
++ return 0;
+}
+
+static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
-+ struct snd_pcm_runtime *runtime = substream->runtime;
+ int ret = 0;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
++ /* return if this is a bufferless transfer e.g.
++ * codec <--> BT codec or GSM modem -- lg FIXME */
++ if (!prtd->params)
++ return 0;
++
+ /* flush the DMA channel */
-+
+ s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_FLUSH);
+ prtd->dma_loaded = 0;
+ prtd->dma_pos = prtd->dma_start;
@@ -31790,11 +31735,10 @@
+{
+ struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;
+ int ret = 0;
-+ unsigned long flags;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
-+ spin_lock_irqsave(&prtd->lock, flags);
++ spin_lock(&prtd->lock);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
@@ -31814,9 +31758,10 @@
+
+ default:
+ ret = -EINVAL;
++ break;
+ }
+
-+ spin_unlock_irqrestore(&prtd->lock, flags);
++ spin_unlock(&prtd->lock);
+
+ return ret;
+}
@@ -31825,14 +31770,12 @@
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct s3c24xx_runtime_data *prtd = runtime->private_data;
-+ unsigned long flags;
+ unsigned long res;
+ dma_addr_t src, dst;
-+ snd_pcm_uframes_t ret;
+
+ DBG("Entered %s\n", __FUNCTION__);
+
-+ spin_lock_irqsave(&prtd->lock, flags);
++ spin_lock(&prtd->lock);
+ s3c2410_dma_getposition(prtd->params->channel, &src, &dst);
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
@@ -31840,7 +31783,7 @@
+ else
+ res = src - prtd->dma_start;
+
-+ spin_unlock_irqrestore(&prtd->lock, flags);
++ spin_unlock(&prtd->lock);
+
+ DBG("Pointer %x %x\n",src,dst);
+
@@ -31850,28 +31793,12 @@
+ * callled... (todo - fix )
+ */
+
-+ if (res > (prtd->dma_end - prtd->dma_start))
-+ // DBG("%s: %lx,%lx (%lx) from %lx,%lx\n", __FUNCTION__,
-+ // (long)src, (long)dst, res,
-+ // (long)prtd->dma_start, (long)prtd->dma_end);
-+
+ if (res >= snd_pcm_lib_buffer_bytes(substream)) {
-+ // DBG("%s: %lx,%lx (%lx) from %lx,%lx\n", __FUNCTION__,
-+ // (long)src, (long)dst, res,
-+ // (long)prtd->dma_start, (long)prtd->dma_end);
-+
-+ // if (res > snd_pcm_lib_buffer_bytes(substream))
-+ // DBG("%s: res %lx >= %lx\n", __FUNCTION__,
-+ // res, (long)snd_pcm_lib_buffer_bytes(substream));
-+
+ if (res == snd_pcm_lib_buffer_bytes(substream))
+ res = 0;
+ }
+
-+ ret = bytes_to_frames(substream->runtime, res);
-+
-+ return ret;
-+
++ return bytes_to_frames(substream->runtime, res);;
+}
+
+static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
@@ -31885,8 +31812,8 @@
+
+ snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
+
-+ if((prtd = kzalloc(sizeof(struct s3c24xx_runtime_data),
-+ GFP_KERNEL)) == NULL) {
++ prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
++ if(prtd == NULL) {
+ ret = -ENOMEM;
+ goto out;
+ }
@@ -31907,15 +31834,14 @@
+
+ if(prtd)
+ kfree(prtd);
-+ else {
++ else
+ DBG("s3c24xx_pcm_close called with prtd == NULL\n");
-+ }
+
+ return 0;
+}
+
-+static int
-+s3c24xx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma)
++static int s3c24xx_pcm_mmap(struct snd_pcm_substream *substream,
++ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
@@ -31927,7 +31853,7 @@
+ runtime->dma_bytes);
+}
+
-+struct snd_pcm_ops s3c24xx_pcm_ops = {
++static struct snd_pcm_ops s3c24xx_pcm_ops = {
+ .open = s3c24xx_pcm_open,
+ .close = s3c24xx_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
@@ -31981,9 +31907,9 @@
+ }
+}
+
-+static u64 s3c24xx_pcm_dmamask = 0xffffffff;
++static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK;
+
-+int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
++static int s3c24xx_pcm_new(struct snd_card *card, struct snd_soc_codec_dai *dai,
+ struct snd_pcm *pcm)
+{
+ int ret = 0;
@@ -32027,31 +31953,31 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/s3c24xx-pcm.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,33 @@
+/*
-+ * linux/sound/arm/pxa2xx-pcm.h -- ALSA PCM interface for the Intel PXA2xx chip
++ * s3c24xx-pcm.h --
+ *
-+ * Author: Nicolas Pitre
-+ * Created: Nov 30, 2004
-+ * Copyright: MontaVista Software, Inc.
++ * 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 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.
++ *
++ * ALSA PCM interface for the Samsung S3C24xx CPU
+ */
+
-+#ifndef _PXA2XX_PCM_H
-+#define _PXA2XX_PCM_H
++#ifndef _S3C24XX_PCM_H
++#define _S3C24XX_PCM_H
+
+#define ST_RUNNING (1<<0)
+#define ST_OPENED (1<<1)
+
-+typedef struct {
++struct s3c24xx_pcm_dma_params {
+ struct s3c2410_dma_client *client; /* stream identifier */
+ int channel; /* Channel ID */
+ dma_addr_t dma_addr;
-+} s3c24xx_pcm_dma_params_t;
++};
+
+#define S3C24XX_DAI_I2S 0
+
@@ -32065,7 +31991,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,320 @@
+/*
+ * smdk2440.c -- ALSA Soc Audio Layer
@@ -32390,7 +32316,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,724 @@
+/*
+ * wm8956.c -- WM8956 ALSA SoC Audio driver
@@ -33119,7 +33045,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8956.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,116 @@
+/*
+ * wm8956.h -- WM8956 Soc Audio driver
@@ -33240,7 +33166,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,766 @@
+/*
+ * wm8960.c -- WM8960 ALSA SoC Audio driver
@@ -34011,7 +33937,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.h 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/wm8960.h 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,121 @@
+/*
+ * wm8960.h -- WM8960 Soc Audio driver
@@ -34137,7 +34063,7 @@
Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440_wm8956.c
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440_wm8956.c 2007-01-31 21:36:05.000000000 +0100
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440_wm8956.c 2007-02-06 15:55:20.000000000 +0100
@@ -0,0 +1,335 @@
+/*
+ * smdk2440.c -- ALSA Soc Audio Layer
@@ -34474,15 +34400,1451 @@
+MODULE_AUTHOR("Ben Dooks, <ben at simtec.co.uk>");
+MODULE_DESCRIPTION("ALSA SoC SMDK2440");
+MODULE_LICENSE("GPL");
-Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440_wm8753.c
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.h
===================================================================
--- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/smdk2440_wm8753.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,721 @@
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,28 @@
+/*
-+ * smdk2440.c -- SoC audio for smdk2440
++ * 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.
++ */
++
++#ifndef _IMX_SSI_H
++#define _IMX_SSI_H
++
++/* pxa2xx DAI SSP ID's */
++#define IMX_DAI_SSI1 0
++#define IMX_DAI_SSI2 1
++
++/* SSI clock sources */
++#define IMX_SSP_SYS_CLK 0
++
++/* SSI audio dividers */
++#define IMX_SSI_DIV_2 0
++#define IMX_SSI_DIV_PSR 1
++#define IMX_SSI_DIV_PM 2
++
++/* SSI Div 2 */
++#define IMX_SSI_DIV_2_OFF ~SSI_STCCR_DIV2
++#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2
++
++extern struct snd_soc_cpu_dai imx_ssi_pcm_dai[2];
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.h
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,27 @@
++/*
++ * at91-i2s.h - ALSA I2S interface for the Atmel AT91 SoC
+ *
++ * Author: Frank Mandarino <fmandarino at endrelia.com>
++ * Endrelia Technologies Inc.
++ * Created: Jan 9, 2007
++ *
++ * 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.
++ */
++
++#ifndef _AT91_I2S_H
++#define _AT91_I2S_H
++
++/* I2S system clock ids */
++#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
++
++/* I2S divider ids */
++#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */
++#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
++#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
++
++extern struct snd_soc_cpu_dai at91_i2s_dai[];
++
++#endif /* _AT91_I2S_H */
++
+Index: linux-2.6.17.14-fic4.test/sound/soc/imx/mx31ads_wm8753.c
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/imx/mx31ads_wm8753.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,400 @@
++/*
++ * mx31ads_wm8753.c -- SoC audio for mx31ads
++ *
+ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Author: Liam Girdwood
++ * liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
++ *
++ * mx31ads audio amplifier code taken from arch/arm/mach-pxa/mx31ads.c
++ * Copyright: MontaVista Software Inc.
++ *
++ * 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.
++ *
++ * Revision history
++ * 30th Oct 2005 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/hardware.h>
++
++#include "../codecs/wm8753.h"
++#include "imx31-pcm.h"
++#include "imx-ssi.h"
++
++static struct snd_soc_machine mx31ads;
++
++static int mx31ads_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai;
++ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++ unsigned int pll_out = 0, bclk = 0, fmt = 0;
++ int ret = 0;
++
++ /*
++ * The WM8753 is better at generating accurate audio clocks than the
++ * MX31 SSI controller, so we will use it as master when we can.
