ASoC: Intel: avs: Set of streaming fixes

Merge series from Cezary Rojewski <cezary.rojewski@intel.com>:

Small set of patches two of which fix problems observed during shutdown
and XRUN scenarios for PCM streaming. These ensure HDAudio HOST stream
is reset and re-setup during XRUNs and synchronize avs_dai_fe_shutdown()
and period-elapsed work (thread) so that no slab-use-after-free panics
occur.

The last change makes the 'format' parameter provided by a
compress-application taken into account - currently its ignored. Such
approach helps us increase the coverage of data probing (debug)
functionality.

Author: broonie

sound/soc/intel/avs/pcm.c

@@ -651,6 +651,7 @@ static void avs_dai_fe_shutdown(struct snd_pcm_substream *substream, struct snd_
 
 	data = snd_soc_dai_get_dma_data(dai, substream);
 
+	disable_work_sync(&data->period_elapsed_work);
 	snd_hdac_ext_stream_release(data->host_stream, HDAC_EXT_STREAM_TYPE_HOST);
 	avs_dai_shutdown(substream, dai);
 }
@@ -754,6 +755,8 @@ static int avs_dai_fe_prepare(struct snd_pcm_substream *substream, struct snd_so
 	data = snd_soc_dai_get_dma_data(dai, substream);
 	host_stream = data->host_stream;
 
+	if (runtime->state == SNDRV_PCM_STATE_XRUN)
+		hdac_stream(host_stream)->prepared = false;
 	if (hdac_stream(host_stream)->prepared)
 		return 0;
 

sound/soc/intel/avs/probes.c

@@ -14,8 +14,8 @@
 #include "debug.h"
 #include "messages.h"
 
-static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id node_id,
-			      size_t buffer_size)
+static int avs_dsp_init_probe(struct avs_dev *adev, struct snd_compr_params *params, int bps,
+			      union avs_connector_node_id node_id, size_t buffer_size)
 {
 	struct avs_probe_cfg cfg = {{0}};
 	struct avs_module_entry mentry;
@@ -27,12 +27,16 @@ static int avs_dsp_init_probe(struct avs_dev *adev, union avs_connector_node_id
 		return ret;
 
 	/*
-	 * Probe module uses no cycles, audio data format and input and output
-	 * frame sizes are unused. It is also not owned by any pipeline.
+	 * Probe module uses no cycles, input and output frame sizes are unused.
+	 * It is also not owned by any pipeline.
 	 */
 	cfg.base.ibs = 1;
 	/* BSS module descriptor is always segment of index=2. */
 	cfg.base.is_pages = mentry.segments[2].flags.length;
+	cfg.base.audio_fmt.sampling_freq = params->codec.sample_rate;
+	cfg.base.audio_fmt.bit_depth = bps;
+	cfg.base.audio_fmt.num_channels = params->codec.ch_out;
+	cfg.base.audio_fmt.valid_bit_depth = bps;
 	cfg.gtw_cfg.node_id = node_id;
 	cfg.gtw_cfg.dma_buffer_size = buffer_size;
 
@@ -128,8 +132,6 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
 	struct hdac_ext_stream *host_stream = avs_compr_get_host_stream(cstream);
 	struct snd_compr_runtime *rtd = cstream->runtime;
 	struct avs_dev *adev = to_avs_dev(dai->dev);
-	/* compr params do not store bit depth, default to S32_LE. */
-	snd_pcm_format_t format = SNDRV_PCM_FORMAT_S32_LE;
 	unsigned int format_val;
 	int bps, ret;
 
@@ -142,7 +144,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
 	ret = snd_compr_malloc_pages(cstream, rtd->buffer_size);
 	if (ret < 0)
 		return ret;
-	bps = snd_pcm_format_physical_width(format);
+	bps = snd_pcm_format_physical_width(params->codec.format);
 	if (bps < 0)
 		return bps;
 	format_val = snd_hdac_stream_format(params->codec.ch_out, bps, params->codec.sample_rate);
@@ -166,7 +168,7 @@ static int avs_probe_compr_set_params(struct snd_compr_stream *cstream,
 		node_id.vindex = hdac_stream(host_stream)->stream_tag - 1;
 		node_id.dma_type = AVS_DMA_HDA_HOST_INPUT;
 
-		ret = avs_dsp_init_probe(adev, node_id, rtd->dma_bytes);
+		ret = avs_dsp_init_probe(adev, params, bps, node_id, rtd->dma_bytes);
 		if (ret < 0) {
 			dev_err(dai->dev, "probe init failed: %d\n", ret);
 			avs_dsp_enable_d0ix(adev);

sound/soc/sdw_utils/soc_sdw_utils.c

@@ -313,6 +313,7 @@ struct asoc_sdw_codec_info codec_info_list[] = {
 	},
 	{
 		.part_id = 0x1321,
+		.name_prefix = "rt1320",
 		.dais = {
 			{
 				.direction = {true, false},