Merge tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This looks like a relatively calm development cycle; there have been
  only few changes in ALSA and ASoC core sides while we get lots of
  device-specific fixes and updates as usual. Most of commits are about
  ASoC, including Intel SOF/AVS and many device tree updates.

  Below are some highlights:

  Core:
   - Improvement in memalloc helper for fallback allocations
   - More cleanups of ASoC DAPM code

  ASoC:
   - Factoring out of mapping hw_params onto SoundWire configuration
   - The ever ongoing overhauls of the Intel DSP code continue,
     including support for loading libraries and probes with IPC4 on
     SOF.
   - Support for more sample formats on JZ4740
   - Lots of device tree conversions and fixups
   - Support for Allwinner D1, a range of AMD and Intel systems,
     Mediatek systems with multiple DMICs, Nuvoton NAU8318, NXP
     fsl_rpmsg and i.MX93, Qualcomm AudioReach Enable, MFC and SAL,
     RealTek RT1318 and Rockchip RK3588

  ALSA:
   - Addition of PCM kselftest; still minimalistic but can be extended
     in future
   - Fixes for corner-case XRUNs with USB-audio implicit feedback mode
   - Usual device-specific quirk updates for USB- and HD-audio
   - FireWire DICE updates

  This also contains a few cross-tree updates:
   - Some OMAP board file updates for removal of relevant OMAP platforms
   - A new I2C API update for I2C probe API adaption
   - A DRM update for the further hdmi-codec updates"

* tag 'sound-6.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (417 commits)
  ALSA: mts64: fix possible null-ptr-defer in snd_mts64_interrupt
  ALSA: patch_realtek: Fix Dell Inspiron Plus 16
  ALSA: hda/cirrus: Add extra 10 ms delay to allow PLL settle and lock.
  ASoC: dt-bindings: Correct Alexandre Belloni email
  ASoC: dt-bindings: maxim,max98504: Convert to DT schema
  ASoC: dt-bindings: maxim,max98357a: Convert to DT schema
  ASoC: dt-bindings: Reference common DAI properties
  ASoC: dt-bindings: Extend name-prefix.yaml into common DAI properties
  ASoC: rt715: Make read-only arrays capture_reg_H and capture_reg_L static const
  ASoC: uniphier: aio-core: Make some read-only arrays static const
  ASoC: wcd938x: Make read-only array minCode_param static const
  ASoC: qcom: lpass-sc7280: Add maybe_unused tag for system PM ops
  ASoC : SOF: amd: Add support for IPC and DSP dumps
  ASoC: SOF: amd: Use poll function instead to read ACP_SHA_DSP_FW_QUALIFIER
  ALSA: usb-audio: Workaround for XRUN at prepare
  ALSA: pcm: Handle XRUN at trigger START
  ALSA: pcm: Set missing stop_operating flag at undoing trigger start
  drm: tda99x: Don't advertise non-existent capture support
  ASoC: hdmi-codec: Allow playback and capture to be disabled
  kselftest/alsa: Add more coverage of sample rates and channel counts
  ...

4 files changed, 46 insertions, 30 deletions

diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c
index 34250e6022ff..3cf5a87d69ea 100644
--- a/sound/core/memalloc.c
+++ b/sound/core/memalloc.c
@@ -719,7 +719,6 @@ static const struct snd_malloc_ops snd_dma_sg_wc_ops = {
 struct snd_dma_sg_fallback {
 	size_t count;
 	struct page **pages;
-	dma_addr_t *addrs;
 };
 
 static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab,
@@ -731,38 +730,49 @@ static void __snd_dma_sg_fallback_free(struct snd_dma_buffer *dmab,
 	for (i = 0; i < sgbuf->count && sgbuf->pages[i]; i++)
 		do_free_pages(page_address(sgbuf->pages[i]), PAGE_SIZE, wc);
 	kvfree(sgbuf->pages);
-	kvfree(sgbuf->addrs);
 	kfree(sgbuf);
 }
 