++ */
++ switch (params_rate(params)) {
++ case 8000:
++ case 16000:
++ fmt = SND_SOC_DAIFMT_CBS_CFS;
++ pll_out = 12288000;
++ break;
++ case 48000:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_4;
++ pll_out = 12288000;
++ break;
++ case 96000:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_2;
++ pll_out = 12288000;
++ break;
++ case 11025:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_16;
++ pll_out = 11289600;
++ break;
++ case 22050:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_8;
++ pll_out = 11289600;
++ break;
++ case 44100:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_4;
++ pll_out = 11289600;
++ break;
++ case 88200:
++ fmt = SND_SOC_DAIFMT_CBM_CFS;
++ bclk = WM8753_BCLK_DIV_2;
++ pll_out = 11289600;
++ break;
++ }
++
++ /* set codec DAI configuration */
++ ret = codec_dai->dai_ops.set_fmt(codec_dai,
++ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
++ if (ret < 0)
++ return ret;
++
++ /* set cpu DAI configuration */
++ ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
++ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
++ if (ret < 0)
++ return ret;
++
++ /* set the codec system clock for DAC and ADC */
++ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
++ SND_SOC_CLOCK_IN);
++ if (ret < 0)
++ return ret;
++
++ /* set the SSI system clock as input (unused) */
++ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
++ SND_SOC_CLOCK_IN);
++ if (ret < 0)
++ return ret;
++
++ /* set codec BCLK division for sample rate */
++ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
++ if (ret < 0)
++ return ret;
++
++ /* codec PLL input is 13 MHz */
++ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 13000000, pll_out);
++ if (ret < 0)
++ return ret;
++
++ return 0;
++}
++
++static int mx31ads_hifi_hw_free(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++ /* disable the PLL */
++ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
++}
++
++/*
++ * mx31ads WM8753 HiFi DAI opserations.
++ */
++static struct snd_soc_ops mx31ads_hifi_ops = {
++ .hw_params = mx31ads_hifi_hw_params,
++ .hw_free = mx31ads_hifi_hw_free,
++};
++
++static int mx31ads_voice_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static void mx31ads_voice_shutdown(struct snd_pcm_substream *substream)
++{
++}
++
++static int mx31ads_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
++ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++ unsigned int pll_out = 0, bclk = 0, pcmdiv = 0;
++ int ret = 0;
++
++ /*
++ * The WM8753 is far better at generating accurate audio clocks than the
++ * pxa2xx SSP controller, so we will use it as master when we can.
++ */
++ switch (params_rate(params)) {
++ case 8000:
++ pll_out = 12288000;
++ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 256kHz */
++ break;
++ case 16000:
++ pll_out = 12288000;
++ pcmdiv = WM8753_PCM_DIV_3; /* 4.096 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 512kHz */
++ break;
++ case 48000:
++ pll_out = 12288000;
++ pcmdiv = WM8753_PCM_DIV_1; /* 12.288 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 1.536 MHz */
++ break;
++ case 11025:
++ pll_out = 11289600;
++ pcmdiv = WM8753_PCM_DIV_4; /* 11.2896 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 352.8 kHz */
++ break;
++ case 22050:
++ pll_out = 11289600;
++ pcmdiv = WM8753_PCM_DIV_2; /* 11.2896 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 705.6 kHz */
++ break;
++ case 44100:
++ pll_out = 11289600;
++ pcmdiv = WM8753_PCM_DIV_1; /* 11.2896 MHz */
++ bclk = WM8753_VXCLK_DIV_8; /* 1.4112 MHz */
++ break;
++ }
++
++ /* set codec DAI configuration */
++ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++ if (ret < 0)
++ return ret;
++
++ /* set cpu DAI configuration */
++ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++ if (ret < 0)
++ return ret;
++
++ /* set the codec system clock for DAC and ADC */
++ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, pll_out,
++ SND_SOC_CLOCK_IN);
++ if (ret < 0)
++ return ret;
++
++ /* set the SSP system clock as input (unused) */
++// ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
++// SND_SOC_CLOCK_IN);
++// if (ret < 0)
++// return ret;
++
++ /* set codec BCLK division for sample rate */
++ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_VXCLKDIV, bclk);
++ if (ret < 0)
++ return ret;
++
++ /* set codec PCM division for sample rate */
++ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
++ if (ret < 0)
++ return ret;
++
++ /* codec PLL input is 13 MHz */
++ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 13000000, pll_out);
++ if (ret < 0)
++ return ret;
++
++ return 0;
++}
++
++static int mx31ads_voice_hw_free(struct snd_pcm_substream *substream)
++{
++ struct snd_soc_pcm_runtime *rtd = substream->private_data;
++ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
++
++ /* disable the PLL */
++ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
++}
++
++static struct snd_soc_ops mx31ads_voice_ops = {
++ .startup = mx31ads_voice_startup,
++ .shutdown = mx31ads_voice_shutdown,
++ .hw_params = mx31ads_voice_hw_params,
++ .hw_free = mx31ads_voice_hw_free,
++};
++
++static int mx31ads_suspend(struct platform_device *pdev, pm_message_t state)
++{
++ return 0;
++}
++
++static int mx31ads_resume(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mx31ads_probe(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int mx31ads_remove(struct platform_device *pdev)
++{
++ return 0;
++}
++
++/* example machine audio_mapnections */
++static const char* audio_map[][3] = {
++
++ /* mic is connected to mic1 - with bias */
++ {"MIC1", NULL, "Mic Bias"},
++ {"MIC1N", NULL, "Mic Bias"},
++ {"Mic Bias", NULL, "Mic1 Jack"},
++ {"Mic Bias", NULL, "Mic1 Jack"},
++
++ {"ACIN", NULL, "ACOP"},
++ {NULL, NULL, NULL},
++};
++
++/* headphone detect support on my board */
++static const char * hp_pol[] = {"Headphone", "Speaker"};
++static const struct soc_enum wm8753_enum =
++ SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol);
++
++static const struct snd_kcontrol_new wm8753_mx31ads_controls[] = {
++ SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0),
++ SOC_ENUM("Headphone Detect Polarity", wm8753_enum),
++};
++
++/*
++ * This is an example machine initialisation for a wm8753 connected to a
++ * mx31ads II. It is missing logic to detect hp/mic insertions and logic
++ * to re-route the audio in such an event.
++ */
++static int mx31ads_wm8753_init(struct snd_soc_codec *codec)
++{
++ int i, err;
++
++ /* set up mx31ads codec pins */
++ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
++ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
++ snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
++
++ /* add mx31ads specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8753_mx31ads_controls); i++) {
++ if ((err = snd_ctl_add(codec->card,
++ snd_soc_cnew(&wm8753_mx31ads_controls[i],codec, NULL))) < 0)
++ return err;
++ }
++
++ /* set up mx31ads specific audio path audio_mapnects */
++ for(i = 0; audio_map[i][0] != NULL; i++) {
++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++ return 0;
++}
++
++static struct snd_soc_dai_link mx31ads_dai[] = {
++{ /* Hifi Playback - for similatious use with voice below */
++ .name = "WM8753",
++ .stream_name = "WM8753 HiFi",
++ .cpu_dai = &imx_ssi_pcm_dai[0],
++ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
++ .init = mx31ads_wm8753_init,
++ .ops = &mx31ads_hifi_ops,
++},
++//{ /* Voice via BT */
++// .name = "Bluetooth",
++// .stream_name = "Voice",
++// .cpu_dai = &pxa_ssp_dai[1],
++// .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
++// .ops = &mx31ads_voice_ops,
++//},
++};
++
++static struct snd_soc_machine mx31ads = {
++ .name = "mx31ads",
++ .probe = mx31ads_probe,
++ .remove = mx31ads_remove,
++ .suspend_pre = mx31ads_suspend,
++ .resume_post = mx31ads_resume,
++ .dai_link = mx31ads_dai,
++ .num_links = ARRAY_SIZE(mx31ads_dai),
++};
++
++static struct wm8753_setup_data mx31ads_wm8753_setup = {
++ .i2c_address = 0x1a,
++};
++
++static struct snd_soc_device mx31ads_snd_devdata = {
++ .machine = &mx31ads,
++ .platform = &mxc_soc_platform,
++ .codec_dev = &soc_codec_dev_wm8753,
++ .codec_data = &mx31ads_wm8753_setup,
++};
++
++static struct platform_device *mx31ads_snd_device;
++
++static int __init mx31ads_init(void)
++{
++ int ret;
++
++ mx31ads_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!mx31ads_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(mx31ads_snd_device, &mx31ads_snd_devdata);
