Merge tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This became wide and scattered updates all over the sound tree as
  diffstat shows: lots of (still ongoing) refactoring works in ASoC,
  fixes and cleanups caught by static analysis, inclusive term
  conversions as well as lots of new drivers. Below are highlights:

  ASoC core:
   - API cleanups and conversions to the unified mute_stream() call
   - Simplify I/O helper functions
   - Use helper macros to retrieve RTD from substreams

  ASoC drivers:
   - Lots of fixes and cleanups in Intel ASoC drivers
   - Lots of new stuff: Freescale MQS and i.MX6sx, Intel KeemBay I2S,
     Maxim MAX98360A and MAX98373 SoundWire, various Mediatek boards,
     nVidia Tegra 186 and 210, RealTek RL6231, Samsung Midas and Aries
     boards, TI J721e EVM

  ALSA core:
   - Minor code refacotring for SG-buffer handling

  HD-audio:
   - Generalization of mute-LED handling with LED classdev
   - Intel silent stream support for HDMI
   - Device-specific fixes: CA0132, Loongson-3

  Others:
   - Usual USB- and HD-audio quirks for various devices
   - Fixes for echoaudio DMA position handling
   - Various documents and trivial fixes for sparse warnings
   - Conversion to adopt inclusive terms"

* tag 'sound-5.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (479 commits)
  ALSA: pci: delete repeated words in comments
  ALSA: isa: delete repeated words in comments
  ALSA: hda/tegra: Add 100us dma stop delay
  ALSA: hda: Add dma stop delay variable
  ASoC: hda/tegra: Set buffer alignment to 128 bytes
  ALSA: seq: oss: Serialize ioctls
  ALSA: hda/hdmi: Add quirk to force connectivity
  ALSA: usb-audio: add startech usb audio dock name
  ALSA: usb-audio: Add support for Lenovo ThinkStation P620
  Revert "ALSA: hda: call runtime_allow() for all hda controllers"
  ALSA: hda/ca0132 - Fix AE-5 microphone selection commands.
  ALSA: hda/ca0132 - Add new quirk ID for Recon3D.
  ALSA: hda/ca0132 - Fix ZxR Headphone gain control get value.
  ALSA: hda/realtek: Add alc269/alc662 pin-tables for Loongson-3 laptops
  ALSA: docs: fix typo
  ALSA: doc: use correct config variable name
  ASoC: core: Two step component registration
  ASoC: core: Simplify snd_soc_component_initialize declaration
  ASoC: core: Relocate and expose snd_soc_component_initialize
  ASoC: sh: Replace 'select' DMADEVICES 'with depends on'
  ...

1 files changed, 17 insertions, 19 deletions

diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index aff57052a735..9b7b218f2a20 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -37,9 +37,6 @@
 #define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
 #define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
 #define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
-#define Q6ASM_DAI_TX_RX	0
-#define Q6ASM_DAI_TX	1
-#define Q6ASM_DAI_RX	2
 
 #define ALAC_CH_LAYOUT_MONO   ((101 << 16) | 1)
 #define ALAC_CH_LAYOUT_STEREO ((101 << 16) | 2)
@@ -215,9 +212,10 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 			     struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 	struct q6asm_dai_data *pdata;
+	struct device *dev = component->dev;
 	int ret, i;
 
 	pdata = snd_soc_component_get_drvdata(component);
@@ -225,7 +223,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 		return -EINVAL;
 
 	if (!prtd || !prtd->audio_client) {
-		pr_err("%s: private data null or audio client freed\n",
+		dev_err(dev, "%s: private data null or audio client freed\n",
 			__func__);
 		return -EINVAL;
 	}
@@ -248,7 +246,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 				       prtd->periods);
 
 	if (ret < 0) {
-		pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+		dev_err(dev, "Audio Start: Buffer Allocation failed rc = %d\n",
 							ret);
 		return -ENOMEM;
 	}
@@ -262,7 +260,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 	}
 
