Merge branch 'topic/usb-audio-refactoring' into for-next

Pull the USB audio improvement

Signed-off-by: Takashi Iwai <tiwai@suse.de>

8 files changed, 60 insertions, 41 deletions

diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig
index d5bae5d1ab6f..a5b446d5af19 100644
--- a/sound/soc/intel/Kconfig
+++ b/sound/soc/intel/Kconfig
@@ -15,22 +15,6 @@ config SND_SOC_INTEL_SST_TOPLEVEL
 
 if SND_SOC_INTEL_SST_TOPLEVEL
 
-config SND_SST_IPC
-	tristate
-	# This option controls the IPC core for HiFi2 platforms
-
-config SND_SST_IPC_PCI
-	tristate
-	select SND_SST_IPC
-	# This option controls the PCI-based IPC for HiFi2 platforms
-	#  (Medfield, Merrifield).
-
-config SND_SST_IPC_ACPI
-	tristate
-	select SND_SST_IPC
-	# This option controls the ACPI-based IPC for HiFi2 platforms
-	# (Baytrail, Cherrytrail)
-
 config SND_SOC_INTEL_SST
 	tristate
 
@@ -57,7 +41,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM
 config SND_SST_ATOM_HIFI2_PLATFORM_PCI
 	tristate "PCI HiFi2 (Merrifield) Platforms"
 	depends on X86 && PCI
-	select SND_SST_IPC_PCI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	help
 	  If you have a Intel Merrifield/Edison platform, then
@@ -70,7 +53,6 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI
 	tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms"
 	default ACPI
 	depends on X86 && ACPI && PCI
-	select SND_SST_IPC_ACPI
 	select SND_SST_ATOM_HIFI2_PLATFORM
 	select SND_SOC_ACPI_INTEL_MATCH
 	select IOSF_MBI
diff --git a/sound/soc/intel/atom/Makefile b/sound/soc/intel/atom/Makefile
index a9326d5ec44c..c66f03f5d8d6 100644
--- a/sound/soc/intel/atom/Makefile
+++ b/sound/soc/intel/atom/Makefile
@@ -6,4 +6,4 @@ snd-soc-sst-atom-hifi2-platform-objs :=	sst-mfld-platform-pcm.o \
 obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-soc-sst-atom-hifi2-platform.o
 
 # DSP driver
-obj-$(CONFIG_SND_SST_IPC) += sst/
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += sst/
diff --git a/sound/soc/intel/atom/sst/Makefile b/sound/soc/intel/atom/sst/Makefile
index f17c905df3e2..5761d30a5f9d 100644
--- a/sound/soc/intel/atom/sst/Makefile
+++ b/sound/soc/intel/atom/sst/Makefile
@@ -3,6 +3,6 @@ snd-intel-sst-core-objs := sst.o sst_ipc.o sst_stream.o sst_drv_interface.o sst_
 snd-intel-sst-pci-objs += sst_pci.o
 snd-intel-sst-acpi-objs += sst_acpi.o
 
-obj-$(CONFIG_SND_SST_IPC) += snd-intel-sst-core.o
-obj-$(CONFIG_SND_SST_IPC_PCI) += snd-intel-sst-pci.o
-obj-$(CONFIG_SND_SST_IPC_ACPI) += snd-intel-sst-acpi.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM) += snd-intel-sst-core.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_PCI) += snd-intel-sst-pci.o
+obj-$(CONFIG_SND_SST_ATOM_HIFI2_PLATFORM_ACPI) += snd-intel-sst-acpi.o
diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c
index 3ea4602dfb3e..9a4b3d0973f6 100644
--- a/sound/soc/intel/boards/kbl_rt5663_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c
@@ -401,17 +401,40 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	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(
-			params, struct snd_soc_dpcm, hw_params);
-	struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link;
-	struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link;
+	struct snd_soc_dpcm *dpcm, *rtd_dpcm = NULL;
+
+	/*
+	 * The following loop will be called only for playback stream
+	 * In this platform, there is only one playback device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_PLAYBACK, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	/*
+	 * This following loop will be called only for capture stream
+	 * In this platform, there is only one capture device on every SSP
+	 */
+	for_each_dpcm_fe(rtd, SNDRV_PCM_STREAM_CAPTURE, dpcm) {
+		rtd_dpcm = dpcm;
+		break;
+	}
+
+	if (!rtd_dpcm)
+		return -EINVAL;
+
+	/*
+	 * The above 2 loops are mutually exclusive based on the stream direction,
+	 * thus rtd_dpcm variable will never be overwritten
+	 */
 