++ mx31ads_snd_devdata.dev = &mx31ads_snd_device->dev;
++ ret = platform_device_add(mx31ads_snd_device);
++
++ if (ret)
++ platform_device_put(mx31ads_snd_device);
++
++ return ret;
++}
++
++static void __exit mx31ads_exit(void)
++{
++ platform_device_unregister(mx31ads_snd_device);
++}
++
++module_init(mx31ads_init);
++module_exit(mx31ads_exit);
++
++/* Module information */
++MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
++MODULE_DESCRIPTION("ALSA SoC WM8753 mx31ads");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/lm4857.h
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/lm4857.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,15 @@
++#ifndef LM4857_H_
++#define LM4857_H_
++
++/* The register offsets in the cache array */
++#define LM4857_MVOL 0
++#define LM4857_LVOL 1
++#define LM4857_RVOL 2
++#define LM4857_CTRL 3
++
++/* the shifts required to set these bits */
++#define LM4857_3D 5
++#define LM4857_WAKEUP 5
++#define LM4857_EPGAIN 4
++
++#endif /*LM4857_H_*/
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/tlv320.c
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/tlv320.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,609 @@
++/*
++ * tlv320.c -- TLV 320 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2006 Atlab srl.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ * Nicola Perrino <nicola.perrino at atlab.it>
++ *
++ * 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.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/version.h>
++#include <linux/kernel.h>
++#include <linux/init.h>
++#include <linux/delay.h>
++#include <linux/pm.h>
++#include <linux/i2c.h>
++#include <linux/platform_device.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/pcm_params.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++#include <sound/initval.h>
++
++#include "tlv320.h"
++
++#define AUDIO_NAME "tlv320"
++#define TLV320_VERSION "0.1"
++
++/*
++ * Debug
++ */
++
++//#define TLV320_DEBUG 0
++
++#ifdef TLV320_DEBUG
++#define dbg(format, arg...) \
++ printk(KERN_DEBUG AUDIO_NAME ": " format "\n" , ## arg)
++#else
++#define dbg(format, arg...) do {} while (0)
++#endif
++#define err(format, arg...) \
++ printk(KERN_ERR AUDIO_NAME ": " format "\n" , ## arg)
++#define info(format, arg...) \
++ printk(KERN_INFO AUDIO_NAME ": " format "\n" , ## arg)
++#define warn(format, arg...) \
++ printk(KERN_WARNING AUDIO_NAME ": " format "\n" , ## arg)
++
++
++#define TLV320_VOICE_RATES \
++ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \
++ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
++ SNDRV_PCM_RATE_48000)
++
++
++#define TLV320_VOICE_BITS \
++ (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \
++ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
++
++
++static int caps_charge = 2000;
++static int setting = 1;
++module_param(caps_charge, int, 0);
++module_param(setting, int, 0);
++MODULE_PARM_DESC(caps_charge, "TLV320 cap charge time (msecs)");
++
++static struct workqueue_struct *tlv320_workq = NULL;
++//static struct work_struct tlv320_dapm_work;
++
++/* codec private data */
++struct tlv320_priv {
++ unsigned int sysclk;
++ unsigned int pcmclk;
++};
++
++
++#ifdef TLV320AIC24K
++/* ADDR table */
++static const unsigned char tlv320_reg_addr[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x01, /* CONTROL REG 1 */
++ 0x02, /* CONTROL REG 2 */
++ 0x03, /* CONTROL REG 3A */
++ 0x03, /* CONTROL REG 3B */
++ 0x03, /* CONTROL REG 3C */
++ 0x03, /* CONTROL REG 3D */
++ 0x04, /* CONTROL REG 4 */
++ 0x04, /* CONTROL REG 4 Bis */
++ 0x05, /* CONTROL REG 5A */
++ 0x05, /* CONTROL REG 5B */
++ 0x05, /* CONTROL REG 5C */
++ 0x05, /* CONTROL REG 5D */
++ 0x06, /* CONTROL REG 6A */
++ 0x06, /* CONTROL REG 6B */
++};
++
++/*
++ * DATA case digital SET1:
++ * SSP -> DAC -> OUT
++ * IN -> ADC -> SSP
++ * IN = HNSI (MIC)
++ * OUT = HDSO (SPKG)
++ * Usage: playback, capture streams
++ */
++static const unsigned char tlv320_reg_data_init_set1[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x49, /* CONTROL REG 1 */
++ 0x20, /* CONTROL REG 2 */
++ 0x01, /* CONTROL REG 3A */
++ 0x40, /* CONTROL REG 3B */
++ 0x81, /* CONTROL REG 3C */
++ 0xc0, /* CONTROL REG 3D */
++ 0x02,//0x42(16khz),//0x10, /* CONTROL REG 4 */
++ 0x88,//0x90, /* CONTROL REG 4 Bis */
++ 0x00, /* CONTROL REG 5A */
++ 0x40,//(0dB) /* CONTROL REG 5B */
++ 0xbf, /* CONTROL REG 5C */
++ 0xc0, /* CONTROL REG 5D */
++ 0x02,//(HNSI) /* CONTROL REG 6A */
++ 0x81 //(HDSO) /* CONTROL REG 6B */
++};
++
++/*
++ * DATA case digital SET2:
++ * SSP -> DAC -> OUT
++ * IN -> ADC -> SSP
++ * IN = HDSI (PHONE IN)
++ * OUT = HNSO (PHONE OUT)
++ * Usage: playback, capture streams
++ */
++static const unsigned char tlv320_reg_data_init_set2[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x49, /* CONTROL REG 1 */
++ 0x20, /* CONTROL REG 2 */
++ 0x01, /* CONTROL REG 3A */
++ 0x40, /* CONTROL REG 3B */
++ 0x81, /* CONTROL REG 3C */
++ 0xc0, /* CONTROL REG 3D */
++ 0x02,//0x42(16khz),//0x10, /* CONTROL REG 4 */
++ 0x88,//0x90, /* CONTROL REG 4 Bis */
++ 0x00, /* CONTROL REG 5A */
++ 0x52,//(-27dB) /* CONTROL REG 5B */
++ 0xbf, /* CONTROL REG 5C */
++ 0xc0, /* CONTROL REG 5D */
++ 0x01,//(PHONE IN) /* CONTROL REG 6A */
++ 0x82 //(PHONE OUT) /* CONTROL REG 6B */
++};
++
++/*
++ * DATA case analog:
++ * ADC, DAC, SSP off
++ * Headset input to output (HDSI2O -> 1)
++ * Handset input to output (HNSI2O -> 1)
++ * Usage: room monitor
++ */
++static const unsigned char tlv320_reg_data_init_set3[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x08, /* CONTROL REG 1 */
++ 0x20, /* CONTROL REG 2 */
++ 0x11, /* CONTROL REG 3A */
++ 0x40, /* CONTROL REG 3B */
++ 0x80, /* CONTROL REG 3C */
++ 0xc0, /* CONTROL REG 3D */
++ 0x00, /* CONTROL REG 4 */
++ 0x00, /* CONTROL REG 5A */
++ 0x40, /* CONTROL REG 5B */
++ 0x80, /* CONTROL REG 5C */
++ 0xc0, /* CONTROL REG 5D */
++ 0x60, /* CONTROL REG 6A */
++ 0x80 /* CONTROL REG 6B */
++};
++
++#else // TLV320AIC14k
++
++/* ADDR table */
++static const unsigned char tlv320_reg_addr[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x01, /* CONTROL REG 1 */
++ 0x02, /* CONTROL REG 2 */
++ 0x03, /* CONTROL REG 3 */
++ 0x04, /* CONTROL REG 4 */
++ 0x04, /* CONTROL REG 4 Bis */
++ 0x05, /* CONTROL REG 5A */
++ 0x05, /* CONTROL REG 5B */
++ 0x05, /* CONTROL REG 5C */
++ 0x05, /* CONTROL REG 5D */
++ 0x06 /* CONTROL REG 6 */
++};
++
++/*
++ * DATA case digital:
++ * SSP -> DAC -> OUT
++ * IN -> ADC -> SSP
++ * Usage: playback, capture streams
++ */
++static const unsigned char tlv320_reg_data_init_set1[] = {
++ 0x00, /* CONTROL REG 0 No Operation */
++ 0x41, /* CONTROL REG 1 */
++ 0x20, /* CONTROL REG 2 */
++ 0x09, /* CONTROL REG 3 */
++ 0x02,//0x42(16khz),//0x10, /* CONTROL REG 4 */
++ 0x88,//0x90, /* CONTROL REG 4 Bis */
++ 0x2A, /* CONTROL REG 5A */
++ 0x6A, /* CONTROL REG 5B */
++ 0xbc, /* CONTROL REG 5C */
++ 0xc0, /* CONTROL REG 5D */
++ 0x00 /* CONTROL REG 6 */
++};
++#endif
++/*
++ * read tlv320 register cache
++ */
++static inline unsigned int tlv320_read_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg)
++{
++ u8 *cache = codec->reg_cache;
++ if (reg > ARRAY_SIZE(tlv320_reg_addr))
++ return -1;
++ return cache[reg];
++}
++
++/*
++ * write tlv320 register cache
++ */
++static inline void tlv320_write_reg_cache(struct snd_soc_codec *codec,
++ unsigned int reg, unsigned int value)
++{
++ u8 *cache = codec->reg_cache;
++ if (reg > ARRAY_SIZE(tlv320_reg_addr))
++ return;
++ cache[reg] = value;
++}
++
++/*
++ * read tlv320
++ */
++static int tlv320_read (struct snd_soc_codec *codec, u8 reg)
++{
++ return i2c_smbus_read_byte_data(codec->control_data, reg);
++}
++
++/*
++ * write tlv320
++ */
++static int tlv320_write(struct snd_soc_codec *codec, unsigned int reg,
++ unsigned int value)
++{
++ if (tlv320_reg_addr[reg] > 0x06)
++ return -1;
++
++ tlv320_write_reg_cache (codec, reg, value);
++
++ return i2c_smbus_write_byte_data(codec->control_data, tlv320_reg_addr[reg], value);
++}
++
++
++/*
++ * write block tlv320
++ */
++static int tlv320_write_block (struct snd_soc_codec *codec,
++ const u8 *data, unsigned int len)
++{
++ int ret = -1;
++ int i;
++
++ for (i=0; i<len; i++) {
++ dbg("addr = 0x%02x, data = 0x%02x", tlv320_reg_addr[i], data[i]);
++ if ((ret = tlv320_write(codec, i, data[i])) < 0)
++ break;
++ }
++
++ return ret;
++}
++
++
++static int tlv320_set_dai_fmt(struct snd_soc_codec_dai *codec_dai,
++ unsigned int fmt)
++{
++ dbg("tlv320_set_dai_fmt enter");
++ return 0;
++}
++
++/*
++ * Set PCM DAI bit size and sample rate.
