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-rw-r--r--sound/soc/qcom/lpass-cdc-dma.c3
-rw-r--r--sound/soc/qcom/lpass-hdmi.c2
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.c55
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.h35
-rw-r--r--sound/soc/qcom/qdsp6/q6apm-dai.c29
-rw-r--r--sound/soc/qcom/qdsp6/q6apm-lpass-dais.c7
-rw-r--r--sound/soc/qcom/qdsp6/q6apm.c3
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c26
-rw-r--r--sound/soc/qcom/qdsp6/topology.c45
-rw-r--r--sound/soc/qcom/sc8280xp.c13
-rw-r--r--sound/soc/qcom/x1e80100.c5
11 files changed, 173 insertions, 50 deletions
diff --git a/sound/soc/qcom/lpass-cdc-dma.c b/sound/soc/qcom/lpass-cdc-dma.c
index 8106c586f68a..2dc8c75c4bf0 100644
--- a/sound/soc/qcom/lpass-cdc-dma.c
+++ b/sound/soc/qcom/lpass-cdc-dma.c
@@ -217,8 +217,9 @@ static int lpass_cdc_dma_daiops_hw_params(struct snd_pcm_substream *substream,
{
struct snd_soc_pcm_runtime *soc_runtime = snd_soc_substream_to_rtd(substream);
struct lpaif_dmactl *dmactl = NULL;
- unsigned int ret, regval;
+ unsigned int regval;
unsigned int channels = params_channels(params);
+ int ret;
int id;
switch (channels) {
diff --git a/sound/soc/qcom/lpass-hdmi.c b/sound/soc/qcom/lpass-hdmi.c
index ce753ebc0894..6d9795306cfa 100644
--- a/sound/soc/qcom/lpass-hdmi.c
+++ b/sound/soc/qcom/lpass-hdmi.c
@@ -23,7 +23,6 @@ static int lpass_hdmi_daiops_hw_params(struct snd_pcm_substream *substream,
snd_pcm_format_t format = params_format(params);
unsigned int rate = params_rate(params);
unsigned int channels = params_channels(params);
- unsigned int ret;
int bitwidth;
unsigned int word_length;
unsigned int ch_sts_buf0;
@@ -33,6 +32,7 @@ static int lpass_hdmi_daiops_hw_params(struct snd_pcm_substream *substream,
unsigned int ch = 0;
struct lpass_dp_metadata_ctl *meta_ctl = drvdata->meta_ctl;
struct lpass_sstream_ctl *sstream_ctl = drvdata->sstream_ctl;
+ int ret;
bitwidth = snd_pcm_format_width(format);
if (bitwidth < 0) {
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 4ebaaf736fb9..2365424a9b42 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -811,6 +811,30 @@ static int audioreach_gapless_set_media_format(struct q6apm_graph *graph,
EARLY_EOS_DELAY_MS);
}
+static int audioreach_set_module_config(struct q6apm_graph *graph,
+ struct audioreach_module *module,
+ struct audioreach_module_config *cfg)
+{
+ int payload_size = le32_to_cpu(module->data->size);
+ struct gpr_pkt *pkt;
+ int rc;
+ void *p;
+
+ pkt = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
+ if (IS_ERR(pkt))
+ return PTR_ERR(pkt);
+
+ p = (void *)pkt + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
+
+ memcpy(p, module->data->data, payload_size);
+
+ rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
+
+ kfree(pkt);
+
+ return rc;
+}
+
static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
@@ -859,6 +883,7 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
struct payload_media_fmt_aac_t *aac_cfg;
struct payload_media_fmt_pcm *mp3_cfg;
struct payload_media_fmt_flac_t *flac_cfg;
+ struct payload_media_fmt_opus_t *opus_cfg;
switch (mcfg->fmt) {
case SND_AUDIOCODEC_MP3:
@@ -901,6 +926,32 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
flac_cfg->min_frame_size = mcfg->codec.options.flac_d.min_frame_size;
flac_cfg->max_frame_size = mcfg->codec.options.flac_d.max_frame_size;
break;
+ case SND_AUDIOCODEC_OPUS_RAW:
+ media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
+ media_fmt_hdr->fmt_id = MEDIA_FMT_ID_OPUS;
+ media_fmt_hdr->payload_size = sizeof(*opus_cfg);
+ p = p + sizeof(*media_fmt_hdr);
+ opus_cfg = p;
+ /* raw opus packets prepended with 4 bytes of length */
+ opus_cfg->bitstream_format = 1;
+ /*
+ * payload_type:
+ * 0 -- read metadata from opus stream;
+ * 1 -- metadata is provided by filling in the struct here.
