Merge tag 'sound-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "There have been lots of changes in ALSA core, HD-audio and ASoC, also
  most of PCI drivers touched by conversions of printks.  All these
  resulted in a high volume and wide ranged patch sets in this release.
  Many changes are fairly trivial, but also lots of nice cleanups and
  refactors.  There are a few new drivers, most notably, the Intel
  Haswell and Baytrail ASoC driver.

  Core changes:
   - A bit modernization; embed the device struct into snd_card struct,
     so that it may be referred from the beginning.  A new
     snd_card_new() function is introduced for that, and all drivers
     have been converted.

   - Simplification in the device management code in ALSA core; now
     managed by a simple priority list instead

   - Converted many kernel messages to use the standard dev_err() & co;
     this would be the pretty visible difference, especially for
     HD-audio.

  HD-audio:
   - Conexant codecs use the auto-parser as default now; the old static
     code still remains in case of regressions.  Some old quirks have
     been rewritten with the fixups for auto-parser.

   - C-Media codecs also use the auto-parser as default now, too.

   - A device struct is assigned to each HD-audio codec, and the
     formerly hwdep attributes are accessible over the codec sysfs, too.
     hwdep attributes still remain for compatibility.

   - Split the PCI-specific stuff for HD-audio controller into a
     separate module, ane make a helper module for the generic
     controller driver.  This is a preliminary change for supporting
     Tegra HDMI controller in near future, which slipped from 3.15
     merge.

   - Device-specific fixes: mute LED support for Lenovo Ideapad, mic LED
     fix for HP laptops, more ASUS subwoofer quirks, yet more Dell
     laptop headset quirks

   - Make the HD-audio codec response a bit more robust

   - A few improvements on Realtek ALC282 / 283 about the pop noises

   - A couple of Intel HDMI fixes

  ASoC:
   - Lots of cleanups for enumerations; refactored lots of error prone
     original codes to use more modern APIs

   - Elimination of the ASoC level wrappers for I2C and SPI moving us
     closer to converting to regmap completely and avoiding some
     randconfig hassle

   - Provide both manually and transparently locked DAPM APIs rather
     than a mix of the two fixing some concurrency issues

   - Start converting CODEC drivers to use separate bus interface
     drivers rather than having them all in one file helping avoid
     dependency issues

   - DPCM support for Intel Haswell and Bay Trail platforms, lots of
     fixes

   - Lots of work on improvements for simple-card, DaVinci and the
     Renesas rcar drivers.

   - New drivers for Analog Devices ADAU1977, TI PCM512x and parts of
     the CSR SiRF SoC, TLV320AIC31XXX, Armada 370 DB, Cirrus cs42xx8

   - Fixes for the simple-card DAI format DT mess

   - DT support for a couple more devices.

   - Use of the tdm_slot mapping in a few drivers

  Others:
   - Support of reset_resume callback for improved S4 in USB-audio
     driver; the device with boot quirks have been little tested, which
     we need to watch out in this development cycle

   - Add PM support for ICE1712 driver (finally!); it's still pretty
     partial support, only for M-Audio devices"

* tag 'sound-3.15-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (610 commits)
  ALSA: ice1712: Add suspend support for M-Audio ICE1712-based cards
  ALSA: ice1712: add suspend support for ICE1712 chip
  ALSA: hda - Enable beep for ASUS 1015E
  ALSA: asihpi: fix some indenting in snd_card_asihpi_pcm_new()
  ALSA: hda - add headset mic detect quirks for three Dell laptops
  ASoC: tegra: move AC97 clock handling to the machine driver
  ASoC: simple-card: Handle many DAI links
  ASoC: simple-card: Add DT documentation for multi-DAI links
  ASoC: simple-card: dynamically allocate the DAI link and properties
  ASoC: imx-ssi: Add .xlate_tdm_slot_mask() support.
  ASoC: fsl-esai: Add .xlate_tdm_slot_mask() support.
  ASoC: fsl-utils: Add fsl_asoc_xlate_tdm_slot_mask() support.
  ASoC: core: remove the 'of_' prefix of of_xlate_tdm_slot_mask.
  ASoC: rcar: subnode tidyup for renesas,rsnd.txt
  ASoC: Remove name_prefix unset during DAI link init hack
  ALSA: hda - Inform the unexpectedly ignored pins by auto-parser
  ASoC: rcar: bugfix: it cares about the non-src case
  ARM: bockw: fixup SND_SOC_DAIFMT_CBx_CFx flags
  ASoC: pcm: Drop incorrect double/extra frees
  ASoC: mfld_machine: Fix compile error
  ...

