---

yaml
---
r: 142718
b: refs/heads/master
c: 5f9c510
h: refs/heads/master
v: v3

[refs]

@@ -1,2 +1,2 @@
 ---
-refs/heads/master: fa075ed2dc80440bf3e9092d38a66c3227b174c1
+refs/heads/master: 5f9c510e9e18cd029e15190d35dd4271f2ef393b

trunk/Documentation/sound/alsa/soc/jack.txt

@@ -0,0 +1,71 @@
+ASoC jack detection
+===================
+
+ALSA has a standard API for representing physical jacks to user space,
+the kernel side of which can be seen in include/sound/jack.h.  ASoC
+provides a version of this API adding two additional features:
+
+ - It allows more than one jack detection method to work together on one
+   user visible jack.  In embedded systems it is common for multiple
+   to be present on a single jack but handled by separate bits of
+   hardware.
+
+ - Integration with DAPM, allowing DAPM endpoints to be updated
+   automatically based on the detected jack status (eg, turning off the
+   headphone outputs if no headphones are present).
+
+This is done by splitting the jacks up into three things working
+together: the jack itself represented by a struct snd_soc_jack, sets of
+snd_soc_jack_pins representing DAPM endpoints to update and blocks of
+code providing jack reporting mechanisms.
+
+For example, a system may have a stereo headset jack with two reporting
+mechanisms, one for the headphone and one for the microphone.  Some
+systems won't be able to use their speaker output while a headphone is
+connected and so will want to make sure to update both speaker and
+headphone when the headphone jack status changes.
+
+The jack - struct snd_soc_jack
+==============================
+
+This represents a physical jack on the system and is what is visible to
+user space.  The jack itself is completely passive, it is set up by the
+machine driver and updated by jack detection methods.
+
+Jacks are created by the machine driver calling snd_soc_jack_new().
+
+snd_soc_jack_pin
+================
+
+These represent a DAPM pin to update depending on some of the status
+bits supported by the jack.  Each snd_soc_jack has zero or more of these
+which are updated automatically.  They are created by the machine driver
+and associated with the jack using snd_soc_jack_add_pins().  The status
+of the endpoint may configured to be the opposite of the jack status if
+required (eg, enabling a built in microphone if a microphone is not
+connected via a jack).
+
+Jack detection methods
+======================
+
+Actual jack detection is done by code which is able to monitor some
+input to the system and update a jack by calling snd_soc_jack_report(),
+specifying a subset of bits to update.  The jack detection code should
+be set up by the machine driver, taking configuration for the jack to
+update and the set of things to report when the jack is connected.
+
+Often this is done based on the status of a GPIO - a handler for this is
+provided by the snd_soc_jack_add_gpio() function.  Other methods are
+also available, for example integrated into CODECs.  One example of
+CODEC integrated jack detection can be see in the WM8350 driver.
+
+Each jack may have multiple reporting mechanisms, though it will need at
+least one to be useful.
+
+Machine drivers
+===============
+
+These are all hooked together by the machine driver depending on the
+system hardware.  The machine driver will set up the snd_soc_jack and
+the list of pins to update then set up one or more jack detection
+mechanisms to update that jack based on their current status.

trunk/sound/arm/pxa2xx-ac97-lib.c

@@ -238,6 +238,8 @@ static inline void pxa_ac97_cold_pxa3xx(void)
 
 bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 {
+	unsigned long gsr;
+
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
 		pxa_ac97_warm_pxa25x();
@@ -254,10 +256,10 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97)
 	else
 #endif
 		BUG();
-
-	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+	gsr = GSR | gsr_bits;
+	if (!(gsr & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n",
-				 __func__, gsr_bits);
+				 __func__, gsr);
 
 		return false;
 	}
@@ -268,6 +270,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset);
 
 bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 {
+	unsigned long gsr;
+
 #ifdef CONFIG_PXA25x
 	if (cpu_is_pxa25x())
 		pxa_ac97_cold_pxa25x();
@@ -285,9 +289,10 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97)
 #endif
 		BUG();
 