 static void *snd_dma_sg_fallback_alloc(struct snd_dma_buffer *dmab, size_t size)
 {
 	struct snd_dma_sg_fallback *sgbuf;
-	struct page **pages;
-	size_t i, count;
+	struct page **pagep, *curp;
+	size_t chunk, npages;
+	dma_addr_t addr;
 	void *p;
 	bool wc = dmab->dev.type == SNDRV_DMA_TYPE_DEV_WC_SG_FALLBACK;
 
 	sgbuf = kzalloc(sizeof(*sgbuf), GFP_KERNEL);
 	if (!sgbuf)
 		return NULL;
-	count = PAGE_ALIGN(size) >> PAGE_SHIFT;
-	pages = kvcalloc(count, sizeof(*pages), GFP_KERNEL);
-	if (!pages)
-		goto error;
-	sgbuf->pages = pages;
-	sgbuf->addrs = kvcalloc(count, sizeof(*sgbuf->addrs), GFP_KERNEL);
-	if (!sgbuf->addrs)
+	size = PAGE_ALIGN(size);
+	sgbuf->count = size >> PAGE_SHIFT;
+	sgbuf->pages = kvcalloc(sgbuf->count, sizeof(*sgbuf->pages), GFP_KERNEL);
+	if (!sgbuf->pages)
 		goto error;
 
-	for (i = 0; i < count; sgbuf->count++, i++) {
-		p = do_alloc_pages(dmab->dev.dev, PAGE_SIZE, &sgbuf->addrs[i], wc);
-		if (!p)
-			goto error;
-		sgbuf->pages[i] = virt_to_page(p);
+	pagep = sgbuf->pages;
+	chunk = size;
+	while (size > 0) {
+		chunk = min(size, chunk);
+		p = do_alloc_pages(dmab->dev.dev, chunk, &addr, wc);
+		if (!p) {
+			if (chunk <= PAGE_SIZE)
+				goto error;
+			chunk >>= 1;
+			chunk = PAGE_SIZE << get_order(chunk);
+			continue;
+		}
+
+		size -= chunk;
+		/* fill pages */
+		npages = chunk >> PAGE_SHIFT;
+		curp = virt_to_page(p);
+		while (npages--)
+			*pagep++ = curp++;
 	}
 
-	p = vmap(pages, count, VM_MAP, PAGE_KERNEL);
+	p = vmap(sgbuf->pages, sgbuf->count, VM_MAP, PAGE_KERNEL);
 	if (!p)
 		goto error;
 	dmab->private_data = sgbuf;
diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c
index 33769ca78cc8..9c122e757efe 100644
--- a/sound/core/pcm_native.c
+++ b/sound/core/pcm_native.c
@@ -288,7 +288,7 @@ static int constrain_mask_params(struct snd_pcm_substream *substream,
 					&substream->runtime->hw_constraints;
 	struct snd_mask *m;
 	unsigned int k;
-	struct snd_mask old_mask;
+	struct snd_mask old_mask __maybe_unused;
 	int changed;
 
 	for (k = SNDRV_PCM_HW_PARAM_FIRST_MASK; k <= SNDRV_PCM_HW_PARAM_LAST_MASK; k++) {
@@ -324,7 +324,7 @@ static int constrain_interval_params(struct snd_pcm_substream *substream,
 					&substream->runtime->hw_constraints;
 	struct snd_interval *i;
 	unsigned int k;
-	struct snd_interval old_interval;
+	struct snd_interval old_interval __maybe_unused;
 	int changed;
 
 	for (k = SNDRV_PCM_HW_PARAM_FIRST_INTERVAL; k <= SNDRV_PCM_HW_PARAM_LAST_INTERVAL; k++) {
@@ -364,8 +364,8 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream,
 	unsigned int stamp;
 	struct snd_pcm_hw_rule *r;
 	unsigned int d;
-	struct snd_mask old_mask;
-	struct snd_interval old_interval;
+	struct snd_mask old_mask __maybe_unused;
+	struct snd_interval old_interval __maybe_unused;
 	bool again;
 	int changed, err = 0;
 
