Merge tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "Mostly many small changes spread as seen in diffstat in sound/*
  directory by this update.  A significant change in the subsystem level
  is the introduction of snd_soc_component, which will help more generic
  handling of SoC and off-SoC components.

  Also, snd_BUG_ON() macro is enabled unconditionally now due to its
  misuses, so people might hit kernel warnings (it's a good thing for
  us).

   - compress-offload: support for capture by Charles Keepax
   - HD-audio: codec delay support by Dylan Reid
   - HD-audio: improvements/fixes in generic parser: better headphone
     mic and headset mic support, jack_modes hint consolidation, proper
     beep attach/detachment, generalized power filter controls by David
     Henningsson, et al
   - HD-audio: Improved management of HDMI codec pins/converters
   - HD-audio: Better pin/DAC assignment for VIA codecs
   - HD-audio: Haswell HDMI workarounds
   - HD-audio: ALC268 codec support, a few new quirks for Chromebooks
   - USB: regression fixes: USB-MIDI autopm fix, the recent ISO latency
     fix by Clemens Ladisch
   - USB: support for DSD formats by Daniel Mack
   - USB: A few UAC2 device endian/cock fixes by Eldad Zack
   - USB: quirks for Emu 192kHz support, Novation Twitch DJ controller,
     Yamaha THRxx devices
   - HDSPM: updates for TCO controls by Adrian Knoth
   - ASoC: Add a snd_soc_component object type for generic handling of
     SoC and off-SoC components by Kuninori Morimoto,
   - dmaengine: a large set of cleanups and conversions by Lars-Peter
     Clausen
   - ASoC DAPM: performance optimizations from Ryo Tsutsui
   - ASoC DAPM: support for mixer control sharing by Stephen Warren
   - ASoC: multiplatform ARM cleanups from Arnd Bergmann
   - ASoC: new codec drivers for AK5385 and TAS5086 from Daniel Mack"

* tag 'sound-3.10' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (315 commits)
  ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
  ALSA: asihpi: add format support check in snd_card_asihpi_capture_formats
  ALSA: pcm_format_to_bits strong-typed conversion
  ALSA: compress: fix the states to check for allowing read
  ALSA: hda - Move Thinkpad X220 to use auto parser
  ALSA: USB: adjust for changed 3.8 USB API
  ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
  sound: oss/dmabuf: use dma_map_single
  ALSA: ali5451: use mdelay instead of large udelay constants
  ALSA: hda - Add the support for ALC286 codec
  ALSA: usb-audio: USB quirk for Yamaha THR10C
  ALSA: usb-audio: USB quirk for Yamaha THR5A
  ALSA: usb-audio: USB quirk for Yamaha THR10
  ALSA: usb-audio: Fix autopm error during probing
  ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
  ALSA: sound kconfig typo
  ALSA: emu10k1: Fix dock firmware loading
  ASoC: ux500: forward declare msp_i2s_platform_data
  ASoC: davinci-mcasp: Add Support BCLK-to-LRCLK ratio for TDM modes
  ASoC: davinci-pcm, davinci-mcasp: Clean up active_serializers
  ...

1 files changed, 76 insertions, 6 deletions

diff --git a/sound/pci/hda/patch_ca0132.c b/sound/pci/hda/patch_ca0132.c
index 0792b5725f9c..90ff7a3f72df 100644
--- a/sound/pci/hda/patch_ca0132.c
+++ b/sound/pci/hda/patch_ca0132.c
@@ -131,6 +131,13 @@ enum {
 /* Effects values size*/
 #define EFFECT_VALS_MAX_COUNT 12
 
+/* Latency introduced by DSP blocks in milliseconds. */
+#define DSP_CAPTURE_INIT_LATENCY        0
+#define DSP_CRYSTAL_VOICE_LATENCY       124
+#define DSP_PLAYBACK_INIT_LATENCY       13
+#define DSP_PLAY_ENHANCEMENT_LATENCY    30
+#define DSP_SPEAKER_OUT_LATENCY         7
+
 struct ct_effect {
 	char name[44];
 	hda_nid_t nid;
@@ -741,6 +748,9 @@ struct ca0132_spec {
 	long voicefx_val;
 	long cur_mic_boost;
 
+	struct hda_codec *codec;
+	struct delayed_work unsol_hp_work;
+
 #ifdef ENABLE_TUNING_CONTROLS
 	long cur_ctl_vals[TUNING_CTLS_COUNT];
 #endif
@@ -2740,6 +2750,31 @@ static int ca0132_playback_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
+static unsigned int ca0132_playback_pcm_delay(struct hda_pcm_stream *info,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int latency = DSP_PLAYBACK_INIT_LATENCY;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return 0;
+
+	/* Add latency if playback enhancement and either effect is enabled. */
+	if (spec->effects_switch[PLAY_ENHANCEMENT - EFFECT_START_NID]) {
+		if ((spec->effects_switch[SURROUND - EFFECT_START_NID]) ||
+		    (spec->effects_switch[DIALOG_PLUS - EFFECT_START_NID]))
+			latency += DSP_PLAY_ENHANCEMENT_LATENCY;
+	}
+
+	/* Applying Speaker EQ adds latency as well. */
+	if (spec->cur_out_type == SPEAKER_OUT)
+		latency += DSP_SPEAKER_OUT_LATENCY;
+
+	return (latency * runtime->rate) / 1000;
+}
+
 /*
  * Digital out
  */
@@ -2808,6 +2843,23 @@ static int ca0132_capture_pcm_cleanup(struct hda_pcm_stream *hinfo,
 	return 0;
 }
 