-	if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) {
+	gsr = GSR | gsr_bits;
+	if (!(gsr & (GSR_PCR | GSR_SCR))) {
 		printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n",
-				 __func__, gsr_bits);
+				 __func__, gsr);
 
 		return false;
 	}

trunk/sound/isa/opl3sa2.c

@@ -481,6 +481,7 @@ OPL3SA2_DOUBLE_TLV("Master Playback Volume", 0, 0x07, 0x08, 0, 0, 15, 1,
 OPL3SA2_SINGLE("Mic Playback Switch", 0, 0x09, 7, 1, 1),
 OPL3SA2_SINGLE_TLV("Mic Playback Volume", 0, 0x09, 0, 31, 1,
 		   db_scale_5bit_12db_max),
+OPL3SA2_SINGLE("ZV Port Switch", 0, 0x02, 0, 1, 0),
 };
 
 static struct snd_kcontrol_new snd_opl3sa2_tone_controls[] = {

trunk/sound/pci/hda/patch_realtek.c

@@ -8764,6 +8764,10 @@ static struct snd_pci_quirk alc883_cfg_tbl[] = {
 	{}
 };
 
+static hda_nid_t alc883_slave_dig_outs[] = {
+	ALC1200_DIGOUT_NID, 0,
+};
+
 static hda_nid_t alc1200_slave_dig_outs[] = {
 	ALC883_DIGOUT_NID, 0,
 };
@@ -8809,6 +8813,7 @@ static struct alc_config_preset alc883_presets[] = {
 		.dac_nids = alc883_dac_nids,
 		.dig_out_nid = ALC883_DIGOUT_NID,
 		.dig_in_nid = ALC883_DIGIN_NID,
+		.slave_dig_outs = alc883_slave_dig_outs,
 		.num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_intel_modes),
 		.channel_mode = alc883_3ST_6ch_intel_modes,
 		.need_dac_fix = 1,

trunk/sound/pci/hda/patch_sigmatel.c

@@ -4895,6 +4895,7 @@ static int patch_stac92hd83xxx(struct hda_codec *codec)
 	switch (codec->vendor_id) {
 	case 0x111d7604:
 	case 0x111d7605:
+	case 0x111d76d5:
 		if (spec->board_config == STAC_92HD83XXX_PWR_REF)
 			break;
 		spec->num_pwrs = 0;
@@ -5707,6 +5708,7 @@ static struct hda_codec_preset snd_hda_preset_sigmatel[] = {
 	{ .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7604, .name = "92HD83C1X5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7605, .name = "92HD81B1X5", .patch = patch_stac92hd83xxx},
+	{ .id = 0x111d76d5, .name = "92HD81B1C5", .patch = patch_stac92hd83xxx},
 	{ .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx},
 	{ .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx },
 	{ .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx },

trunk/sound/ppc/powermac.c

@@ -51,7 +51,7 @@ static struct platform_device *device;
 /*
  */
 
-static int __init snd_pmac_probe(struct platform_device *devptr)
+static int __devinit snd_pmac_probe(struct platform_device *devptr)
 {
 	struct snd_card *card;
 	struct snd_pmac *chip;

trunk/sound/soc/codecs/twl4030.c

@@ -122,6 +122,9 @@ struct twl4030_priv {
 	unsigned int bypass_state;
 	unsigned int codec_powered;
 	unsigned int codec_muted;
+
+	struct snd_pcm_substream *master_substream;
+	struct snd_pcm_substream *slave_substream;
 };
 
 /*
@@ -1217,15 +1220,64 @@ static int twl4030_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
+static int twl4030_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct twl4030_priv *twl4030 = codec->private_data;
+
+	/* If we already have a playback or capture going then constrain
+	 * this substream to match it.
+	 */
+	if (twl4030->master_substream) {
+		struct snd_pcm_runtime *master_runtime;
+		master_runtime = twl4030->master_substream->runtime;
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_RATE,
+					     master_runtime->rate,
+					     master_runtime->rate);
+
+		snd_pcm_hw_constraint_minmax(substream->runtime,
+					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+					     master_runtime->sample_bits,
+					     master_runtime->sample_bits);
+
+		twl4030->slave_substream = substream;
+	} else
+		twl4030->master_substream = substream;
+
+	return 0;
+}
+
+static void twl4030_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_device *socdev = rtd->socdev;
+	struct snd_soc_codec *codec = socdev->codec;
+	struct twl4030_priv *twl4030 = codec->private_data;
+
+	if (twl4030->master_substream == substream)
+		twl4030->master_substream = twl4030->slave_substream;
+
+	twl4030->slave_substream = NULL;
+}
+
 static int twl4030_hw_params(struct snd_pcm_substream *substream,
 			   struct snd_pcm_hw_params *params,
 			   struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
+	struct twl4030_priv *twl4030 = codec->private_data;
 	u8 mode, old_mode, format, old_format;
 