+ */
+ opus_cfg->payload_type = 1;
+ opus_cfg->version = mcfg->codec.options.opus_d.version;
+ opus_cfg->num_channels = mcfg->codec.options.opus_d.num_channels;
+ opus_cfg->pre_skip = mcfg->codec.options.opus_d.pre_skip;
+ opus_cfg->sample_rate = mcfg->codec.options.opus_d.sample_rate;
+ opus_cfg->output_gain = mcfg->codec.options.opus_d.output_gain;
+ opus_cfg->mapping_family = mcfg->codec.options.opus_d.mapping_family;
+ opus_cfg->stream_count = mcfg->codec.options.opus_d.chan_map.stream_count;
+ opus_cfg->coupled_count = mcfg->codec.options.opus_d.chan_map.coupled_count;
+ memcpy(opus_cfg->channel_mapping, mcfg->codec.options.opus_d.chan_map.channel_map,
+ sizeof(opus_cfg->channel_mapping));
+ opus_cfg->reserved[0] = opus_cfg->reserved[1] = opus_cfg->reserved[2] = 0;
+ break;
default:
return -EINVAL;
}
@@ -971,6 +1022,7 @@ static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
param_data->param_id = PARAM_ID_I2S_INTF_CFG;
param_data->param_size = ic_sz - APM_MODULE_PARAM_DATA_SIZE;
+ intf_cfg->cfg.lpaif_type = module->hw_interface_type;
intf_cfg->cfg.intf_idx = module->hw_interface_idx;
intf_cfg->cfg.sd_line_idx = module->sd_line_idx;
@@ -1247,6 +1299,9 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
case MODULE_ID_DISPLAY_PORT_SINK:
rc = audioreach_display_port_set_media_format(graph, module, cfg);
break;
+ case MODULE_ID_SMECNS_V2:
+ rc = audioreach_set_module_config(graph, module, cfg);
+ break;
case MODULE_ID_I2S_SOURCE:
case MODULE_ID_I2S_SINK:
rc = audioreach_i2s_set_media_format(graph, module, cfg);
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 61a69df4f50f..d1b60b36468a 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -4,6 +4,7 @@
#define __AUDIOREACH_H__
#include <linux/types.h>
#include <linux/soc/qcom/apr.h>
+#include <uapi/sound/snd_ar_tokens.h>
#include <sound/soc.h>
struct q6apm;
struct q6apm_graph;
@@ -17,18 +18,20 @@ struct q6apm_graph;
#define MODULE_ID_PCM_DEC 0x07001005
#define MODULE_ID_PLACEHOLDER_ENCODER 0x07001008
#define MODULE_ID_PLACEHOLDER_DECODER 0x07001009
-#define MODULE_ID_SAL 0x07001010
-#define MODULE_ID_MFC 0x07001015
-#define MODULE_ID_CODEC_DMA_SINK 0x07001023
-#define MODULE_ID_CODEC_DMA_SOURCE 0x07001024
#define MODULE_ID_I2S_SINK 0x0700100A
#define MODULE_ID_I2S_SOURCE 0x0700100B
+#define MODULE_ID_SAL 0x07001010
+#define MODULE_ID_MFC 0x07001015
#define MODULE_ID_DATA_LOGGING 0x0700101A
#define MODULE_ID_AAC_DEC 0x0700101F
+#define MODULE_ID_CODEC_DMA_SINK 0x07001023
+#define MODULE_ID_CODEC_DMA_SOURCE 0x07001024
#define MODULE_ID_FLAC_DEC 0x0700102F
+#define MODULE_ID_SMECNS_V2 0x07001031
#define MODULE_ID_MP3_DECODE 0x0700103B
#define MODULE_ID_GAPLESS 0x0700104D
#define MODULE_ID_DISPLAY_PORT_SINK 0x07001069
+#define MODULE_ID_OPUS_DEC 0x07001174
#define APM_CMD_GET_SPF_STATE 0x01001021
#define APM_CMD_RSP_GET_SPF_STATE 0x02001007
@@ -255,6 +258,22 @@ struct payload_media_fmt_aac_t {
uint32_t sample_rate;
} __packed;
+#define MEDIA_FMT_ID_OPUS 0x09001039
+struct payload_media_fmt_opus_t {
+ uint16_t bitstream_format;
+ uint16_t payload_type;
+ uint8_t version;
+ uint8_t num_channels;
+ uint16_t pre_skip;
+ uint32_t sample_rate;
+ uint16_t output_gain;
+ uint8_t mapping_family;
+ uint8_t stream_count;
