Merge tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "As the diffstat shows we've had again a lot of works done for this
  cycle: the majority of changes are the continued componentization and
  code refactoring in ASoC, the tree-wide PCM API updates and cleanups
  and SOF updates while a few ASoC driver updates are seen, too.

  Here we go, some highlights:

  Core:
   - Finally y2038 support landed to ALSA ABI; some ioctls have been
     extended and lots of tricks were applied
   - Applying the new managed PCM buffer API to all drivers; the API
     itself was already merged in 5.5
   - The already deprecated dimension support in ALSA control API is
     dropped completely now
   - Verification of ALSA control elements to catch API misuses

  ASoC:
   - Further code refactorings and moving things to the component level
   - Lots of updates and improvements on SOF / Intel drivers; now
     including common HDMI driver and SoundWire support
   - New driver support for Ingenic JZ4770, Mediatek MT6660, Qualcomm
     WCD934x and WSA881x, and Realtek RT700, RT711, RT715, RT1011,
     RT1015 and RT1308

  HD-audio:
   - Improved ring-buffer communications using waitqueue
   - Drop the superfluous buffer preallocation on x86

  Others:
   - Many code cleanups, mostly constifications over the whole tree
   - USB-audio: quirks for MOTU, Corsair Virtuoso, Line6 Helix
   - FireWire: code refactoring for oxfw and dice drivers"

* tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (638 commits)
  ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
  ALSA: hda: Add Clevo W65_67SB the power_save blacklist
  ASoC: soc-core: remove null_snd_soc_ops
  ASoC: soc-pcm: add soc_rtd_trigger()
  ASoC: soc-pcm: add soc_rtd_hw_free()
  ASoC: soc-pcm: add soc_rtd_hw_params()
  ASoC: soc-pcm: add soc_rtd_prepare()
  ASoC: soc-pcm: add soc_rtd_shutdown()
  ASoC: soc-pcm: add soc_rtd_startup()
  ASoC: rt1015: add rt1015 amplifier driver
  ASoC: madera: Correct some kernel doc
  ASoC: topology: fix soc_tplg_fe_link_create() - link->dobj initialization order
  ASoC: Intel: skl_hda_dsp_common: Fix global-out-of-bounds bug
  ASoC: madera: Correct DMIC only input hook ups
  ALSA: cs46xx: fix spelling mistake "to" -> "too"
  ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
  ASoC: Add MediaTek MT6660 Speaker Amp Driver
  ASoC: dt-bindings: rt5645: add suppliers
  ASoC: max98090: fix deadlock in max98090_dapm_put_enum_double()
  ASoC: dapm: add snd_soc_dapm_put_enum_double_locked
  ...

1 files changed, 15 insertions, 12 deletions

diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c
index 829f95fc4179..7a13e9b35187 100644
--- a/sound/soc/intel/boards/kbl_da7219_max98927.c
+++ b/sound/soc/intel/boards/kbl_da7219_max98927.c
@@ -176,7 +176,7 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream,
 	struct snd_pcm_hw_params *params)
 {
 	struct snd_soc_pcm_runtime *runtime = substream->private_data;
-	int ret = 0, j;
+	int ret, j;
 
 	for (j = 0; j < runtime->num_codecs; j++) {
 		struct snd_soc_dai *codec_dai = runtime->codec_dais[j];
@@ -279,7 +279,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 {
 	struct snd_interval *rate = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_RATE);
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 			SNDRV_PCM_HW_PARAM_CHANNELS);
 	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
 	struct snd_soc_dpcm *dpcm = container_of(
@@ -298,7 +298,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 		!strcmp(rtd->card->name, "kblmax98373")) {
 		/* The ADSP will convert the FE rate to 48k, stereo */
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = DUAL_CHANNEL;
+		chan->min = chan->max = DUAL_CHANNEL;
 
 		/* set SSP to 24 bit */
 		snd_mask_none(fmt);
@@ -313,7 +313,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
 	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE);
 	}
@@ -341,6 +341,9 @@ static int kabylake_da7219_codec_init(struct snd_soc_pcm_runtime *rtd)
 			kabylake_ssp1_map,
 			ARRAY_SIZE(kabylake_ssp1_map));
 
+	if (ret)
+		return ret;
+
 	/*
 	 * Headset buttons map to the google Reference headset.
 	 * These can be configured by userspace.
@@ -491,7 +494,7 @@ static const struct snd_soc_ops kabylake_da7219_fe_ops = {
 static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 		struct snd_pcm_hw_params *params)
 {
-	struct snd_interval *channels = hw_param_interval(params,
+	struct snd_interval *chan = hw_param_interval(params,
 				SNDRV_PCM_HW_PARAM_CHANNELS);
 
 	/*
@@ -499,9 +502,9 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd,
 	 */
 
 	if (params_channels(params) == 2)
-		channels->min = channels->max = 2;
+		chan->min = chan->max = 2;
 	else
-		channels->min = channels->max = 4;
+		chan->min = chan->max = 4;
 
 	return 0;
 }
@@ -571,12 +574,12 @@ static struct snd_soc_ops skylake_refcap_ops = {
 static struct snd_soc_codec_conf max98927_codec_conf[] = {
 
 	{
-		.dev_name = MAX98927_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98927_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 
 	{
-		.dev_name = MAX98927_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98927_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -584,12 +587,12 @@ static struct snd_soc_codec_conf max98927_codec_conf[] = {
 static struct snd_soc_codec_conf max98373_codec_conf[] = {
 
 	{
-		.dev_name = MAX98373_DEV0_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV0_NAME),
 		.name_prefix = "Right",
 	},
 
 	{
-		.dev_name = MAX98373_DEV1_NAME,
+		.dlc = COMP_CODEC_CONF(MAX98373_DEV1_NAME),
 		.name_prefix = "Left",
 	},
 };
@@ -1092,7 +1095,7 @@ static int kabylake_audio_probe(struct platform_device *pdev)
 	struct kbl_codec_private *ctx;
 	struct snd_soc_dai_link *kbl_dai_link;
 	struct snd_soc_dai_link_component **codecs;
-	int i = 0;
+	int i;
 
 	ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL);
 	if (!ctx)