++ */
++static int tlv320_pcm_hw_params(struct snd_pcm_substream *substream,
++ struct snd_pcm_hw_params *params)
++{
++ dbg("tlv320_pcm_hw_params enter");
++ return 0;
++}
++
++
++static int tlv320_config_pcm_sysclk(struct snd_soc_codec_dai *codec_dai,
++ int clk_id, unsigned int freq, int dir)
++{
++ dbg("tlv320_config_pcm_sysclk enter");
++ return 0;
++}
++
++
++/*
++ * Voice over PCM DAI
++ */
++struct snd_soc_codec_dai tlv320_dai[] = {
++{ .name = "TLV320 Voice",
++ .id = 1,
++ .playback = {
++ .stream_name = "Voice Playback",
++ .channels_min = 1,
++ .channels_max = 2,
++ .rates = TLV320_VOICE_RATES,
++ .formats = TLV320_VOICE_BITS,},
++ .capture = {
++ .stream_name = "Voice Capture",
++ .channels_min = 1,
++ .channels_max = 2,
++ .rates = TLV320_VOICE_RATES,
++ .formats = TLV320_VOICE_BITS,},
++ .ops = {
++ .hw_params = tlv320_pcm_hw_params,},
++ .dai_ops = {
++ .digital_mute = NULL,
++ .set_fmt = tlv320_set_dai_fmt,
++ .set_clkdiv = NULL,
++ .set_pll = NULL,
++ .set_sysclk = tlv320_config_pcm_sysclk,
++ },
++},
++
++
++};
++EXPORT_SYMBOL_GPL(tlv320_dai);
++
++
++static void tlv320_work(struct work_struct *work)
++{
++#if 0
++ struct snd_soc_codec *codec =
++ container_of(work, struct snd_soc_codec, delayed_work.work);
++ //wm8753_dapm_event(codec, codec->dapm_state);
++#endif
++}
++
++/*
++ * initialise the TLV320 driver
++ * register the mixer and dsp interfaces with the kernel
++ */
++static int tlv320_init(struct snd_soc_device *socdev)
++{
++ struct snd_soc_codec *codec = socdev->codec;
++ int ret = 0;
++
++ codec->name = "TLV320";
++ codec->owner = THIS_MODULE;
++ codec->read = tlv320_read_reg_cache;
++ codec->write = tlv320_write;
++ codec->dai = tlv320_dai;
++ codec->num_dai = ARRAY_SIZE(tlv320_dai);
++ codec->reg_cache_size = ARRAY_SIZE(tlv320_reg_addr);
++
++ codec->reg_cache =
++ kzalloc(sizeof(u8) * ARRAY_SIZE(tlv320_reg_addr), GFP_KERNEL);
++ if (codec->reg_cache == NULL)
++ return -ENOMEM;
++ memcpy(codec->reg_cache, tlv320_reg_addr,
++ sizeof(u8) * ARRAY_SIZE(tlv320_reg_addr));
++ codec->reg_cache_size = sizeof(u8) * ARRAY_SIZE(tlv320_reg_addr);
++
++ /* register pcms */
++ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
++ if (ret < 0) {
++ kfree(codec->reg_cache);
++ return ret;
++ }
++
++ queue_delayed_work(tlv320_workq,
++ &codec->delayed_work, msecs_to_jiffies(caps_charge));
++
++ ret = snd_soc_register_card(socdev);
++ if (ret < 0) {
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++ }
++
++ return ret;
++}
++
++/* If the i2c layer weren't so broken, we could pass this kind of data
++ around */
++static struct snd_soc_device *tlv320_socdev;
++
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++
++#define I2C_DRIVERID_TLV320 0xfefe /* liam - need a proper id */
++
++static unsigned short normal_i2c[] = { 0, I2C_CLIENT_END };
++
++/* Magic definition of all other variables and things */
++I2C_CLIENT_INSMOD;
++
++static struct i2c_driver tlv320_i2c_driver;
++static struct i2c_client client_template;
++
++static int tlv320_codec_probe(struct i2c_adapter *adap, int addr, int kind)
++{
++ struct snd_soc_device *socdev = tlv320_socdev;
++ struct tlv320_setup_data *setup = socdev->codec_data;
++ struct snd_soc_codec *codec = socdev->codec;
++ struct i2c_client *i2c;
++ int ret, len;
++ const unsigned char *data;
++
++ if (addr != setup->i2c_address)
++ return -ENODEV;
++
++ client_template.adapter = adap;
++ client_template.addr = addr;
++
++ i2c = kzalloc(sizeof(struct i2c_client), GFP_KERNEL);
++ if (i2c == NULL){
++ kfree(codec);
++ return -ENOMEM;
++ }
++ memcpy(i2c, &client_template, sizeof(struct i2c_client));
++ i2c_set_clientdata(i2c, codec);
++ codec->control_data = i2c;
++
++ ret = i2c_attach_client(i2c);
++ if (ret < 0) {
++ err("failed to attach codec at addr %x\n", addr);
++ goto err;
++ }
++
++ ret = tlv320_init(socdev);
++ if (ret < 0) {
++ err("failed to initialise TLV320\n");
++ goto err;
++ }
++
++ switch(setting) {
++ case 1:
++ data = tlv320_reg_data_init_set1;
++ len = sizeof(tlv320_reg_data_init_set1);
++ break;
++ case 2:
++ data = tlv320_reg_data_init_set2;
++ len = sizeof(tlv320_reg_data_init_set2);
++ break;
++ case 3:
++ data = tlv320_reg_data_init_set3;
++ len = sizeof(tlv320_reg_data_init_set3);
++ break;
++ default:
++ data = tlv320_reg_data_init_set1;
++ len = sizeof(tlv320_reg_data_init_set1);
++ break;
++ }
++
++ ret = tlv320_write_block(codec, data, len);
++
++ if (ret < 0) {
++ err("attach error: init status %d\n", ret);
++ } else {
++ info("attach: chip tlv320 at address 0x%02x",
++ tlv320_read(codec, 0x02) << 1);
++ }
++
++ //tlv320_write(codec, CODEC_REG6B, 0x80);
++#if 0
++ int value;
++ int i;
++
++ for (i=0; i<len; i++) {
++ value = tlv320_read(codec, tlv320_reg_addr[i]);
++ dbg("read addr = 0x%02x, data = 0x%02x", tlv320_reg_addr[i], value);
++ mdelay(10);
++ }
++
++#endif
++
++
++ return ret;
++
++err:
++ kfree(codec);
++ kfree(i2c);
++ return ret;
++}
++
++static int tlv320_i2c_detach(struct i2c_client *client)
++{
++ struct snd_soc_codec *codec = i2c_get_clientdata(client);
++ i2c_detach_client(client);
++ kfree(codec->reg_cache);
++ kfree(client);
++ return 0;
++}
++
++static int tlv320_i2c_attach(struct i2c_adapter *adap)
++{
++ return i2c_probe(adap, &addr_data, tlv320_codec_probe);
++}
++
++/* tlv320 i2c codec control layer */
++static struct i2c_driver tlv320_i2c_driver = {
++ .driver = {
++ .name = "tlv320 I2C Codec",
++ .owner = THIS_MODULE,
++ },
++ .id = I2C_DRIVERID_TLV320,
++ .attach_adapter = tlv320_i2c_attach,
++ .detach_client = tlv320_i2c_detach,
++ .command = NULL,
++};
++
++static struct i2c_client client_template = {
++ .name = "tlv320",
++ .driver = &tlv320_i2c_driver,
++};
++#endif
++
++static int tlv320_probe(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct tlv320_setup_data *setup;
++ struct snd_soc_codec *codec;
++ int ret = 0;
++ struct tlv320_priv *tlv320;
++
++ info("TLV320 Audio Codec %s", TLV320_VERSION);
++
++ setup = socdev->codec_data;
++ codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
++ if (codec == NULL)
++ return -ENOMEM;
++
++ tlv320 = kzalloc(sizeof(struct tlv320_priv), GFP_KERNEL);
++ if (tlv320 == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++
++ codec->private_data = tlv320;
++
++ socdev->codec = codec;
++ mutex_init(&codec->mutex);
++ INIT_LIST_HEAD(&codec->dapm_widgets);
++ INIT_LIST_HEAD(&codec->dapm_paths);
++ tlv320_socdev = socdev;
++
++ INIT_DELAYED_WORK(&codec->delayed_work, tlv320_work);
++ tlv320_workq = create_workqueue("tlv320");
++ if (tlv320_workq == NULL) {
++ kfree(codec);
++ return -ENOMEM;
++ }
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ if (setup->i2c_address) {
++ normal_i2c[0] = setup->i2c_address;
++ codec->hw_write = (hw_write_t)i2c_master_send;
++ ret = i2c_add_driver(&tlv320_i2c_driver);
++ if (ret != 0)
++ printk(KERN_ERR "can't add i2c driver");
++ }
++#else
++ /* Add other interfaces here */
++#endif
++ return ret;
++}
++
++/* power down chip */
++static int tlv320_remove(struct platform_device *pdev)
++{
++ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
++ struct snd_soc_codec *codec = socdev->codec;
++
++ if (tlv320_workq)
++ destroy_workqueue(tlv320_workq);
++ snd_soc_free_pcms(socdev);
++ snd_soc_dapm_free(socdev);
++#if defined (CONFIG_I2C) || defined (CONFIG_I2C_MODULE)
++ i2c_del_driver(&tlv320_i2c_driver);
++#endif
++ kfree(codec->private_data);
++ kfree(codec);
++
++ return 0;
++}
++
++struct snd_soc_codec_device soc_codec_dev_tlv320 = {
++ .probe = tlv320_probe,
++ .remove = tlv320_remove,
++};
++
++EXPORT_SYMBOL_GPL(soc_codec_dev_tlv320);
++
++MODULE_DESCRIPTION("ASoC TLV320 driver");
++MODULE_AUTHOR("Nicola Perrino");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/codecs/tlv320.h
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/codecs/tlv320.h 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,111 @@
++/*
++ * tlv320.h -- TLV 320 ALSA Soc Audio driver
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2006 Atlab srl.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ * Nicola Perrino <nicola.perrino at atlab.it>
++ *
++ * 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.
