ASoC: qcom: audioreach: fix S24_LE playback and route mixer disconnects

Two AudioReach fixes found while bringing up 24-bit headphone playback on the Radxa Dragon Q6A (QCS6490, sc8280xp machine driver, WCD9385).

Patch 1: S24_LE was described to the DSP as packed 3-byte samples, so the ADSP audio_process crashed as soon as an S24_LE stream started. Using the 32-bit container size makes S24_LE work.

Patch 2: writing 0 to any "<BE> Audio Mixer <FE>" route control cleared the back end's link even if it belonged to another front end. As a result alsactl restore lost DisplayPort on every boot, and the board's UCM HDMI sequence disconnected the headphones (the PCM then stalls with hw_ptr stuck at 0).

Both were tested on 7.2.9 with the board's topology changed to allow S24_LE on MultiMedia1, and both apply to and build on asoc/for-next.

---
Marc Herrero (2):
ASoC: qcom: audioreach: use a 32-bit container for S24_LE PCM
ASoC: qcom: audioreach: only disconnect the route a mixer control owns

 sound/soc/qcom/qdsp6/audioreach.c | 4 ++--
 sound/soc/qcom/qdsp6/topology.c   | 2 +-
 2 files changed, 3 insertions(+), 3 deletions(-)

---
base-commit: cde5523c75f1320a55ac1d55a717ccbb566557f0
change-id: 20261111-q6apm-fixes-21db1bb2d6a1

Best regards,

Marc Herrero <[email protected]>

SNDRV_PCM_FORMAT_S24_LE carries 24 valid bits in a 32-bit container, but audioreach_shmem_set_media_format() and audioreach_pcm_set_media_format() set bits_per_sample = bit_width = 24 in PARAM_ID_MEDIA_FORMAT and PARAM_ID_PCM_OUTPUT_FORMAT_CFG, which tells the DSP that the samples are packed 3-byte words.

q6apm-dai advertises S24_LE, so with a topology that allows it on a playback front end the DSP walks the buffer with the wrong stride. On QCS6490 (Radxa Dragon Q6A) this crashes the ADSP audio_process:

PDM: service 'audio_process' crash: 'EX:audio_process:0x2:...' q6apm-lpass-dais ...: Failed to start APM port 113 remoteproc remoteproc0: crash detected in adsp: type fatal error

Use the container size (32) for 24-bit data and keep bit_width = 24, alignment = PCM_LSB_ALIGNED and q_factor = 23. With this change S24_LE playback works on the headphone path (codec DMA RX -> LPASS RX macro -> WCD9385), tested with the back end at S16_LE, S24_LE and S32_LE and at 48, 96 and 192 kHz.

The DSP then reads each 32-bit word as Q23, so S24_LE samples must be sign-extended into the top byte, as alsa-lib's conversions do.

---
 sound/soc/qcom/qdsp6/audioreach.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

index d2dc5d37bccf..937506edf34a 100644
@@ -1216,7 +1216,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph,
 	media_cfg->interleaved = module->interleave_type;
 	media_cfg->num_channels = mcfg->num_channels;
 	media_cfg->q_factor = mcfg->bit_width - 1;
-	media_cfg->bits_per_sample = mcfg->bit_width;
+	media_cfg->bits_per_sample = mcfg->bit_width == 24 ? 32 : mcfg->bit_width;
 	/* Convert the physical mapping to a logical mapping of the channels */
 	for (i = 0, j = 0; i < AR_PCM_MAX_NUM_CHANNEL && j < mcfg->num_channels; i++) {
 		if (!mcfg->channel_map[i])

@@ -1308,7 +1308,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
 		cfg->sample_rate = mcfg->sample_rate;
 		cfg->bit_width = mcfg->bit_width;
 		cfg->alignment = PCM_LSB_ALIGNED;
-		cfg->bits_per_sample = mcfg->bit_width;
+		cfg->bits_per_sample = mcfg->bit_width == 24 ? 32 : mcfg->bit_width;
 		cfg->q_factor = mcfg->bit_width - 1;
 		cfg->endianness = PCM_LITTLE_ENDIAN;
 		cfg->num_channels = mcfg->num_channels;

2.53.0

Each AudioReach virtual mixer route control ("<BE> Audio Mixer <FE>") connects a front-end stream graph to a back-end device graph by storing the source module in the device graph's audioreach_graph_info. Writing 0 to any of these controls clears that link unconditionally, even when the device graph is connected to a different front end.

When two front ends can feed the same back end the result therefore depends on write order. On QCS6490 (Radxa Dragon Q6A):

- alsactl restore writes 'DISPLAY_PORT_RX_0 Audio Mixer MultiMedia2' = 1 and then 'DISPLAY_PORT_RX_0 Audio Mixer MultiMedia1' = 0, which leaves DisplayPort disconnected although the saved state has it connected;
- the UCM HDMI enable sequence writes
'RX_CODEC_DMA_RX_0 Audio Mixer MultiMedia2' = 0, which disconnects MultiMedia1 from the headphones. A PCM started on MultiMedia1 after that never advances (hw_ptr stays at 0).

Only drop the link when it belongs to the control being cleared, that is when audioreach_is_vmixer_connected() reports it as connected. Writing 0 to a control that is not the active route becomes a no-op, which matches what its get() callback already returns.

---
 sound/soc/qcom/qdsp6/topology.c | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

index 612d8f03e6e9..e92408e09bac 100644
@@ -1300,7 +1300,7 @@ static int audioreach_put_audio_mixer(struct snd_kcontrol *kcontrol,
 	if (ucontrol->value.integer.value[0]) {
 		audioreach_connect_sub_graphs(data, scontrol, dapm_scontrol, true);
 		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 1, NULL);
-	} else {
+	} else if (audioreach_is_vmixer_connected(data, scontrol, dapm_scontrol)) {
 		audioreach_connect_sub_graphs(data, scontrol, dapm_scontrol, false);
 		snd_soc_dapm_mixer_update_power(dapm, kcontrol, 0, NULL);
 	}

2.53.0