+ uint8_t coupled_count;
+ uint8_t channel_mapping[8];
+ uint8_t reserved[3];
+} __packed;
+
#define DATA_CMD_WR_SH_MEM_EP_EOS 0x04001002
#define WR_SH_MEM_EP_EOS_POLICY_LAST 1
#define WR_SH_MEM_EP_EOS_POLICY_EACH 2
@@ -461,8 +480,8 @@ struct param_id_i2s_intf_cfg {
} __packed;
#define I2S_INTF_TYPE_PRIMARY 0
-#define I2S_INTF_TYPE_SECOINDARY 1
-#define I2S_INTF_TYPE_TERTINARY 2
+#define I2S_INTF_TYPE_SECONDARY 1
+#define I2S_INTF_TYPE_TERTIARY 2
#define I2S_INTF_TYPE_QUATERNARY 3
#define I2S_INTF_TYPE_QUINARY 4
#define I2S_SD0 1
@@ -707,9 +726,6 @@ struct audioreach_module {
uint32_t max_ip_port;
uint32_t max_op_port;
- uint32_t in_port;
- uint32_t out_port;
-
uint32_t num_connections;
/* Connections */
uint32_t src_mod_inst_id;
@@ -745,6 +761,7 @@ struct audioreach_module {
struct list_head node;
struct audioreach_container *container;
struct snd_soc_dapm_widget *widget;
+ struct audioreach_module_priv_data *data;
};
struct audioreach_module_config {
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 2cd522108221..4ecaff45c518 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -11,6 +11,7 @@
#include <sound/soc-dapm.h>
#include <linux/spinlock.h>
#include <sound/pcm.h>
+#include <asm/div64.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <sound/pcm_params.h>
@@ -65,9 +66,9 @@ struct q6apm_dai_rtd {
unsigned int pcm_size;
unsigned int pcm_count;
unsigned int periods;
- unsigned int bytes_sent;
- unsigned int bytes_received;
- unsigned int copied_total;
+ uint64_t bytes_sent;
+ uint64_t bytes_received;
+ uint64_t copied_total;
uint16_t bits_per_sample;
snd_pcm_uframes_t queue_ptr;
bool next_track;
@@ -550,10 +551,11 @@ static int q6apm_dai_compr_get_caps(struct snd_soc_component *component,
caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
- caps->num_codecs = 3;
+ caps->num_codecs = 4;
caps->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_AAC;
caps->codecs[2] = SND_AUDIOCODEC_FLAC;
+ caps->codecs[3] = SND_AUDIOCODEC_OPUS_RAW;
return 0;
}
@@ -575,15 +577,17 @@ static int q6apm_dai_compr_get_codec_caps(struct snd_soc_component *component,
static int q6apm_dai_compr_pointer(struct snd_soc_component *component,
struct snd_compr_stream *stream,
- struct snd_compr_tstamp *tstamp)
+ struct snd_compr_tstamp64 *tstamp)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
+ uint64_t temp_copied_total;
spin_lock_irqsave(&prtd->lock, flags);
tstamp->copied_total = prtd->copied_total;
- tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+ temp_copied_total = tstamp->copied_total;
+ tstamp->byte_offset = do_div(temp_copied_total, prtd->pcm_size);
spin_unlock_irqrestore(&prtd->lock, flags);
return 0;
@@ -760,21 +764,24 @@ static int q6apm_compr_copy(struct snd_soc_component *component,
size_t copy;
u32 wflags = 0;
u32 app_pointer;
- u32 bytes_received;
+ uint64_t bytes_received;
+ uint64_t temp_bytes_received;
uint32_t bytes_to_write;
- int avail, bytes_in_flight = 0;
+ uint64_t avail, bytes_in_flight = 0;
bytes_received = prtd->bytes_received;
+ temp_bytes_received = bytes_received;
/**
* Make sure that next track data pointer is aligned at 32 bit boundary
* This is a Mandatory requirement from DSP data buffers alignment
*/
- if (prtd->next_track)