++ *
++ */
++
++#ifndef _TLV320_H
++#define _TLV320_H
++
++#define TLV320AIC24K
++
++
++/* TLV320 register space */
++#define CODEC_NOOP 0x00
++#define CODEC_REG1 0x01
++#define CODEC_REG2 0x02
++#define CODEC_REG3A 0x03
++#define CODEC_REG3B 0x04
++#define CODEC_REG3C 0x05
++#define CODEC_REG3D 0x06
++#define CODEC_REG4A 0x07
++#define CODEC_REG4B 0x08
++#define CODEC_REG5A 0x09
++#define CODEC_REG5B 0x0a
++#define CODEC_REG5C 0x0b
++#define CODEC_REG5D 0x0c
++#define CODEC_REG6A 0x0d
++#define CODEC_REG6B 0x0e
++
++
++// Control Register 1
++#define REG1_CONTINUOUS 0x40
++#define REG1_IIR_EN 0x20
++#define REG1_MIC_BIAS_235 0x08
++#define REG1_ANALOG_LOOP_BACK 0x04
++#define REG1_DIGITAL_LOOP_BACK 0x02
++#define REG1_DAC16 0x01
++
++// Control Register 2
++#define REG2_TURBO_EN 0x80
++#define REG2_FIR_BYPASS 0x40
++#define REG2_GPIO 0x02
++#define REG2_GPIO_1 0x06
++
++// Control Register 3A
++#define REG3_PWDN_ALL 0x30
++#define REG3_PWDN_ADC 0x10
++#define REG3_PWDN_DAC 0x20
++#define REG3_SW_RESET 0x08
++#define REG3_SAMPLING_FACTOR1 0x01
++#define REG3_SAMPLING_FACTOR2 0x02
++
++// Control Register 3B
++#define REG3_8KBP_EN 0x60
++#define REG3_MUTE_OUTP1 0x42
++#define REG3_MUTE_OUTP2 0x48
++#define REG3_MUTE_OUTP3 0x44
++
++// Control Register 4
++#define REG4_FSDIV_M 0x85 //M=5
++#define REG4_FSDIV_NP 0x08 //N=1, P=8
++//#define REG4_FSDIV_NP 0x01 //N=1, P=8
++#define REG4_FSDIV_NP1 0x02 //N=16, P=2
++
++// Control Register 5
++#define REG5A_ADC_GAIN 0x02 //3dB
++#define REG5A_ADC_MUTE 0x0f //Mute
++#define REG5B_DAC_GAIN 0x42 //-3dB
++#define REG5B_DAC_MUTE 0x4f //Mute
++#define REG5C_SIDETONE_MUTE 0xBF
++
++// Control Register 6
++#define REG6A_AIC24A_CH1_IN 0x08 //INP1 to ADC
++#define REG6B_AIC24A_CH1_OUT 0x82 //OUTP2 to DAC
++#define REG6A_AIC24A_CH2_IN 0x02 //INP2 to ADC
++#define REG6B_AIC24A_CH2_OUT 0x81 //OUTP3 to DAC
++
++/* clock inputs */
++#define TLV320_MCLK 0
++#define TLV320_PCMCLK 1
++
++
++struct tlv320_setup_data {
++ unsigned short i2c_address;
++};
++
++/* DAI ifmodes */
++/* mode 1 IFMODE = 00 */
++#define TLV320_DAI_MODE1_VOICE 0
++#define TLV320_DAI_MODE1_HIFI 1
++/* mode 2 IFMODE = 01 */
++#define TLV320_DAI_MODE2_VOICE 2
++/* mode 3 IFMODE = 10 */
++#define TLV320_DAI_MODE3_HIFI 3
++/* mode 4 IFMODE = 11 */
++#define TLV320_DAI_MODE4_HIFI 4
++
++extern struct snd_soc_codec_dai tlv320_dai[5];
++extern struct snd_soc_codec_device soc_codec_dev_tlv320;
++
++#endif
+Index: linux-2.6.17.14-fic4.test/sound/soc/pxa/amesom_tlv320.c
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/pxa/amesom_tlv320.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,211 @@
++/*
++ * amesom_tlv320.c -- SoC audio for Amesom
++ *
++ * Copyright 2005 Wolfson Microelectronics PLC.
++ * Copyright 2006 Atlab srl.
++ *
++ * Authors: Liam Girdwood <liam.girdwood at wolfsonmicro.com>
++ * Nicola Perrino <nicola.perrino at atlab.it>
++ *
++ * 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.
++ *
++ * Revision history
++ * 5th Dec 2006 Initial version.
++ *
++ */
++
++#include <linux/module.h>
++#include <linux/moduleparam.h>
++#include <linux/device.h>
++#include <linux/i2c.h>
++#include <sound/driver.h>
++#include <sound/core.h>
++#include <sound/pcm.h>
++#include <sound/soc.h>
++#include <sound/soc-dapm.h>
++
++#include <asm/hardware.h>
++#include <asm/arch/pxa-regs.h>
++#include <asm/arch/audio.h>
++
++#include "../codecs/tlv320.h"
++#include "pxa2xx-pcm.h"
++#include "pxa2xx-i2s.h"
++#include "pxa2xx-ssp.h"
++
++
++/*
++ * SSP2 GPIO's
++ */
++
++#define GPIO11_SSP2RX_MD (11 | GPIO_ALT_FN_2_IN)
++#define GPIO13_SSP2TX_MD (13 | GPIO_ALT_FN_1_OUT)
++#define GPIO50_SSP2CLKS_MD (50 | GPIO_ALT_FN_3_IN)
++#define GPIO14_SSP2FRMS_MD (14 | GPIO_ALT_FN_2_IN)
++#define GPIO50_SSP2CLKM_MD (50 | GPIO_ALT_FN_3_OUT)
++#define GPIO14_SSP2FRMM_MD (14 | GPIO_ALT_FN_2_OUT)
++
++
++static struct snd_soc_machine amesom;
++
++
++static int amesom_probe(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int amesom_remove(struct platform_device *pdev)
++{
++ return 0;
++}
++
++static int tlv320_voice_startup(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static void tlv320_voice_shutdown(struct snd_pcm_substream *substream)
++{
++ return;
++}
++
++/*
++ * Tlv320 uses SSP port for playback.
++ */
++static int tlv320_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
++ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
++ int ret = 0;
++
++ //printk("tlv320_voice_hw_params enter\n");
++ switch(params_rate(params)) {
++ case 8000:
++ //printk("tlv320_voice_hw_params 8000\n");
++ break;
++ case 16000:
++ //printk("tlv320_voice_hw_params 16000\n");
++ break;
++ default:
++ break;
++ }
++
++ // CODEC MASTER, SSP SLAVE
++
++ /* set codec DAI configuration */
++ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_MSB |
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++ if (ret < 0)
++ return ret;
++
++ /* set cpu DAI configuration */
++ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_MSB |
++ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
++ if (ret < 0)
++ return ret;
++
++ /* set the SSP system clock as input (unused) */
++ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_NET_PLL, 0,
++ SND_SOC_CLOCK_IN);
++ if (ret < 0)
++ return ret;
++
++ /* set SSP slots */
++ //ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x1, slots);
++ ret = cpu_dai->dai_ops.set_tdm_slot(cpu_dai, 0x3, 1);
++ if (ret < 0)
++ return ret;
++
++ return 0;
++}
++
++static int tlv320_voice_hw_free(struct snd_pcm_substream *substream)
++{
++ return 0;
++}
++
++static struct snd_soc_ops tlv320_voice_ops = {
++ .startup = tlv320_voice_startup,
++ .shutdown = tlv320_voice_shutdown,
++ .hw_params = tlv320_voice_hw_params,
++ .hw_free = tlv320_voice_hw_free,
++};
++
++
++static struct snd_soc_dai_link amesom_dai[] = {
++{
++ .name = "TLV320",
++ .stream_name = "TLV320 Voice",
++ .cpu_dai = &pxa_ssp_dai[PXA2XX_DAI_SSP2],
++ .codec_dai = &tlv320_dai[TLV320_DAI_MODE1_VOICE],
++ .ops = &tlv320_voice_ops,
++},
++};
++
++static struct snd_soc_machine amesom = {
++ .name = "Amesom",
++ .probe = amesom_probe,
++ .remove = amesom_remove,
++ .dai_link = amesom_dai,
++ .num_links = ARRAY_SIZE(amesom_dai),
++};
++
++static struct tlv320_setup_data amesom_tlv320_setup = {
++#ifdef TLV320AIC24K //codec2
++ .i2c_address = 0x41,
++#else // TLV320AIC14k
++ .i2c_address = 0x40,
++#endif
++};
++
++static struct snd_soc_device amesom_snd_devdata = {
++ .machine = &amesom,
++ .platform = &pxa2xx_soc_platform,
++ .codec_dev = &soc_codec_dev_tlv320,
++ .codec_data = &amesom_tlv320_setup,
++};
++
++static struct platform_device *amesom_snd_device;
++
++static int __init amesom_init(void)
++{
++ int ret;
++
++ amesom_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!amesom_snd_device)
++ return -ENOMEM;
++
++ platform_set_drvdata(amesom_snd_device, &amesom_snd_devdata);
++ amesom_snd_devdata.dev = &amesom_snd_device->dev;
++ ret = platform_device_add(amesom_snd_device);
++
++ if (ret)
++ platform_device_put(amesom_snd_device);
++
++
++ /* SSP port 2 slave */
++ pxa_gpio_mode(GPIO11_SSP2RX_MD);
++ pxa_gpio_mode(GPIO13_SSP2TX_MD);
++ pxa_gpio_mode(GPIO50_SSP2CLKS_MD);
++ pxa_gpio_mode(GPIO14_SSP2FRMS_MD);
++
++ return ret;
++}
++
++static void __exit amesom_exit(void)
++{
++ platform_device_unregister(amesom_snd_device);
++}
++
++module_init(amesom_init);
++module_exit(amesom_exit);
++
++/* Module information */
++MODULE_AUTHOR("Nicola Perrino");
++MODULE_DESCRIPTION("ALSA SoC TLV320 Amesom");
++MODULE_LICENSE("GPL");
+Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/neo1973_wm8753.c
+===================================================================
+--- /dev/null 1970-01-01 00:00:00.000000000 +0000
++++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/neo1973_wm8753.c 2007-02-06 15:55:20.000000000 +0100
+@@ -0,0 +1,686 @@
++/*
++ * neo1973_wm8753.c -- SoC audio for Neo1973
++ *
++ * Copyright 2007 Wolfson Microelectronics PLC.
+ * Author: Graeme Gregory
+ * graeme.gregory at wolfsonmicro.com or linux at wolfsonmicro.com
+ *
@@ -34493,6 +35855,7 @@
+ *
+ * Revision history
+ * 20th Jan 2007 Initial version.
++ * 05th Feb 2007 Rename all to Neo1973
+ *
+ */
+
@@ -34522,8 +35885,8 @@
+#include "s3c24xx-pcm.h"
+#include "s3c24xx-i2s.h"
+
-+#define SMDK2440_DEBUG 0
-+#if SMDK2440_DEBUG
++#define NEO1973_DEBUG 0
++#if NEO1973_DEBUG
+#define DBG(x...) printk(KERN_DEBUG x)
+#else
+#define DBG(x...)