 	/*
 	 * The ADSP will convert the FE rate to 48k, stereo, 24 bit
 	 */
-	if (!strcmp(fe_dai_link->name, "Kbl Audio Port") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Headset Playback") ||
-	    !strcmp(fe_dai_link->name, "Kbl Audio Capture Port")) {
+	if (!strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Port") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Headset Playback") ||
+	    !strcmp(rtd_dpcm->fe->dai_link->name, "Kbl Audio Capture Port")) {
 		rate->min = rate->max = 48000;
 		chan->min = chan->max = 2;
 		snd_mask_none(fmt);
@@ -421,7 +444,7 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd,
 	 * The speaker on the SSP0 supports S16_LE and not S24_LE.
 	 * thus changing the mask here
 	 */
-	if (!strcmp(be_dai_link->name, "SSP0-Codec"))
+	if (!strcmp(rtd_dpcm->be->dai_link->name, "SSP0-Codec"))
 		snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
 
 	return 0;
diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
index 922cd0176e1f..f95546c184aa 100644
--- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
+++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c
@@ -700,6 +700,8 @@ static int kabylake_set_bias_level(struct snd_soc_card *card,
 	switch (level) {
 	case SND_SOC_BIAS_PREPARE:
 		if (dapm->bias_level == SND_SOC_BIAS_ON) {
+			if (!__clk_is_enabled(priv->mclk))
+				return 0;
 			dev_dbg(card->dev, "Disable mclk");
 			clk_disable_unprepare(priv->mclk);
 		} else {
diff --git a/sound/soc/intel/catpt/dsp.c b/sound/soc/intel/catpt/dsp.c
index 7d2968571951..9e807b941732 100644
--- a/sound/soc/intel/catpt/dsp.c
+++ b/sound/soc/intel/catpt/dsp.c
@@ -267,9 +267,12 @@ static int catpt_dsp_select_lpclock(struct catpt_dev *cdev, bool lp, bool waiti)
 					    reg, (reg & CATPT_ISD_DCPWM),
 					    500, 10000);
 		if (ret) {
-			dev_err(cdev->dev, "await WAITI timeout\n");
-			mutex_unlock(&cdev->clk_mutex);
-			return ret;
+			dev_warn(cdev->dev, "await WAITI timeout\n");
+			/* no signal - only high clock selection allowed */
+			if (lp) {
+				mutex_unlock(&cdev->clk_mutex);
+				return 0;
+			}
 		}
 	}
 
diff --git a/sound/soc/intel/catpt/pcm.c b/sound/soc/intel/catpt/pcm.c
index f78018c857b8..408e64e3b5fb 100644
--- a/sound/soc/intel/catpt/pcm.c
+++ b/sound/soc/intel/catpt/pcm.c
@@ -458,10 +458,6 @@ static int catpt_dai_prepare(struct snd_pcm_substream *substream,
 	if (ret)
 		return CATPT_IPC_ERROR(ret);
 
-	ret = catpt_dsp_update_lpclock(cdev);
-	if (ret)
-		return ret;
-
 	ret = catpt_dai_apply_usettings(dai, stream);
 	if (ret)
 		return ret;
@@ -500,6 +496,7 @@ static int catpt_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 	resume_stream:
+		catpt_dsp_update_lpclock(cdev);
 		ret = catpt_ipc_resume_stream(cdev, stream->info.stream_hw_id);
 		if (ret)
 			return CATPT_IPC_ERROR(ret);
@@ -507,11 +504,11 @@ static int catpt_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 
 	case SNDRV_PCM_TRIGGER_STOP:
 		stream->prepared = false;
-		catpt_dsp_update_lpclock(cdev);
 		fallthrough;
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
 		ret = catpt_ipc_pause_stream(cdev, stream->info.stream_hw_id);
+		catpt_dsp_update_lpclock(cdev);
 		if (ret)
 			return CATPT_IPC_ERROR(ret);
 		break;
@@ -534,6 +531,8 @@ void catpt_stream_update_position(struct catpt_dev *cdev,
 
 	dsppos = bytes_to_frames(r, pos->stream_position);
 
+	if (!stream->prepared)
+		goto exit;
 	/* only offload is set_write_pos driven */
 	if (stream->template->type != CATPT_STRM_TYPE_RENDER)
 		goto exit;
@@ -667,7 +666,17 @@ static int catpt_dai_pcm_new(struct snd_soc_pcm_runtime *rtm,
 		break;
 	}
 
+	/* see if this is a new configuration */
+	if (!memcmp(&cdev->devfmt[devfmt.iface], &devfmt, sizeof(devfmt)))
+		return 0;
+
+	pm_runtime_get_sync(cdev->dev);
+
 	ret = catpt_ipc_set_device_format(cdev, &devfmt);
+
+	pm_runtime_mark_last_busy(cdev->dev);
+	pm_runtime_put_autosuspend(cdev->dev);
+
 	if (ret)
 		return CATPT_IPC_ERROR(ret);
 
diff --git a/sound/soc/intel/keembay/kmb_platform.c b/sound/soc/intel/keembay/kmb_platform.c
index f54b710ee1c2..291a686568c2 100644
--- a/sound/soc/intel/keembay/kmb_platform.c
+++ b/sound/soc/intel/keembay/kmb_platform.c
@@ -487,9 +487,9 @@ static int kmb_dai_hw_params(struct snd_pcm_substream *substream,
 		kmb_i2s->xfer_resolution = 0x02;
 		break;
 	case SNDRV_PCM_FORMAT_S24_LE:
-		config->data_width = 24;
-		kmb_i2s->ccr = 0x08;
-		kmb_i2s->xfer_resolution = 0x04;
+		config->data_width = 32;
+		kmb_i2s->ccr = 0x14;
+		kmb_i2s->xfer_resolution = 0x05;
 		break;
 	case SNDRV_PCM_FORMAT_S32_LE:
 		config->data_width = 32;