+ if (prtd->next_track) {
bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+ temp_bytes_received = bytes_received;
+ }
- app_pointer = bytes_received/prtd->pcm_size;
- app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
+ app_pointer = do_div(temp_bytes_received, prtd->pcm_size);
dstn = prtd->dma_buffer.area + app_pointer;
if (count < prtd->pcm_size - app_pointer) {
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index a0d90462fd6a..528756f1332b 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -213,8 +213,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
return 0;
err:
- q6apm_graph_close(dai_data->graph[dai->id]);
- dai_data->graph[dai->id] = NULL;
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
return rc;
}
@@ -260,6 +262,7 @@ static const struct snd_soc_dai_ops q6i2s_ops = {
.shutdown = q6apm_lpass_dai_shutdown,
.set_channel_map = q6dma_set_channel_map,
.hw_params = q6dma_hw_params,
+ .set_fmt = q6i2s_set_fmt,
};
static const struct snd_soc_dai_ops q6hdmi_ops = {
diff --git a/sound/soc/qcom/qdsp6/q6apm.c b/sound/soc/qcom/qdsp6/q6apm.c
index b4ffa0f0b188..0e667a7eb546 100644
--- a/sound/soc/qcom/qdsp6/q6apm.c
+++ b/sound/soc/qcom/qdsp6/q6apm.c
@@ -354,6 +354,9 @@ int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph,
case SND_AUDIOCODEC_FLAC:
module_id = MODULE_ID_FLAC_DEC;
break;
+ case SND_AUDIOCODEC_OPUS_RAW:
+ module_id = MODULE_ID_OPUS_DEC;
+ break;
default:
return -EINVAL;
}
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index a400c9a31fea..b616ce316d2f 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -14,6 +14,7 @@
#include <sound/pcm.h>
#include <linux/spinlock.h>
#include <sound/compress_driver.h>
+#include <asm/div64.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <sound/pcm_params.h>
@@ -59,9 +60,9 @@ struct q6asm_dai_rtd {
unsigned int pcm_count;
unsigned int pcm_irq_pos; /* IRQ position */
unsigned int periods;
- unsigned int bytes_sent;
- unsigned int bytes_received;
- unsigned int copied_total;
+ uint64_t bytes_sent;
+ uint64_t bytes_received;
+ uint64_t copied_total;
uint16_t bits_per_sample;
uint16_t source; /* Encoding source bit mask */
struct audio_client *audio_client;
@@ -1026,16 +1027,18 @@ static int q6asm_dai_compr_trigger(struct snd_soc_component *component,
static int q6asm_dai_compr_pointer(struct snd_soc_component *component,
struct snd_compr_stream *stream,
- struct snd_compr_tstamp *tstamp)
+ struct snd_compr_tstamp64 *tstamp)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6asm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
+ uint64_t temp_copied_total;
spin_lock_irqsave(&prtd->lock, flags);
tstamp->copied_total = prtd->copied_total;
- tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
+ temp_copied_total = tstamp->copied_total;
+ tstamp->byte_offset = do_div(temp_copied_total, prtd->pcm_size);
spin_unlock_irqrestore(&prtd->lock, flags);
@@ -1050,23 +1053,26 @@ static int q6asm_compr_copy(struct snd_soc_component *component,
struct q6asm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
u32 wflags = 0;
- int avail, bytes_in_flight = 0;
+ uint64_t avail, bytes_in_flight = 0;
void *dstn;
size_t copy;
u32 app_pointer;
- u32 bytes_received;
+ uint64_t bytes_received;