+	if (substream == twl4030->slave_substream)
+		/* Ignoring hw_params for slave substream */
+		return 0;
+
 	/* bit rate */
 	old_mode = twl4030_read_reg_cache(codec,
 			TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ;
@@ -1259,6 +1311,9 @@ static int twl4030_hw_params(struct snd_pcm_substream *substream,
 	case 48000:
 		mode |= TWL4030_APLL_RATE_48000;
 		break;
+	case 96000:
+		mode |= TWL4030_APLL_RATE_96000;
+		break;
 	default:
 		printk(KERN_ERR "TWL4030 hw params: unknown rate %d\n",
 			params_rate(params));
@@ -1384,6 +1439,8 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai,
 #define TWL4030_FORMATS	 (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
 
 static struct snd_soc_dai_ops twl4030_dai_ops = {
+	.startup	= twl4030_startup,
+	.shutdown	= twl4030_shutdown,
 	.hw_params	= twl4030_hw_params,
 	.set_sysclk	= twl4030_set_dai_sysclk,
 	.set_fmt	= twl4030_set_dai_fmt,
@@ -1395,7 +1452,7 @@ struct snd_soc_dai twl4030_dai = {
 		.stream_name = "Playback",
 		.channels_min = 2,
 		.channels_max = 2,
-		.rates = TWL4030_RATES,
+		.rates = TWL4030_RATES | SNDRV_PCM_RATE_96000,
 		.formats = TWL4030_FORMATS,},
 	.capture = {
 		.stream_name = "Capture",

trunk/sound/soc/codecs/twl4030.h

@@ -109,6 +109,7 @@
 #define TWL4030_APLL_RATE_32000		0x80
 #define TWL4030_APLL_RATE_44100		0x90
 #define TWL4030_APLL_RATE_48000		0xA0
+#define TWL4030_APLL_RATE_96000		0xE0
 #define TWL4030_SEL_16K			0x04
 #define TWL4030_CODECPDZ		0x02
 #define TWL4030_OPT_MODE		0x01

trunk/sound/soc/codecs/wm9705.c

@@ -317,6 +317,41 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 	return -EIO;
 }
 
+#ifdef CONFIG_PM
+static int wm9705_soc_suspend(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+
+	soc_ac97_ops.write(codec->ac97, AC97_POWERDOWN, 0xffff);
+
+	return 0;
+}
+
+static int wm9705_soc_resume(struct platform_device *pdev)
+{
+	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+	struct snd_soc_codec *codec = socdev->card->codec;
+	int i, ret;
+	u16 *cache = codec->reg_cache;
+
+	ret = wm9705_reset(codec);
+	if (ret < 0) {
+		printk(KERN_ERR "could not reset AC97 codec\n");
+		return ret;
+	}
+
+	for (i = 2; i < ARRAY_SIZE(wm9705_reg) << 1; i += 2) {
+		soc_ac97_ops.write(codec->ac97, i, cache[i>>1]);
+	}
+
+	return 0;
+}
+#else
+#define wm9705_soc_suspend NULL
+#define wm9705_soc_resume NULL
+#endif
+
 static int wm9705_soc_probe(struct platform_device *pdev)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
@@ -407,6 +442,8 @@ static int wm9705_soc_remove(struct platform_device *pdev)
 struct snd_soc_codec_device soc_codec_dev_wm9705 = {
 	.probe = 	wm9705_soc_probe,
 	.remove = 	wm9705_soc_remove,
+	.suspend =	wm9705_soc_suspend,
+	.resume =	wm9705_soc_resume,
 };
 EXPORT_SYMBOL_GPL(soc_codec_dev_wm9705);
 

trunk/sound/soc/fsl/fsl_dma.c

@@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream)
 	else
 		position = in_be32(&dma_channel->dar);
 
+	/*
+	 * When capture is started, the SSI immediately starts to fill its FIFO.
+	 * This means that the DMA controller is not started until the FIFO is
+	 * full.  However, ALSA calls this function before that happens, when
+	 * MR.DAR is still zero.  In this case, just return zero to indicate
+	 * that nothing has been received yet.
+	 */
+	if (!position)
+		return 0;
+
+	if ((position < dma_private->dma_buf_phys) ||
+	    (position > dma_private->dma_buf_end)) {
+		dev_err(substream->pcm->card->dev,
+			"dma pointer is out of range, halting stream\n");
+		return SNDRV_PCM_POS_XRUN;
+	}
+
 	frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys);
 
 	/*