@@ -648,8 +648,8 @@ static int snd_pcm_hw_params_choose(struct snd_pcm_substream *pcm,
 		-1
 	};
 	const int *v;
-	struct snd_mask old_mask;
-	struct snd_interval old_interval;
+	struct snd_mask old_mask __maybe_unused;
+	struct snd_interval old_interval __maybe_unused;
 	int changed;
 
 	for (v = vars; *v != -1; v++) {
@@ -1424,16 +1424,24 @@ static int snd_pcm_pre_start(struct snd_pcm_substream *substream,
 static int snd_pcm_do_start(struct snd_pcm_substream *substream,
 			    snd_pcm_state_t state)
 {
+	int err;
+
 	if (substream->runtime->trigger_master != substream)
 		return 0;
-	return substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_START);
+	err = substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_START);
+	/* XRUN happened during the start */
+	if (err == -EPIPE)
+		__snd_pcm_set_state(substream->runtime, SNDRV_PCM_STATE_XRUN);
+	return err;
 }
 
 static void snd_pcm_undo_start(struct snd_pcm_substream *substream,
 			       snd_pcm_state_t state)
 {
-	if (substream->runtime->trigger_master == substream)
+	if (substream->runtime->trigger_master == substream) {
 		substream->ops->trigger(substream, SNDRV_PCM_TRIGGER_STOP);
+		substream->runtime->stop_operating = true;
+	}
 }
 
 static void snd_pcm_post_start(struct snd_pcm_substream *substream,
diff --git a/sound/core/pcm_trace.h b/sound/core/pcm_trace.h
index f18da2050772..350b40b906ca 100644
--- a/sound/core/pcm_trace.h
+++ b/sound/core/pcm_trace.h
@@ -88,19 +88,19 @@ TRACE_EVENT(hw_ptr_error,
 		__field( unsigned int, device )
 		__field( unsigned int, number )
 		__field( unsigned int, stream )
-		__field( const char *, reason )
+		__string( reason, why )
 	),
 	TP_fast_assign(
 		__entry->card = (substream)->pcm->card->number;
 		__entry->device = (substream)->pcm->device;
 		__entry->number = (substream)->number;
 		__entry->stream = (substream)->stream;
-		__entry->reason = (why);
+		__assign_str(reason, why);
 	),
 	TP_printk("pcmC%dD%d%s/sub%d: ERROR: %s",
 		  __entry->card, __entry->device,
 		  __entry->stream == SNDRV_PCM_STREAM_PLAYBACK ? "p" : "c",
-		  __entry->number, __entry->reason)
+		  __entry->number, __get_str(reason))
 );
 
 TRACE_EVENT(applptr,
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index d8edb6055072..7147fda66d93 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -1050,7 +1050,6 @@ static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream,
 	struct snd_rawmidi_runtime *runtime = substream->runtime;
 	struct snd_rawmidi_framing_tstamp *dest_ptr;
 	struct snd_rawmidi_framing_tstamp frame = { .tv_sec = tstamp->tv_sec, .tv_nsec = tstamp->tv_nsec };
-	int dest_frames = 0;
 	int orig_count = src_count;
 	int frame_size = sizeof(struct snd_rawmidi_framing_tstamp);
 
@@ -1077,7 +1076,6 @@ static int receive_with_tstamp_framing(struct snd_rawmidi_substream *substream,
 		runtime->avail += frame_size;
 		runtime->hw_ptr += frame_size;
 		runtime->hw_ptr %= runtime->buffer_size;
-		dest_frames++;
 	}
 	return orig_count - src_count;
 }