+ uint64_t temp_bytes_received;
bytes_received = prtd->bytes_received;
+ temp_bytes_received = bytes_received;
/**
* Make sure that next track data pointer is aligned at 32 bit boundary
* This is a Mandatory requirement from DSP data buffers alignment
*/
- if (prtd->next_track)
+ if (prtd->next_track) {
bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
+ temp_bytes_received = bytes_received;
+ }
- app_pointer = bytes_received/prtd->pcm_size;
- app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
+ app_pointer = do_div(temp_bytes_received, prtd->pcm_size);
dstn = prtd->dma_buffer.area + app_pointer;
if (count < prtd->pcm_size - app_pointer) {
diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c
index 83319a928f29..f61285e7dcf2 100644
--- a/sound/soc/qcom/qdsp6/topology.c
+++ b/sound/soc/qcom/qdsp6/topology.c
@@ -305,6 +305,34 @@ static struct snd_soc_tplg_vendor_array *audioreach_get_module_array(
return NULL;
}
+static struct audioreach_module_priv_data *audioreach_get_module_priv_data(
+ struct snd_soc_tplg_private *private)
+{
+ int sz;
+
+ for (sz = 0; sz < le32_to_cpu(private->size); ) {
+ struct snd_soc_tplg_vendor_array *mod_array;
+
+ mod_array = (struct snd_soc_tplg_vendor_array *)((u8 *)private->array + sz);
+ if (le32_to_cpu(mod_array->type) == SND_SOC_AR_TPLG_MODULE_CFG_TYPE) {
+ struct audioreach_module_priv_data *pdata;
+
+ pdata = kzalloc(struct_size(pdata, data, le32_to_cpu(mod_array->size)),
+ GFP_KERNEL);
+ if (!pdata)
+ return ERR_PTR(-ENOMEM);
+
+ memcpy(pdata, ((u8 *)private->data + sz), struct_size(pdata, data,
+ le32_to_cpu(mod_array->size)));
+ return pdata;
+ }
+
+ sz = sz + le32_to_cpu(mod_array->size);
+ }
+
+ return NULL;
+}
+
static struct audioreach_sub_graph *audioreach_parse_sg_tokens(struct q6apm *apm,
struct snd_soc_tplg_private *private)
{
@@ -412,7 +440,7 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap
struct snd_soc_tplg_private *private,
struct snd_soc_dapm_widget *w)
{
- uint32_t max_ip_port = 0, max_op_port = 0, in_port = 0, out_port = 0;
+ uint32_t max_ip_port = 0, max_op_port = 0;
uint32_t src_mod_op_port_id[AR_MAX_MOD_LINKS] = { 0, };
uint32_t dst_mod_inst_id[AR_MAX_MOD_LINKS] = { 0, };
uint32_t dst_mod_ip_port_id[AR_MAX_MOD_LINKS] = { 0, };
@@ -455,12 +483,6 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap
case AR_TKN_U32_MODULE_MAX_OP_PORTS:
max_op_port = le32_to_cpu(mod_elem->value);
break;
- case AR_TKN_U32_MODULE_IN_PORTS:
- in_port = le32_to_cpu(mod_elem->value);
- break;
- case AR_TKN_U32_MODULE_OUT_PORTS:
- out_port = le32_to_cpu(mod_elem->value);
- break;
case AR_TKN_U32_MODULE_SRC_INSTANCE_ID:
src_mod_inst_id = le32_to_cpu(mod_elem->value);
break;
@@ -550,8 +572,6 @@ static struct audioreach_module *audioreach_parse_common_tokens(struct q6apm *ap
mod->module_id = module_id;
mod->max_ip_port = max_ip_port;
mod->max_op_port = max_op_port;
- mod->in_port = in_port;
- mod->out_port = out_port;
mod->src_mod_inst_id = src_mod_inst_id;
for (pn = 0; pn < mod->max_op_port; pn++) {
if (src_mod_op_port_id[pn] && dst_mod_inst_id[pn] &&
@@ -587,8 +607,10 @@ static int audioreach_widget_load_module_common(struct snd_soc_component *compon
return PTR_ERR(cont);
mod = audioreach_parse_common_tokens(apm, cont, &tplg_w->priv, w);
