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
  ...

5 files changed, 79 insertions, 70 deletions

diff --git a/sound/firewire/oxfw/oxfw-command.c b/sound/firewire/oxfw/oxfw-command.c
index 16dc337c7093..d2e57c76070d 100644
--- a/sound/firewire/oxfw/oxfw-command.c
+++ b/sound/firewire/oxfw/oxfw-command.c
@@ -38,7 +38,7 @@ int avc_stream_set_format(struct fw_unit *unit, enum avc_general_plug_dir dir,
 	else if (err < len + 10)
 		err = -EIO;
 	else if (buf[0] == 0x08) /* NOT IMPLEMENTED */
-		err = -ENOSYS;
+		err = -ENXIO;
 	else if (buf[0] == 0x0a) /* REJECTED */
 		err = -EINVAL;
 	else
@@ -83,7 +83,7 @@ int avc_stream_get_format(struct fw_unit *unit,
 	else if (err < 12)
 		err = -EIO;
 	else if (buf[0] == 0x08)	/* NOT IMPLEMENTED */
-		err = -ENOSYS;
+		err = -ENXIO;
 	else if (buf[0] == 0x0a)	/* REJECTED */
 		err = -EINVAL;
 	else if (buf[0] == 0x0b)	/* IN TRANSITION */
@@ -147,7 +147,7 @@ int avc_general_inquiry_sig_fmt(struct fw_unit *unit, unsigned int rate,
 	else if (err < 8)
 		err = -EIO;
 	else if (buf[0] == 0x08)	/* NOT IMPLEMENTED */
-		err = -ENOSYS;
+		err = -ENXIO;
 	if (err < 0)
 		goto end;
 
diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c
index 67fd3e844dd6..2dfa7e179cb6 100644
--- a/sound/firewire/oxfw/oxfw-pcm.c
+++ b/sound/firewire/oxfw/oxfw-pcm.c
@@ -237,11 +237,7 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream,
 				 struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_oxfw *oxfw = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
+	int err = 0;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		unsigned int rate = params_rate(hw_params);
@@ -264,11 +260,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream,
 				  struct snd_pcm_hw_params *hw_params)
 {
 	struct snd_oxfw *oxfw = substream->private_data;
-	int err;
-
-	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
-	if (err < 0)
-		return err;
+	int err = 0;
 
 	if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) {
 		unsigned int rate = params_rate(hw_params);
@@ -301,7 +293,7 @@ static int pcm_capture_hw_free(struct snd_pcm_substream *substream)
 
 	mutex_unlock(&oxfw->mutex);
 
-	return snd_pcm_lib_free_pages(substream);
+	return 0;
 }
 static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
 {
@@ -316,7 +308,7 @@ static int pcm_playback_hw_free(struct snd_pcm_substream *substream)
 
 	mutex_unlock(&oxfw->mutex);
 
-	return snd_pcm_lib_free_pages(substream);
+	return 0;
 }
 
 static int pcm_capture_prepare(struct snd_pcm_substream *substream)
@@ -419,7 +411,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 	static const struct snd_pcm_ops capture_ops = {
 		.open      = pcm_open,
 		.close     = pcm_close,
-		.ioctl     = snd_pcm_lib_ioctl,
 		.hw_params = pcm_capture_hw_params,
 		.hw_free   = pcm_capture_hw_free,
 		.prepare   = pcm_capture_prepare,
@@ -430,7 +421,6 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 	static const struct snd_pcm_ops playback_ops = {
 		.open      = pcm_open,
 		.close     = pcm_close,
-		.ioctl     = snd_pcm_lib_ioctl,
 		.hw_params = pcm_playback_hw_params,
 		.hw_free   = pcm_playback_hw_free,
 		.prepare   = pcm_playback_prepare,
@@ -454,8 +444,7 @@ int snd_oxfw_create_pcm(struct snd_oxfw *oxfw)
 	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &playback_ops);
 	if (cap > 0)
 		snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &capture_ops);
-	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_VMALLOC,
-					      NULL, 0, 0);
+	snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_VMALLOC, NULL, 0, 0);
 