+static unsigned int ca0132_capture_pcm_delay(struct hda_pcm_stream *info,
+			struct hda_codec *codec,
+			struct snd_pcm_substream *substream)
+{
+	struct ca0132_spec *spec = codec->spec;
+	unsigned int latency = DSP_CAPTURE_INIT_LATENCY;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (spec->dsp_state != DSP_DOWNLOADED)
+		return 0;
+
+	if (spec->effects_switch[CRYSTAL_VOICE - EFFECT_START_NID])
+		latency += DSP_CRYSTAL_VOICE_LATENCY;
+
+	return (latency * runtime->rate) / 1000;
+}
+
 /*
  * Controls stuffs.
  */
@@ -3227,6 +3279,14 @@ exit:
 	return err < 0 ? err : 0;
 }
 
+static void ca0132_unsol_hp_delayed(struct work_struct *work)
+{
+	struct ca0132_spec *spec = container_of(
+		to_delayed_work(work), struct ca0132_spec, unsol_hp_work);
+	ca0132_select_out(spec->codec);
+	snd_hda_jack_report_sync(spec->codec);
+}
+
 static void ca0132_set_dmic(struct hda_codec *codec, int enable);
 static int ca0132_mic_boost_set(struct hda_codec *codec, long val);
 static int ca0132_effects_set(struct hda_codec *codec, hda_nid_t nid, long val);
@@ -3991,7 +4051,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_playback = {
 	.channels_max = 6,
 	.ops = {
 		.prepare = ca0132_playback_pcm_prepare,
-		.cleanup = ca0132_playback_pcm_cleanup
+		.cleanup = ca0132_playback_pcm_cleanup,
+		.get_delay = ca0132_playback_pcm_delay,
 	},
 };
 
@@ -4001,7 +4062,8 @@ static struct hda_pcm_stream ca0132_pcm_analog_capture = {
 	.channels_max = 2,
 	.ops = {
 		.prepare = ca0132_capture_pcm_prepare,
-		.cleanup = ca0132_capture_pcm_cleanup
+		.cleanup = ca0132_capture_pcm_cleanup,
+		.get_delay = ca0132_capture_pcm_delay,
 	},
 };
 
@@ -4399,8 +4461,7 @@ static void ca0132_process_dsp_response(struct hda_codec *codec)
 
 static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res)
 {
-	snd_printdd(KERN_INFO "ca0132_unsol_event: 0x%x\n", res);
-
+	struct ca0132_spec *spec = codec->spec;
 
 	if (((res >> AC_UNSOL_RES_TAG_SHIFT) & 0x3f) == UNSOL_TAG_DSP) {
 		ca0132_process_dsp_response(codec);
@@ -4412,8 +4473,13 @@ static void ca0132_unsol_event(struct hda_codec *codec, unsigned int res)
 
 		switch (res) {
 		case UNSOL_TAG_HP:
-			ca0132_select_out(codec);
-			snd_hda_jack_report_sync(codec);
+			/* Delay enabling the HP amp, to let the mic-detection
+			 * state machine run.
+			 */
+			cancel_delayed_work_sync(&spec->unsol_hp_work);
+			queue_delayed_work(codec->bus->workq,
+					   &spec->unsol_hp_work,
+					   msecs_to_jiffies(500));
 			break;
 		case UNSOL_TAG_AMIC1:
 			ca0132_select_mic(codec);
@@ -4588,6 +4654,7 @@ static void ca0132_free(struct hda_codec *codec)
 {
 	struct ca0132_spec *spec = codec->spec;
 
+	cancel_delayed_work_sync(&spec->unsol_hp_work);
 	snd_hda_power_up(codec);
 	snd_hda_sequence_write(codec, spec->base_exit_verbs);
 	ca0132_exit_chip(codec);
@@ -4653,6 +4720,7 @@ static int patch_ca0132(struct hda_codec *codec)
 	if (!spec)
 		return -ENOMEM;
 	codec->spec = spec;
+	spec->codec = codec;
 
 	spec->num_mixers = 1;
 	spec->mixers[0] = ca0132_mixer;
@@ -4663,6 +4731,8 @@ static int patch_ca0132(struct hda_codec *codec)
 	spec->init_verbs[1] = ca0132_init_verbs1;
 	spec->num_init_verbs = 2;
 
+	INIT_DELAYED_WORK(&spec->unsol_hp_work, ca0132_unsol_hp_delayed);
+
 	ca0132_init_chip(codec);
 
 	ca0132_config(codec);