summaryrefslogtreecommitdiff
path: root/sound/soc/qcom/qdsp6
diff options
context:
space:
mode:
Diffstat (limited to 'sound/soc/qcom/qdsp6')
-rw-r--r--sound/soc/qcom/qdsp6/Makefile4
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.c30
-rw-r--r--sound/soc/qcom/qdsp6/audioreach.h2
-rw-r--r--sound/soc/qcom/qdsp6/q6afe-dai.c16
-rw-r--r--sound/soc/qcom/qdsp6/q6apm-dai.c19
-rw-r--r--sound/soc/qcom/qdsp6/q6apm-lpass-dais.c53
-rw-r--r--sound/soc/qcom/qdsp6/q6asm-dai.c33
-rw-r--r--sound/soc/qcom/qdsp6/q6dsp-common.c2
-rw-r--r--sound/soc/qcom/qdsp6/q6routing.c2
-rw-r--r--sound/soc/qcom/qdsp6/topology.c38
10 files changed, 107 insertions, 92 deletions
diff --git a/sound/soc/qcom/qdsp6/Makefile b/sound/soc/qcom/qdsp6/Makefile
index 3963bf234664..26b7c55c9c11 100644
--- a/sound/soc/qcom/qdsp6/Makefile
+++ b/sound/soc/qcom/qdsp6/Makefile
@@ -1,6 +1,6 @@
# SPDX-License-Identifier: GPL-2.0-only
-snd-q6dsp-common-objs := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
-snd-q6apm-objs := q6apm.o audioreach.o topology.o
+snd-q6dsp-common-y := q6dsp-common.o q6dsp-lpass-ports.o q6dsp-lpass-clocks.o
+snd-q6apm-y := q6apm.o audioreach.o topology.o
obj-$(CONFIG_SND_SOC_QDSP6_COMMON) += snd-q6dsp-common.o
obj-$(CONFIG_SND_SOC_QDSP6_CORE) += q6core.o
diff --git a/sound/soc/qcom/qdsp6/audioreach.c b/sound/soc/qcom/qdsp6/audioreach.c
index 5291deac0a0b..4ebaaf736fb9 100644
--- a/sound/soc/qcom/qdsp6/audioreach.c
+++ b/sound/soc/qcom/qdsp6/audioreach.c
@@ -267,7 +267,7 @@ void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t token
}
EXPORT_SYMBOL_GPL(audioreach_alloc_apm_cmd_pkt);
-static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels)
+void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels)
{
if (num_channels == 1) {
ch_map[0] = PCM_CHANNEL_FL;
@@ -281,6 +281,7 @@ static void audioreach_set_channel_mapping(u8 *ch_map, int num_channels)
ch_map[3] = PCM_CHANNEL_RS;
}
}
+EXPORT_SYMBOL_GPL(audioreach_set_default_channel_mapping);
static void apm_populate_container_config(struct apm_container_obj *cfg,
struct audioreach_container *cont)
@@ -819,7 +820,7 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
uint32_t num_channels = cfg->num_channels;
int payload_size;
struct gpr_pkt *pkt;
- int rc;
+ int rc, i;
void *p;
payload_size = APM_MFC_CFG_PSIZE(media_format, num_channels) +
@@ -842,18 +843,8 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
media_format->sample_rate = cfg->sample_rate;
media_format->bit_width = cfg->bit_width;
media_format->num_channels = cfg->num_channels;
-
- if (num_channels == 1) {
- media_format->channel_mapping[0] = PCM_CHANNEL_FL;
- } else if (num_channels == 2) {
- media_format->channel_mapping[0] = PCM_CHANNEL_FL;
- media_format->channel_mapping[1] = PCM_CHANNEL_FR;
- } else if (num_channels == 4) {
- media_format->channel_mapping[0] = PCM_CHANNEL_FL;
- media_format->channel_mapping[1] = PCM_CHANNEL_FR;
- media_format->channel_mapping[2] = PCM_CHANNEL_LS;
- media_format->channel_mapping[3] = PCM_CHANNEL_RS;
- }
+ for (i = 0; i < num_channels; i++)
+ media_format->channel_mapping[i] = cfg->channel_map[i];
rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
@@ -883,9 +874,6 @@ static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
mp3_cfg->q_factor = mcfg->bit_width - 1;
mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
mp3_cfg->num_channels = mcfg->num_channels;
-
- audioreach_set_channel_mapping(mp3_cfg->channel_mapping,
- mcfg->num_channels);
break;