 	return 0;
 }
diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c
index 501a80094bf7..80c9dc13f1b5 100644
--- a/sound/firewire/oxfw/oxfw-stream.c
+++ b/sound/firewire/oxfw/oxfw-stream.c
@@ -513,7 +513,7 @@ int snd_oxfw_stream_parse_format(u8 *format,
 	 *  Level 1:	AM824 Compound  (0x40)
 	 */
 	if ((format[0] != 0x90) || (format[1] != 0x40))
-		return -ENOSYS;
+		return -ENXIO;
 
 	/* check the sampling rate */
 	for (i = 0; i < ARRAY_SIZE(avc_stream_rate_table); i++) {
@@ -521,7 +521,7 @@ int snd_oxfw_stream_parse_format(u8 *format,
 			break;
 	}
 	if (i == ARRAY_SIZE(avc_stream_rate_table))
-		return -ENOSYS;
+		return -ENXIO;
 
 	formation->rate = oxfw_rate_table[i];
 
@@ -565,13 +565,13 @@ int snd_oxfw_stream_parse_format(u8 *format,
 		/* Don't care */
 		case 0xff:
 		default:
-			return -ENOSYS;	/* not supported */
+			return -ENXIO;	/* not supported */
 		}
 	}
 
 	if (formation->pcm  > AM824_MAX_CHANNELS_FOR_PCM ||
 	    formation->midi > AM824_MAX_CHANNELS_FOR_MIDI)
-		return -ENOSYS;
+		return -ENXIO;
 
 	return 0;
 }
@@ -656,7 +656,7 @@ static int fill_stream_formats(struct snd_oxfw *oxfw,
 	/* get first entry */
 	len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
 	err = avc_stream_get_format_list(oxfw->unit, dir, 0, buf, &len, 0);
-	if (err == -ENOSYS) {
+	if (err == -ENXIO) {
 		/* LIST subfunction is not implemented */
 		len = AVC_GENERIC_FRAME_MAXIMUM_BYTES;
 		err = assume_stream_formats(oxfw, dir, pid, buf, &len,
@@ -728,49 +728,63 @@ int snd_oxfw_stream_discover(struct snd_oxfw *oxfw)
 			err);
 		goto end;
 	} else if ((plugs[0] == 0) && (plugs[1] == 0)) {
-		err = -ENOSYS;
+		err = -ENXIO;
 		goto end;
 	}
 
 	/* use oPCR[0] if exists */
 	if (plugs[1] > 0) {
 		err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_OUT, 0);
-		if (err < 0)
-			goto end;
+		if (err < 0) {
+			if (err != -ENXIO)
+				return err;
 
-		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
-			format = oxfw->tx_stream_formats[i];
-			if (format == NULL)
-				continue;
-			err = snd_oxfw_stream_parse_format(format, &formation);
-			if (err < 0)
-				continue;
-
-			/* Add one MIDI port. */
-			if (formation.midi > 0)
-				oxfw->midi_input_ports = 1;
-		}
+			// The oPCR is not available for isoc communication.
+			err = 0;
+		} else {
+			for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+				format = oxfw->tx_stream_formats[i];
+				if (format == NULL)
+					continue;
+				err = snd_oxfw_stream_parse_format(format,
+								   &formation);
+				if (err < 0)
+					continue;
+
+				/* Add one MIDI port. */
+				if (formation.midi > 0)
+					oxfw->midi_input_ports = 1;
+			}
 