1 files changed, 85 insertions, 27 deletions

diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 28522bd03b8e..2cedf09f6d96 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -35,6 +35,86 @@
 #define DPCM_MAX_BE_USERS	8
 
 /**
+ * snd_soc_runtime_activate() - Increment active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is activated
+ * @stream: Direction of the PCM stream
+ *
+ * Increments the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is opened.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_activate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	lockdep_assert_held(&rtd->pcm_mutex);
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback_active++;
+		codec_dai->playback_active++;
+	} else {
+		cpu_dai->capture_active++;
+		codec_dai->capture_active++;
+	}
+
+	cpu_dai->active++;
+	codec_dai->active++;
+	cpu_dai->component->active++;
+	codec_dai->component->active++;
+}
+
+/**
+ * snd_soc_runtime_deactivate() - Decrement active count for PCM runtime components
+ * @rtd: ASoC PCM runtime that is deactivated
+ * @stream: Direction of the PCM stream
+ *
+ * Decrements the active count for all the DAIs and components attached to a PCM
+ * runtime. Should typically be called when a stream is closed.
+ *
+ * Must be called with the rtd->pcm_mutex being held
+ */
+void snd_soc_runtime_deactivate(struct snd_soc_pcm_runtime *rtd, int stream)
+{
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+
+	lockdep_assert_held(&rtd->pcm_mutex);
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		cpu_dai->playback_active--;
+		codec_dai->playback_active--;
+	} else {
+		cpu_dai->capture_active--;
+		codec_dai->capture_active--;
+	}
+
+	cpu_dai->active--;
+	codec_dai->active--;
+	cpu_dai->component->active--;
+	codec_dai->component->active--;
+}
+
+/**
+ * snd_soc_runtime_ignore_pmdown_time() - Check whether to ignore the power down delay
+ * @rtd: The ASoC PCM runtime that should be checked.
+ *
+ * This function checks whether the power down delay should be ignored for a
+ * specific PCM runtime. Returns true if the delay is 0, if it the DAI link has
+ * been configured to ignore the delay, or if none of the components benefits
+ * from having the delay.
+ */
+bool snd_soc_runtime_ignore_pmdown_time(struct snd_soc_pcm_runtime *rtd)
+{
+	if (!rtd->pmdown_time || rtd->dai_link->ignore_pmdown_time)
+		return true;
+
+	return rtd->cpu_dai->component->ignore_pmdown_time &&
+			rtd->codec_dai->component->ignore_pmdown_time;
+}
+
+/**
  * snd_soc_set_runtime_hwparams - set the runtime hardware parameters
  * @substream: the pcm substream
  * @hw: the hardware parameters
@@ -378,16 +458,9 @@ static int soc_pcm_open(struct snd_pcm_substream *substream)
 		 runtime->hw.rate_max);
 
 dynamic:
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active++;
-		codec_dai->playback_active++;
-	} else {
-		cpu_dai->capture_active++;
-		codec_dai->capture_active++;
-	}
-	cpu_dai->active++;
-	codec_dai->active++;
-	rtd->codec->active++;
+
+	snd_soc_runtime_activate(rtd, substream->stream);
+
 	mutex_unlock(&rtd->pcm_mutex);
 	return 0;
 
@@ -459,21 +532,10 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	struct snd_soc_platform *platform = rtd->platform;
 	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
 	struct snd_soc_dai *codec_dai = rtd->codec_dai;
-	struct snd_soc_codec *codec = rtd->codec;
 
 	mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass);
 
-	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		cpu_dai->playback_active--;
-		codec_dai->playback_active--;
-	} else {
-		cpu_dai->capture_active--;
-		codec_dai->capture_active--;
-	}
-
-	cpu_dai->active--;
-	codec_dai->active--;
-	codec->active--;
+	snd_soc_runtime_deactivate(rtd, substream->stream);
 
 	/* clear the corresponding DAIs rate when inactive */
 	if (!cpu_dai->active)
@@ -496,8 +558,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream)
 	cpu_dai->runtime = NULL;
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
-		if (!rtd->pmdown_time || codec->ignore_pmdown_time ||
-		    rtd->dai_link->ignore_pmdown_time) {
+		if (snd_soc_runtime_ignore_pmdown_time(rtd)) {
 			/* powered down playback stream now */
 			snd_soc_dapm_stream_event(rtd,
 						  SNDRV_PCM_STREAM_PLAYBACK,
@@ -1989,7 +2050,6 @@ int soc_dpcm_runtime_update(struct snd_soc_card *card)
 
 		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_PLAYBACK, &list);
 		if (paths < 0) {
-			dpcm_path_put(&list);
 			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
 					fe->dai_link->name,  "playback");
 			mutex_unlock(&card->mutex);
@@ -2019,7 +2079,6 @@ capture:
 
 		paths = dpcm_path_get(fe, SNDRV_PCM_STREAM_CAPTURE, &list);
 		if (paths < 0) {
-			dpcm_path_put(&list);
 			dev_warn(fe->dev, "ASoC: %s no valid %s path\n",
 					fe->dai_link->name,  "capture");
 			mutex_unlock(&card->mutex);
@@ -2084,7 +2143,6 @@ static int dpcm_fe_dai_open(struct snd_pcm_substream *fe_substream)
 	fe->dpcm[stream].runtime = fe_substream->runtime;
 
 	if (dpcm_path_get(fe, stream, &list) <= 0) {
-		dpcm_path_put(&list);
 		dev_dbg(fe->dev, "ASoC: %s no valid %s route\n",
 			fe->dai_link->name, stream ? "capture" : "playback");
 	}