@@ -34536,11 +35899,14 @@
+#define NEO_GSM_CALL_AUDIO_BLUETOOTH 3
+#define NEO_STEREO_TO_SPEAKERS 4
+#define NEO_STEREO_TO_HEADPHONES 5
++#define NEO_CAPTURE_HANDSET 6
++#define NEO_CAPTURE_HEADSET 7
++#define NEO_CAPTURE_BLUETOOTH 8
+
-+static struct snd_soc_machine smdk2440;
++static struct snd_soc_machine neo1973;
+static struct i2c_client *i2c;
+
-+static int smdk2440_hifi_hw_params(struct snd_pcm_substream *substream,
++static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -34619,14 +35985,15 @@
+ return ret;
+
+ /* codec PLL input is PCLK/4 */
-+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, iis_clkrate/4, pll_out);
++ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, iis_clkrate/4,
++ pll_out);
+ if (ret < 0)
+ return ret;
+
+ return 0;
+}
+
-+static int smdk2440_hifi_hw_free(struct snd_pcm_substream *substream)
++static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
@@ -34636,32 +36003,23 @@
+}
+
+/*
-+ * smdk2440 WM8753 HiFi DAI opserations.
++ * Neo1973 WM8753 HiFi DAI opserations.
+ */
-+static struct snd_soc_ops smdk2440_hifi_ops = {
-+ .hw_params = smdk2440_hifi_hw_params,
-+ .hw_free = smdk2440_hifi_hw_free,
++static struct snd_soc_ops neo1973_hifi_ops = {
++ .hw_params = neo1973_hifi_hw_params,
++ .hw_free = neo1973_hifi_hw_free,
+};
+
-+static int smdk2440_voice_startup(struct snd_pcm_substream *substream)
-+{
-+ return 0;
-+}
-+
-+static void smdk2440_voice_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+static int smdk2440_voice_hw_params(struct snd_pcm_substream *substream,
++static int neo1973_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
-+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
+ unsigned int pcmdiv = 0;
+ int ret = 0;
+ unsigned long iis_clkrate;
-+// gg where is sysclk coming from for voice ??
++
++ /* todo: gg where is sysclk coming from for voice ?? */
+ iis_clkrate = s3c24xx_i2s_get_clockrate();
+
+ if (params_rate(params) != 8000)
@@ -34671,7 +36029,7 @@
+
+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
+
-+// gg check mode (DSP_B) against CSR datasheet
++ /* todo: gg check mode (DSP_B) against CSR datasheet */
+ /* set codec DAI configuration */
+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
@@ -34698,7 +36056,7 @@
+ return 0;
+}
+
-+static int smdk2440_voice_hw_free(struct snd_pcm_substream *substream)
++static int neo1973_voice_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
@@ -34707,46 +36065,23 @@
+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
+}
+
-+static struct snd_soc_ops smdk2440_voice_ops = {
-+ .startup = smdk2440_voice_startup,
-+ .shutdown = smdk2440_voice_shutdown,
-+ .hw_params = smdk2440_voice_hw_params,
-+ .hw_free = smdk2440_voice_hw_free,
++static struct snd_soc_ops neo1973_voice_ops = {
++ .hw_params = neo1973_voice_hw_params,
++ .hw_free = neo1973_voice_hw_free,
+};
+
-+static int smdk2440_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+ return 0;
-+}
++static int neo1973_scenario = 0;
+
-+static int smdk2440_resume(struct platform_device *pdev)
-+{
-+ return 0;
-+}
-+
-+static int smdk2440_probe(struct platform_device *pdev)
-+{
-+
-+ return 0;
-+}
-+
-+static int smdk2440_remove(struct platform_device *pdev)
-+{
-+ return 0;
-+}
-+
-+static int smdk2440_scenario = 0;
-+
-+static int smdk2440_get_scenario(struct snd_kcontrol *kcontrol,
++static int neo1973_get_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
-+ ucontrol->value.integer.value[0] = smdk2440_scenario;
++ ucontrol->value.integer.value[0] = neo1973_scenario;
+ return 0;
+}
+
+static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario)
+{
-+ switch(smdk2440_scenario) {
++ switch(neo1973_scenario) {
+ case NEO_AUDIO_OFF:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
@@ -34770,8 +36105,8 @@
+ break;
+ case NEO_GSM_CALL_AUDIO_BLUETOOTH:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
-+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
-+ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1);
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
@@ -34789,6 +36124,27 @@
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ break;
++ case NEO_CAPTURE_HANDSET:
++ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
++ snd_soc_dapm_set_endpoint(codec, "Call Mic", 1);
++ break;
++ case NEO_CAPTURE_HEADSET:
++ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1);
++ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
++ break;
++ case NEO_CAPTURE_BLUETOOTH:
++ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
++ snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0);
++ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
++ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
++ break;
+ default:
+ snd_soc_dapm_set_endpoint(codec, "Audio Out", 0);
+ snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0);
@@ -34796,19 +36152,23 @@
+ snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0);
+ snd_soc_dapm_set_endpoint(codec, "Call Mic", 0);
+ }
++
++ snd_soc_dapm_sync_endpoints(codec);
++
++ return 0;
+}
+
-+static int smdk2440_set_scenario(struct snd_kcontrol *kcontrol,
++static int neo1973_set_scenario(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+
-+ if (smdk2440_scenario == ucontrol->value.integer.value[0])
++ if (neo1973_scenario == ucontrol->value.integer.value[0])
+ return 0;
+
-+ smdk2440_scenario = ucontrol->value.integer.value[0];
++ neo1973_scenario = ucontrol->value.integer.value[0];
+
-+ set_scenario_endpoints(codec, smdk2440_scenario);
++ set_scenario_endpoints(codec, neo1973_scenario);
+
+ return 1;
+}
@@ -34885,8 +36245,8 @@
+ SND_SOC_DAPM_LINE("Audio Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
+ SND_SOC_DAPM_LINE("GSM Line In", NULL),
-+ SND_SOC_DAPM_LINE("GSM Line Out", NULL),
-+ SND_SOC_DAPM_HP("Headset Jack", NULL),
++ SND_SOC_DAPM_MIC("Headset Mic", NULL),
++ SND_SOC_DAPM_MIC("Call Mic", NULL),
+};
+
+
@@ -34912,6 +36272,9 @@
+ {"MIC2N", NULL, "Mic Bias"},
+ {"Mic Bias", NULL, "Call Mic"},
+
++ /* Connect the ALC pins */
++ {"ACIN", NULL, "ACOP"},
++
+ {NULL, NULL, NULL},
+};
+
@@ -34933,14 +36296,17 @@
+ "GSM Headset",
+ "GSM Bluetooth",
+ "Speakers",
-+ "Headphones"
++ "Headphones",
++ "Capture Handset",
++ "Capture Headset",
++ "Capture Bluetooth"
+};
+
+static const struct soc_enum neo_scenario_enum[] = {
-+ SOC_ENUM_SINGLE_EXT(6,neo_scenarios),
++ SOC_ENUM_SINGLE_EXT(9,neo_scenarios),
+};
+
-+static const struct snd_kcontrol_new wm8753_smdk2440_controls[] = {
++static const struct snd_kcontrol_new wm8753_neo1973_controls[] = {
+ SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0,
@@ -34950,7 +36316,7 @@
+ SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0],
+ lm4857_get_mode, lm4857_set_mode),
+ SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0],
-+ smdk2440_get_scenario, smdk2440_set_scenario),
++ neo1973_get_scenario, neo1973_set_scenario),
+ SOC_SINGLE_EXT("Amp Spk 3D Playback Switch", LM4857_LVOL, 5, 1, 0,
+ lm4857_get_reg, lm4857_set_reg),
+ SOC_SINGLE_EXT("Amp HP 3d Playback Switch", LM4857_RVOL, 5, 1, 0,
@@ -34963,10 +36329,10 @@
+
+/*
+ * This is an example machine initialisation for a wm8753 connected to a
-+ * smdk2440 II. It is missing logic to detect hp/mic insertions and logic
++ * neo1973 II. It is missing logic to detect hp/mic insertions and logic
+ * to re-route the audio in such an event.