-		oxfw->has_output = true;
+			oxfw->has_output = true;
+		}
 	}
 
 	/* use iPCR[0] if exists */
 	if (plugs[0] > 0) {
 		err = fill_stream_formats(oxfw, AVC_GENERAL_PLUG_DIR_IN, 0);
-		if (err < 0)
-			goto end;
+		if (err < 0) {
+			if (err != -ENXIO)
+				return err;
+
+			// The iPCR is not available for isoc communication.
+			err = 0;
+		} else {
+			for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
+				format = oxfw->rx_stream_formats[i];
+				if (format == NULL)
+					continue;
+				err = snd_oxfw_stream_parse_format(format,
+								   &formation);
+				if (err < 0)
+					continue;
+
+				/* Add one MIDI port. */
+				if (formation.midi > 0)
+					oxfw->midi_output_ports = 1;
+			}
 
-		for (i = 0; i < SND_OXFW_STREAM_FORMAT_ENTRIES; i++) {
-			format = oxfw->rx_stream_formats[i];
-			if (format == NULL)
-				continue;
-			err = snd_oxfw_stream_parse_format(format, &formation);
-			if (err < 0)
-				continue;
-
-			/* Add one MIDI port. */
-			if (formation.midi > 0)
-				oxfw->midi_output_ports = 1;
+			oxfw->has_input = true;
 		}
 	}
 end:
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index fb6df3fc018e..1f1e3236efb8 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -118,7 +118,8 @@ static void oxfw_card_free(struct snd_card *card)
 {
 	struct snd_oxfw *oxfw = card->private_data;
 
-	snd_oxfw_stream_destroy_duplex(oxfw);
+	if (oxfw->has_output || oxfw->has_input)
+		snd_oxfw_stream_destroy_duplex(oxfw);
 }
 
 static int detect_quirks(struct snd_oxfw *oxfw)
@@ -206,23 +207,25 @@ static void do_registration(struct work_struct *work)
 	if (err < 0)
 		goto error;
 
-	err = snd_oxfw_stream_init_duplex(oxfw);
-	if (err < 0)
-		goto error;
+	if (oxfw->has_output || oxfw->has_input) {
+		err = snd_oxfw_stream_init_duplex(oxfw);
+		if (err < 0)
+			goto error;
 
-	err = snd_oxfw_create_pcm(oxfw);
-	if (err < 0)
-		goto error;
+		err = snd_oxfw_create_pcm(oxfw);
+		if (err < 0)
+			goto error;
 
-	snd_oxfw_proc_init(oxfw);
+		snd_oxfw_proc_init(oxfw);
 
-	err = snd_oxfw_create_midi(oxfw);
-	if (err < 0)
-		goto error;
+		err = snd_oxfw_create_midi(oxfw);
+		if (err < 0)
+			goto error;
 
-	err = snd_oxfw_create_hwdep(oxfw);
-	if (err < 0)
-		goto error;
+		err = snd_oxfw_create_hwdep(oxfw);
+		if (err < 0)
+			goto error;
+	}
 
 	err = snd_card_register(oxfw->card);
 	if (err < 0)
@@ -274,9 +277,11 @@ static void oxfw_bus_reset(struct fw_unit *unit)
 	fcp_bus_reset(oxfw->unit);
 
 	if (oxfw->registered) {
-		mutex_lock(&oxfw->mutex);
-		snd_oxfw_stream_update_duplex(oxfw);
-		mutex_unlock(&oxfw->mutex);
+		if (oxfw->has_output || oxfw->has_input) {
+			mutex_lock(&oxfw->mutex);
+			snd_oxfw_stream_update_duplex(oxfw);
+			mutex_unlock(&oxfw->mutex);
+		}
 
 		if (oxfw->entry->vendor_id == OUI_STANTON)
 			snd_oxfw_scs1x_update(oxfw);
diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h
index c30e537087b0..fa2d7f9e2dc3 100644
--- a/sound/firewire/oxfw/oxfw.h
+++ b/sound/firewire/oxfw/oxfw.h
@@ -45,6 +45,7 @@ struct snd_oxfw {
 
 	bool wrong_dbs;
 	bool has_output;
+	bool has_input;
 	u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
 	u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES];
 	bool assumed;