- if (IS_ERR(mod))
- return PTR_ERR(mod);
+ if (IS_ERR_OR_NULL(mod))
+ return mod ? PTR_ERR(mod) : -ENODEV;
+
+ mod->data = audioreach_get_module_priv_data(&tplg_w->priv);
dobj = &w->dobj;
dobj->private = mod;
@@ -947,6 +969,7 @@ static int audioreach_widget_unload(struct snd_soc_component *scomp,
cont->num_modules--;
list_del(&mod->node);
+ kfree(mod->data);
kfree(mod);
/* Graph Info has N sub-graphs, sub-graph has N containers, Container has N Modules */
if (list_empty(&cont->modules_list)) { /* if no modules in the container then remove it */
diff --git a/sound/soc/qcom/sc8280xp.c b/sound/soc/qcom/sc8280xp.c
index 73f9f82c4e25..78e327bc2f07 100644
--- a/sound/soc/qcom/sc8280xp.c
+++ b/sound/soc/qcom/sc8280xp.c
@@ -7,6 +7,7 @@
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
+#include <sound/pcm_params.h>
#include <linux/soundwire/sdw.h>
#include <sound/jack.h>
#include <linux/input-event-codes.h>
@@ -32,6 +33,10 @@ static int sc8280xp_snd_init(struct snd_soc_pcm_runtime *rtd)
int dp_pcm_id = 0;
switch (cpu_dai->id) {
+ case PRIMARY_MI2S_RX...QUATERNARY_MI2S_TX:
+ case QUINARY_MI2S_RX...QUINARY_MI2S_TX:
+ snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_BP_FP);
+ break;
case WSA_CODEC_DMA_RX_0:
case WSA_CODEC_DMA_RX_1:
/*
@@ -82,8 +87,10 @@ static int sc8280xp_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
SNDRV_PCM_HW_PARAM_RATE);
struct snd_interval *channels = hw_param_interval(params,
SNDRV_PCM_HW_PARAM_CHANNELS);
+ struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
rate->min = rate->max = 48000;
+ snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
channels->min = 2;
channels->max = 2;
switch (cpu_dai->id) {
@@ -186,9 +193,9 @@ static int sc8280xp_platform_probe(struct platform_device *pdev)
static const struct of_device_id snd_sc8280xp_dt_match[] = {
{.compatible = "qcom,qcm6490-idp-sndcard", "qcm6490"},
{.compatible = "qcom,qcs6490-rb3gen2-sndcard", "qcs6490"},
- {.compatible = "qcom,qcs8275-sndcard", "qcs8275"},
- {.compatible = "qcom,qcs9075-sndcard", "qcs9075"},
- {.compatible = "qcom,qcs9100-sndcard", "qcs9100"},
+ {.compatible = "qcom,qcs8275-sndcard", "qcs8300"},
+ {.compatible = "qcom,qcs9075-sndcard", "sa8775p"},
+ {.compatible = "qcom,qcs9100-sndcard", "sa8775p"},
{.compatible = "qcom,sc8280xp-sndcard", "sc8280xp"},
{.compatible = "qcom,sm8450-sndcard", "sm8450"},
{.compatible = "qcom,sm8550-sndcard", "sm8550"},
diff --git a/sound/soc/qcom/x1e80100.c b/sound/soc/qcom/x1e80100.c
index 8eb57fc12f0d..444f2162889f 100644
--- a/sound/soc/qcom/x1e80100.c
+++ b/sound/soc/qcom/x1e80100.c
@@ -210,14 +210,15 @@ static int x1e80100_platform_probe(struct platform_device *pdev)
if (ret)
return ret;
- card->driver_name = "x1e80100";
+ card->driver_name = of_device_get_match_data(dev);
x1e80100_add_be_ops(card);
return devm_snd_soc_register_card(dev, card);
}
static const struct of_device_id snd_x1e80100_dt_match[] = {
- { .compatible = "qcom,x1e80100-sndcard", },
+ { .compatible = "qcom,x1e80100-sndcard", .data = "x1e80100" },
+ { .compatible = "qcom,glymur-sndcard", .data = "glymur" },
{}
};
MODULE_DEVICE_TABLE(of, snd_x1e80100_dt_match);
generated by cgit (git 2.34.1) at 2025-10-04 03:22:55 +0000