+ */
-+static int smdk2440_wm8753_init(struct snd_soc_codec *codec)
++static int neo1973_wm8753_init(struct snd_soc_codec *codec)
+{
+ int i, err;
+
@@ -34982,54 +36348,27 @@
+ /* set endpoints to default mode */
+ set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
-+ /* Add smdk2440 specific widgets */
-+ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) {
++ /* Add neo1973 specific widgets */
++ for(i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++)
+ snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]);
-+ }
+
-+ /* add smdk2440 specific controls */
-+ for (i = 0; i < ARRAY_SIZE(wm8753_smdk2440_controls); i++) {
++ /* add neo1973 specific controls */
++ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
+ if ((err = snd_ctl_add(codec->card,
-+ snd_soc_cnew(&wm8753_smdk2440_controls[i],codec, NULL))) < 0)
++ snd_soc_cnew(&wm8753_neo1973_controls[i],codec, NULL))) < 0)
+ return err;
+ }
+
-+ /* set up smdk2440 specific audio path audio_mapnects */
++ /* set up neo1973 specific audio path audio_mapnects */
+ for(i = 0; audio_map[i][0] != NULL; i++) {
-+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
++ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1],
++ audio_map[i][2]);
+ }
+
+ snd_soc_dapm_sync_endpoints(codec);
+ return 0;
+}
+
-+/* Do specific bluetooth PCM startup here */
-+static int bt_startup(struct snd_pcm_substream *substream)
-+{
-+ return 0;
-+}
-+
-+/* Do specific bluetooth PCM shutdown here */
-+static void bt_shutdown (struct snd_pcm_substream *substream)
-+{
-+}
-+
-+/* Do pecific bluetooth PCM hw params init here */
-+static int bt_hw_params(struct snd_pcm_substream *substream,
-+ struct snd_pcm_hw_params *params)
-+{
-+ return 0;
-+}
-+
-+/* Do specific bluetooth PCM hw params free here */
-+static int bt_hw_free(struct snd_pcm_substream *substream)
-+{
-+ return 0;
-+}
-+
-+#define BT_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-+ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100)
-+
+/*
+ * BT Codec DAI
+ */
@@ -35040,61 +36379,50 @@
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 1,
-+ .rates = BT_RATES,
++ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 1,
-+ .rates = BT_RATES,
++ .rates = SNDRV_PCM_RATE_8000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,},
-+ .ops = {
-+ .startup = bt_startup,
-+ .shutdown = bt_shutdown,
-+ .hw_params = bt_hw_params,
-+ .hw_free = bt_hw_free,
-+ },
+};
+
-+static struct snd_soc_dai_link smdk2440_dai[] = {
++static struct snd_soc_dai_link neo1973_dai[] = {
+{ /* Hifi Playback - for similatious use with voice below */
+ .name = "WM8753",
+ .stream_name = "WM8753 HiFi",
+ .cpu_dai = &s3c24xx_i2s_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
-+ .init = smdk2440_wm8753_init,
-+ .ops = &smdk2440_hifi_ops,
++ .init = neo1973_wm8753_init,
++ .ops = &neo1973_hifi_ops,
+},
+{ /* Voice via BT */
+ .name = "Bluetooth",
+ .stream_name = "Voice",
+ .cpu_dai = &bt_dai,
+ .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
-+ .ops = &smdk2440_voice_ops,
++ .ops = &neo1973_voice_ops,
+},
+};
+
-+static struct snd_soc_machine smdk2440 = {
-+ .name = "smdk2440",
-+ .probe = smdk2440_probe,
-+ .remove = smdk2440_remove,
-+ .suspend_pre = smdk2440_suspend,
-+ .resume_post = smdk2440_resume,
-+ .dai_link = smdk2440_dai,
-+ .num_links = ARRAY_SIZE(smdk2440_dai),
++static struct snd_soc_machine neo1973 = {
++ .name = "neo1973",
++ .dai_link = neo1973_dai,
++ .num_links = ARRAY_SIZE(neo1973_dai),
+};
+
-+static struct wm8753_setup_data smdk2440_wm8753_setup = {
++static struct wm8753_setup_data neo1973_wm8753_setup = {
+ .i2c_address = 0x1a,
+};
+
-+static struct snd_soc_device smdk2440_snd_devdata = {
-+ .machine = &smdk2440,
++static struct snd_soc_device neo1973_snd_devdata = {
++ .machine = &neo1973,
+ .platform = &s3c24xx_soc_platform,
+ .codec_dev = &soc_codec_dev_wm8753,
-+ .codec_data = &smdk2440_wm8753_setup,
++ .codec_data = &neo1973_wm8753_setup,
+};
+
-+static struct i2c_driver wm8753_i2c_driver;
+static struct i2c_client client_template;
+
+static unsigned short normal_i2c[] = { 0x7C, I2C_CLIENT_END };
@@ -35116,7 +36444,6 @@
+ return -ENOMEM;
+ }
+ memcpy(i2c, &client_template, sizeof(struct i2c_client));
-+ //i2c_set_clientdata(i2c, codec);
+
+ ret = i2c_attach_client(i2c);
+ if (ret < 0) {
@@ -35164,22 +36491,22 @@
+ .driver = &lm4857_i2c_driver,
+};
+
-+static struct platform_device *smdk2440_snd_device;
++static struct platform_device *neo1973_snd_device;
+
-+static int __init smdk2440_init(void)
++static int __init neo1973_init(void)
+{
+ int ret;
+
-+ smdk2440_snd_device = platform_device_alloc("soc-audio", -1);
-+ if (!smdk2440_snd_device)
++ neo1973_snd_device = platform_device_alloc("soc-audio", -1);
++ if (!neo1973_snd_device)
+ return -ENOMEM;
+
-+ platform_set_drvdata(smdk2440_snd_device, &smdk2440_snd_devdata);
-+ smdk2440_snd_devdata.dev = &smdk2440_snd_device->dev;
-+ ret = platform_device_add(smdk2440_snd_device);
++ platform_set_drvdata(neo1973_snd_device, &neo1973_snd_devdata);
++ neo1973_snd_devdata.dev = &neo1973_snd_device->dev;
++ ret = platform_device_add(neo1973_snd_device);
+
+ if (ret)
-+ platform_device_put(smdk2440_snd_device);
++ platform_device_put(neo1973_snd_device);
+
+ ret = i2c_add_driver(&lm4857_i2c_driver);
+ if (ret != 0)
@@ -35188,505 +36515,15 @@
+ return ret;
+}
+
-+static void __exit smdk2440_exit(void)
++static void __exit neo1973_exit(void)
+{
-+ platform_device_unregister(smdk2440_snd_device);
++ platform_device_unregister(neo1973_snd_device);
+}
+
-+module_init(smdk2440_init);
-+module_exit(smdk2440_exit);
++module_init(neo1973_init);
++module_exit(neo1973_exit);
+
+/* Module information */
+MODULE_AUTHOR("Graeme Gregory, graeme.gregory at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM8753 smdk2440");
++MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973");
+MODULE_LICENSE("GPL");
-Index: linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.h
-===================================================================
---- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/imx-ssi.h 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,28 @@
-+/*
-+ * 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.
-+ */
-+
-+#ifndef _IMX_SSI_H
-+#define _IMX_SSI_H
-+
-+/* pxa2xx DAI SSP ID's */
-+#define IMX_DAI_SSI1 0
-+#define IMX_DAI_SSI2 1
-+
-+/* SSI clock sources */
-+#define IMX_SSP_SYS_CLK 0
-+
-+/* SSI audio dividers */
-+#define IMX_SSI_DIV_2 0
-+#define IMX_SSI_DIV_PSR 1
-+#define IMX_SSI_DIV_PM 2
-+
-+/* SSI Div 2 */
-+#define IMX_SSI_DIV_2_OFF ~SSI_STCCR_DIV2
-+#define IMX_SSI_DIV_2_ON SSI_STCCR_DIV2
-+
-+extern struct snd_soc_cpu_dai imx_ssi_pcm_dai[2];
-+
-+#endif
-Index: linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.h
-===================================================================
---- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/at91/at91-i2s.h 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,27 @@
-+/*
-+ * at91-i2s.h - ALSA I2S interface for the Atmel AT91 SoC
-+ *
-+ * Author: Frank Mandarino <fmandarino at endrelia.com>
-+ * Endrelia Technologies Inc.
-+ * Created: Jan 9, 2007
-+ *
-+ * 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.
-+ */
-+
-+#ifndef _AT91_I2S_H
-+#define _AT91_I2S_H
-+
-+/* I2S system clock ids */
-+#define AT91_SYSCLK_MCK 0 /* SSC uses AT91 MCK as system clock */
-+
-+/* I2S divider ids */
-+#define AT91SSC_CMR_DIV 0 /* MCK divider for BCLK */
-+#define AT91SSC_TCMR_PERIOD 1 /* BCLK divider for transmit FS */
-+#define AT91SSC_RCMR_PERIOD 2 /* BCLK divider for receive FS */
-+
-+extern struct snd_soc_cpu_dai at91_i2s_dai[];
-+
-+#endif /* _AT91_I2S_H */
-+
-Index: linux-2.6.17.14-fic4.test/sound/soc/imx/mx31ads_wm8753.c
-===================================================================
---- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/imx/mx31ads_wm8753.c 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,400 @@
-+/*
-+ * mx31ads_wm8753.c -- SoC audio for mx31ads
-+ *
-+ * Copyright 2005 Wolfson Microelectronics PLC.
-+ * Author: Liam Girdwood
-+ * liam.girdwood at wolfsonmicro.com or linux at wolfsonmicro.com
-+ *
-+ * mx31ads audio amplifier code taken from arch/arm/mach-pxa/mx31ads.c
-+ * Copyright: MontaVista Software Inc.
-+ *
-+ * 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.
-+ *
-+ * Revision history
-+ * 30th Oct 2005 Initial version.
-+ *
-+ */
-+
-+#include <linux/module.h>
-+#include <linux/moduleparam.h>
-+#include <linux/device.h>
-+#include <linux/i2c.h>
-+#include <sound/driver.h>
-+#include <sound/core.h>
-+#include <sound/pcm.h>
-+#include <sound/soc.h>
-+#include <sound/soc-dapm.h>
-+
-+#include <asm/hardware.h>
-+
-+#include "../codecs/wm8753.h"
-+#include "imx31-pcm.h"
-+#include "imx-ssi.h"
-+
-+static struct snd_soc_machine mx31ads;
-+
-+static int mx31ads_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai;
-+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+ unsigned int pll_out = 0, bclk = 0, fmt = 0;
-+ int ret = 0;
-+
-+ /*
-+ * The WM8753 is better at generating accurate audio clocks than the
-+ * MX31 SSI controller, so we will use it as master when we can.
-+ */
-+ switch (params_rate(params)) {
-+ case 8000:
-+ case 16000:
-+ fmt = SND_SOC_DAIFMT_CBS_CFS;
-+ pll_out = 12288000;
-+ break;
-+ case 48000:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_4;
-+ pll_out = 12288000;
-+ break;
-+ case 96000:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_2;
-+ pll_out = 12288000;
-+ break;
-+ case 11025:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_16;
-+ pll_out = 11289600;
-+ break;
-+ case 22050:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_8;
-+ pll_out = 11289600;
-+ break;
-+ case 44100:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_4;
-+ pll_out = 11289600;
-+ break;
-+ case 88200:
-+ fmt = SND_SOC_DAIFMT_CBM_CFS;
-+ bclk = WM8753_BCLK_DIV_2;
-+ pll_out = 11289600;
-+ break;
-+ }
-+
-+ /* set codec DAI configuration */
-+ ret = codec_dai->dai_ops.set_fmt(codec_dai,
-+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set cpu DAI configuration */
-+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai,
-+ SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | fmt);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set the codec system clock for DAC and ADC */
-+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out,
-+ SND_SOC_CLOCK_IN);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set the SSI system clock as input (unused) */
-+ ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0,
-+ SND_SOC_CLOCK_IN);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set codec BCLK division for sample rate */
-+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* codec PLL input is 13 MHz */
-+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 13000000, pll_out);
-+ if (ret < 0)
-+ return ret;
-+
-+ return 0;
-+}
-+
-+static int mx31ads_hifi_hw_free(struct snd_pcm_substream *substream)
-+{
-+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+
-+ /* disable the PLL */
-+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0);
-+}
-+
-+/*
-+ * mx31ads WM8753 HiFi DAI opserations.