 	if (ret < 0) {
-		pr_err("%s: q6asm_open_write failed\n", __func__);
+		dev_err(dev, "%s: q6asm_open_write failed\n", __func__);
 		q6asm_audio_client_free(prtd->audio_client);
 		prtd->audio_client = NULL;
 		return -ENOMEM;
@@ -272,7 +270,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 	ret = q6routing_stream_open(soc_prtd->dai_link->id, LEGACY_PCM_MODE,
 			      prtd->session_id, substream->stream);
 	if (ret) {
-		pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+		dev_err(dev, "%s: stream reg failed ret:%d\n", __func__, ret);
 		return ret;
 	}
 
@@ -292,7 +290,7 @@ static int q6asm_dai_prepare(struct snd_soc_component *component,
 
 	}
 	if (ret < 0)
-		pr_info("%s: CMD Format block failed\n", __func__);
+		dev_info(dev, "%s: CMD Format block failed\n", __func__);
 
 	prtd->state = Q6ASM_STREAM_RUNNING;
 
@@ -332,7 +330,7 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 			  struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
 	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(soc_prtd, 0);
 	struct q6asm_dai_rtd *prtd;
 	struct q6asm_dai_data *pdata;
@@ -344,7 +342,7 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 
 	pdata = snd_soc_component_get_drvdata(component);
 	if (!pdata) {
-		pr_err("Drv data not found ..\n");
+		dev_err(dev, "Drv data not found ..\n");
 		return -EINVAL;
 	}
 
@@ -357,7 +355,7 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 				(q6asm_cb)event_handler, prtd, stream_id,
 				LEGACY_PCM_MODE);
 	if (IS_ERR(prtd->audio_client)) {
-		pr_info("%s: Could not allocate memory\n", __func__);
+		dev_info(dev, "%s: Could not allocate memory\n", __func__);
 		ret = PTR_ERR(prtd->audio_client);
 		kfree(prtd);
 		return ret;
@@ -372,12 +370,12 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 				SNDRV_PCM_HW_PARAM_RATE,
 				&constraints_sample_rates);
 	if (ret < 0)
-		pr_info("snd_pcm_hw_constraint_list failed\n");
+		dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
 	/* Ensure that buffer size is a multiple of period size */
 	ret = snd_pcm_hw_constraint_integer(runtime,
 					    SNDRV_PCM_HW_PARAM_PERIODS);
 	if (ret < 0)
-		pr_info("snd_pcm_hw_constraint_integer failed\n");
+		dev_info(dev, "snd_pcm_hw_constraint_integer failed\n");
 
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		ret = snd_pcm_hw_constraint_minmax(runtime,
@@ -385,21 +383,21 @@ static int q6asm_dai_open(struct snd_soc_component *component,
 			PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
 			PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
 		if (ret < 0) {
-			pr_err("constraint for buffer bytes min max ret = %d\n",
-									ret);
+			dev_err(dev, "constraint for buffer bytes min max ret = %d\n",
+				ret);
 		}
 	}
 
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
 		SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
 	if (ret < 0) {
-		pr_err("constraint for period bytes step ret = %d\n",
+		dev_err(dev, "constraint for period bytes step ret = %d\n",
 								ret);
 	}
 	ret = snd_pcm_hw_constraint_step(runtime, 0,
 		SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
 	if (ret < 0) {
-		pr_err("constraint for buffer bytes step ret = %d\n",
+		dev_err(dev, "constraint for buffer bytes step ret = %d\n",
 								ret);
 	}
 
@@ -424,7 +422,7 @@ static int q6asm_dai_close(struct snd_soc_component *component,
 			   struct snd_pcm_substream *substream)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
-	struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+	struct snd_soc_pcm_runtime *soc_prtd = asoc_substream_to_rtd(substream);
 	struct q6asm_dai_rtd *prtd = runtime->private_data;
 
 	if (prtd->audio_client) {