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

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (377 commits)
  ASoC: au1x: PSC-AC97 bugfixes
  ALSA: dummy - Increase MAX_PCM_SUBSTREAMS to 128
  ALSA: dummy - Add debug proc file
  ALSA: Add const prefix to proc helper functions
  ALSA: Re-export snd_pcm_format_name() function
  ALSA: hda - Use auto model for HP laptops with ALC268 codec
  ALSA: cs46xx - Fix minimum period size
  ASoC: Fix WM835x Out4 capture enumeration
  ALSA: Remove unneeded ifdef from sound/core.h
  ALSA: Remove struct snd_monitor_file from public sound/core.h
  ASoC: Remove unuused hw_read_t
  sound: oxygen: work around MCE when changing volume
  ALSA: dummy - Fake buffer allocations
  ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
  ASoC: fix pxa2xx-ac97.c breakage
  ALSA: dummy - Fix the timer calculation in systimer mode
  ALSA: dummy - Add more description
  ALSA: dummy - Better jiffies handling
  ALSA: dummy - Support high-res timer mode
  ALSA: Release v1.0.21
  ...

1 files changed, 24 insertions, 16 deletions

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 352d7eee9b6d..97ca9af414dc 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -27,8 +27,8 @@ struct snd_pcm_substream;
 #define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
 #define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
 #define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A		3 /* L data msb after FRM LRC */
-#define SND_SOC_DAIFMT_DSP_B		4 /* L data msb during FRM LRC */
+#define SND_SOC_DAIFMT_DSP_A		3 /* L data MSB after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B		4 /* L data MSB during FRM LRC */
 #define SND_SOC_DAIFMT_AC97		5 /* AC97 */
 
 /* left and right justified also known as MSB and LSB respectively */
@@ -38,7 +38,7 @@ struct snd_pcm_substream;
 /*
  * DAI Clock gating.
  *
- * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * DAI bit clocks can be be gated (disabled) when the DAI is not
  * sending or receiving PCM data in a frame. This can be used to save power.
  */
 #define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
@@ -51,21 +51,21 @@ struct snd_pcm_substream;
  * format.
  */
 #define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
-#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal bclk + inv frm */
-#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert bclk + nor frm */
-#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert bclk + frm */
+#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal BCLK + inv FRM */
+#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert BCLK + nor FRM */
+#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert BCLK + FRM */
 
 /*
  * DAI hardware clock masters.
  *
  * This is wrt the codec, the inverse is true for the interface
- * i.e. if the codec is clk and frm master then the interface is
+ * i.e. if the codec is clk and FRM master then the interface is
  * clk and frame slave.
  */
-#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & frm master */
-#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & FRM master */
+#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & FRM master */
 #define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & frm slave */
+#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & FRM slave */
 
 #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
@@ -78,7 +78,13 @@ struct snd_pcm_substream;
 #define SND_SOC_CLOCK_IN		0
 #define SND_SOC_CLOCK_OUT		1
 
-#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+			       SNDRV_PCM_FMTBIT_S16_LE |\
+			       SNDRV_PCM_FMTBIT_S16_BE |\
+			       SNDRV_PCM_FMTBIT_S20_3LE |\
+			       SNDRV_PCM_FMTBIT_S20_3BE |\
+			       SNDRV_PCM_FMTBIT_S24_3LE |\
+			       SNDRV_PCM_FMTBIT_S24_3BE |\
                                SNDRV_PCM_FMTBIT_S32_LE |\
                                SNDRV_PCM_FMTBIT_S32_BE)
 
@@ -106,7 +112,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
 
 int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
-	unsigned int mask, int slots);
+	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
 
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 
@@ -116,12 +122,12 @@ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
 /*
  * Digital Audio Interface.
  *
- * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
- * operations an capabilities. Codec and platfom drivers will register a this
+ * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
+ * operations and capabilities. Codec and platform drivers will register this
  * structure for every DAI they have.
  *
  * This structure covers the clocking, formating and ALSA operations for each
- * interface a
+ * interface.
  */
 struct snd_soc_dai_ops {
 	/*
@@ -140,7 +146,8 @@ struct snd_soc_dai_ops {
 	 */
 	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
 	int (*set_tdm_slot)(struct snd_soc_dai *dai,
-		unsigned int mask, int slots);
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width);
 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
 	/*
@@ -179,6 +186,7 @@ struct snd_soc_dai {
 	int ac97_control;
 
 	struct device *dev;
+	void *ac97_pdata;	/* platform_data for the ac97 codec */
 
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,