-+ */
-+static struct snd_soc_ops mx31ads_hifi_ops = {
-+ .hw_params = mx31ads_hifi_hw_params,
-+ .hw_free = mx31ads_hifi_hw_free,
-+};
-+
-+static int mx31ads_voice_startup(struct snd_pcm_substream *substream)
-+{
-+ return 0;
-+}
-+
-+static void mx31ads_voice_shutdown(struct snd_pcm_substream *substream)
-+{
-+}
-+
-+static int mx31ads_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai;
-+ struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai;
-+ unsigned int pll_out = 0, bclk = 0, pcmdiv = 0;
-+ int ret = 0;
-+
-+ /*
-+ * The WM8753 is far better at generating accurate audio clocks than the
-+ * pxa2xx SSP controller, so we will use it as master when we can.
-+ */
-+ switch (params_rate(params)) {
-+ case 8000:
-+ pll_out = 12288000;
-+ pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 256kHz */
-+ break;
-+ case 16000:
-+ pll_out = 12288000;
-+ pcmdiv = WM8753_PCM_DIV_3; /* 4.096 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 512kHz */
-+ break;
-+ case 48000:
-+ pll_out = 12288000;
-+ pcmdiv = WM8753_PCM_DIV_1; /* 12.288 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 1.536 MHz */
-+ break;
-+ case 11025:
-+ pll_out = 11289600;
-+ pcmdiv = WM8753_PCM_DIV_4; /* 11.2896 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 352.8 kHz */
-+ break;
-+ case 22050:
-+ pll_out = 11289600;
-+ pcmdiv = WM8753_PCM_DIV_2; /* 11.2896 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 705.6 kHz */
-+ break;
-+ case 44100:
-+ pll_out = 11289600;
-+ pcmdiv = WM8753_PCM_DIV_1; /* 11.2896 MHz */
-+ bclk = WM8753_VXCLK_DIV_8; /* 1.4112 MHz */
-+ break;
-+ }
-+
-+ /* set codec DAI configuration */
-+ ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_A |
-+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set cpu DAI configuration */
-+ ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
-+ SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set the codec system clock for DAC and ADC */
-+ ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, pll_out,
-+ SND_SOC_CLOCK_IN);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set the SSP system clock as input (unused) */
-+// ret = cpu_dai->dai_ops.set_sysclk(cpu_dai, PXA2XX_SSP_CLK_PLL, 0,
-+// SND_SOC_CLOCK_IN);
-+// if (ret < 0)
-+// return ret;
-+
-+ /* set codec BCLK division for sample rate */
-+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_VXCLKDIV, bclk);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* set codec PCM division for sample rate */
-+ ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv);
-+ if (ret < 0)
-+ return ret;
-+
-+ /* codec PLL input is 13 MHz */
-+ ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 13000000, pll_out);
-+ if (ret < 0)
-+ return ret;
-+
-+ return 0;
-+}
-+
-+static int mx31ads_voice_hw_free(struct snd_pcm_substream *substream)
-+{
-+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
-+ struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai;
-+
-+ /* disable the PLL */
-+ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0);
-+}
-+
-+static struct snd_soc_ops mx31ads_voice_ops = {
-+ .startup = mx31ads_voice_startup,
-+ .shutdown = mx31ads_voice_shutdown,
-+ .hw_params = mx31ads_voice_hw_params,
-+ .hw_free = mx31ads_voice_hw_free,
-+};
-+
-+static int mx31ads_suspend(struct platform_device *pdev, pm_message_t state)
-+{
-+ return 0;
-+}
-+
-+static int mx31ads_resume(struct platform_device *pdev)
-+{
-+ return 0;
-+}
-+
-+static int mx31ads_probe(struct platform_device *pdev)
-+{
-+ return 0;
-+}
-+
-+static int mx31ads_remove(struct platform_device *pdev)
-+{
-+ return 0;
-+}
-+
-+/* example machine audio_mapnections */
-+static const char* audio_map[][3] = {
-+
-+ /* mic is connected to mic1 - with bias */
-+ {"MIC1", NULL, "Mic Bias"},
-+ {"MIC1N", NULL, "Mic Bias"},
-+ {"Mic Bias", NULL, "Mic1 Jack"},
-+ {"Mic Bias", NULL, "Mic1 Jack"},
-+
-+ {"ACIN", NULL, "ACOP"},
-+ {NULL, NULL, NULL},
-+};
-+
-+/* headphone detect support on my board */
-+static const char * hp_pol[] = {"Headphone", "Speaker"};
-+static const struct soc_enum wm8753_enum =
-+ SOC_ENUM_SINGLE(WM8753_OUTCTL, 1, 2, hp_pol);
-+
-+static const struct snd_kcontrol_new wm8753_mx31ads_controls[] = {
-+ SOC_SINGLE("Headphone Detect Switch", WM8753_OUTCTL, 6, 1, 0),
-+ SOC_ENUM("Headphone Detect Polarity", wm8753_enum),
-+};
-+
-+/*
-+ * This is an example machine initialisation for a wm8753 connected to a
-+ * mx31ads II. It is missing logic to detect hp/mic insertions and logic
-+ * to re-route the audio in such an event.
-+ */
-+static int mx31ads_wm8753_init(struct snd_soc_codec *codec)
-+{
-+ int i, err;
-+
-+ /* set up mx31ads codec pins */
-+ snd_soc_dapm_set_endpoint(codec, "RXP", 0);
-+ snd_soc_dapm_set_endpoint(codec, "RXN", 0);
-+ snd_soc_dapm_set_endpoint(codec, "MIC2", 0);
-+
-+ /* add mx31ads specific controls */
-+ for (i = 0; i < ARRAY_SIZE(wm8753_mx31ads_controls); i++) {
-+ if ((err = snd_ctl_add(codec->card,
-+ snd_soc_cnew(&wm8753_mx31ads_controls[i],codec, NULL))) < 0)
-+ return err;
-+ }
-+
-+ /* set up mx31ads specific audio path audio_mapnects */
-+ for(i = 0; audio_map[i][0] != NULL; i++) {
-+ snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]);
-+ }
-+
-+ snd_soc_dapm_sync_endpoints(codec);
-+ return 0;
-+}
-+
-+static struct snd_soc_dai_link mx31ads_dai[] = {
-+{ /* Hifi Playback - for similatious use with voice below */
-+ .name = "WM8753",
-+ .stream_name = "WM8753 HiFi",
-+ .cpu_dai = &imx_ssi_pcm_dai[0],
-+ .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
-+ .init = mx31ads_wm8753_init,
-+ .ops = &mx31ads_hifi_ops,
-+},
-+//{ /* Voice via BT */
-+// .name = "Bluetooth",
-+// .stream_name = "Voice",
-+// .cpu_dai = &pxa_ssp_dai[1],
-+// .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
-+// .ops = &mx31ads_voice_ops,
-+//},
-+};
-+
-+static struct snd_soc_machine mx31ads = {
-+ .name = "mx31ads",
-+ .probe = mx31ads_probe,
-+ .remove = mx31ads_remove,
-+ .suspend_pre = mx31ads_suspend,
-+ .resume_post = mx31ads_resume,
-+ .dai_link = mx31ads_dai,
-+ .num_links = ARRAY_SIZE(mx31ads_dai),
-+};
-+
-+static struct wm8753_setup_data mx31ads_wm8753_setup = {
-+ .i2c_address = 0x1a,
-+};
-+
-+static struct snd_soc_device mx31ads_snd_devdata = {
-+ .machine = &mx31ads,
-+ .platform = &mxc_soc_platform,
-+ .codec_dev = &soc_codec_dev_wm8753,
-+ .codec_data = &mx31ads_wm8753_setup,
-+};
-+
-+static struct platform_device *mx31ads_snd_device;
-+
-+static int __init mx31ads_init(void)
-+{
-+ int ret;
-+
-+ mx31ads_snd_device = platform_device_alloc("soc-audio", -1);
-+ if (!mx31ads_snd_device)
-+ return -ENOMEM;
-+
-+ platform_set_drvdata(mx31ads_snd_device, &mx31ads_snd_devdata);
-+ mx31ads_snd_devdata.dev = &mx31ads_snd_device->dev;
-+ ret = platform_device_add(mx31ads_snd_device);
-+
-+ if (ret)
-+ platform_device_put(mx31ads_snd_device);
-+
-+ return ret;
-+}
-+
-+static void __exit mx31ads_exit(void)
-+{
-+ platform_device_unregister(mx31ads_snd_device);
-+}
-+
-+module_init(mx31ads_init);
-+module_exit(mx31ads_exit);
-+
-+/* Module information */
-+MODULE_AUTHOR("Liam Girdwood, liam.girdwood at wolfsonmicro.com, www.wolfsonmicro.com");
-+MODULE_DESCRIPTION("ALSA SoC WM8753 mx31ads");
-+MODULE_LICENSE("GPL");
-Index: linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/lm4857.h
-===================================================================
---- /dev/null 1970-01-01 00:00:00.000000000 +0000
-+++ linux-2.6.17.14-fic4.test/sound/soc/s3c24xx/lm4857.h 2007-01-31 21:36:05.000000000 +0100
-@@ -0,0 +1,15 @@
-+#ifndef LM4857_H_
-+#define LM4857_H_
-+
-+/* The register offsets in the cache array */
-+#define LM4857_MVOL 0
-+#define LM4857_LVOL 1
-+#define LM4857_RVOL 2
-+#define LM4857_CTRL 3
-+
-+/* the shifts required to set these bits */
-+#define LM4857_3D 5
-+#define LM4857_WAKEUP 5
-+#define LM4857_EPGAIN 4
-+
-+#endif /*LM4857_H_*/
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