case SND_AUDIOCODEC_AAC:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
@@ -1104,9 +1092,7 @@ static int audioreach_pcm_set_media_format(struct q6apm_graph *graph,
media_cfg->num_channels = mcfg->num_channels;
media_cfg->q_factor = mcfg->bit_width - 1;
media_cfg->bits_per_sample = mcfg->bit_width;
-
- audioreach_set_channel_mapping(media_cfg->channel_mapping,
- num_channels);
+ memcpy(media_cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels);
rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
@@ -1163,9 +1149,7 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
cfg->q_factor = mcfg->bit_width - 1;
cfg->endianness = PCM_LITTLE_ENDIAN;
cfg->num_channels = mcfg->num_channels;
-
- audioreach_set_channel_mapping(cfg->channel_mapping,
- num_channels);
+ memcpy(cfg->channel_mapping, mcfg->channel_map, mcfg->num_channels);
} else {
rc = audioreach_set_compr_media_format(header, p, mcfg);
if (rc) {
diff --git a/sound/soc/qcom/qdsp6/audioreach.h b/sound/soc/qcom/qdsp6/audioreach.h
index 2c82917b7162..61a69df4f50f 100644
--- a/sound/soc/qcom/qdsp6/audioreach.h
+++ b/sound/soc/qcom/qdsp6/audioreach.h
@@ -755,7 +755,6 @@ struct audioreach_module_config {
u16 data_format;
u16 num_channels;
- u16 active_channels_mask;
u16 dp_idx;
u32 channel_allocation;
u32 sd_line_mask;
@@ -767,6 +766,7 @@ struct audioreach_module_config {
/* Packet Allocation routines */
void *audioreach_alloc_apm_cmd_pkt(int pkt_size, uint32_t opcode, uint32_t
token);
+void audioreach_set_default_channel_mapping(u8 *ch_map, int num_channels);
void *audioreach_alloc_cmd_pkt(int payload_size, uint32_t opcode,
uint32_t token, uint32_t src_port,
uint32_t dest_port);
diff --git a/sound/soc/qcom/qdsp6/q6afe-dai.c b/sound/soc/qcom/qdsp6/q6afe-dai.c
index a9c4f896a7df..7d9628cda875 100644
--- a/sound/soc/qcom/qdsp6/q6afe-dai.c
+++ b/sound/soc/qcom/qdsp6/q6afe-dai.c
@@ -172,8 +172,8 @@ static int q6tdm_set_tdm_slot(struct snd_soc_dai *dai,
}
static int q6tdm_set_channel_map(struct snd_soc_dai *dai,
- unsigned int tx_num, unsigned int *tx_slot,
- unsigned int rx_num, unsigned int *rx_slot)
+ unsigned int tx_num, const unsigned int *tx_slot,
+ unsigned int rx_num, const unsigned int *rx_slot)
{
struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
@@ -250,8 +250,10 @@ static int q6tdm_hw_params(struct snd_pcm_substream *substream,
}
static int q6dma_set_channel_map(struct snd_soc_dai *dai,
- unsigned int tx_num, unsigned int *tx_ch_mask,
- unsigned int rx_num, unsigned int *rx_ch_mask)
+ unsigned int tx_num,
+ const unsigned int *tx_ch_mask,
+ unsigned int rx_num,
+ const unsigned int *rx_ch_mask)
{
struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
@@ -407,8 +409,10 @@ static int q6afe_dai_prepare(struct snd_pcm_substream *substream,
}
static int q6slim_set_channel_map(struct snd_soc_dai *dai,
- unsigned int tx_num, unsigned int *tx_slot,
- unsigned int rx_num, unsigned int *rx_slot)
+ unsigned int tx_num,
+ const unsigned int *tx_slot,
+ unsigned int rx_num,
+ const unsigned int *rx_slot)
{
struct q6afe_dai_data *dai_data = dev_get_drvdata(dai->dev);
struct q6afe_port_config *pcfg = &dai_data->port_config[dai->id];
diff --git a/sound/soc/qcom/qdsp6/q6apm-dai.c b/sound/soc/qcom/qdsp6/q6apm-dai.c
index 00bbd291be5c..c9404b5934c7 100644
--- a/sound/soc/qcom/qdsp6/q6apm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6apm-dai.c
@@ -70,14 +70,10 @@ struct q6apm_dai_rtd {
unsigned int bytes_received;
unsigned int copied_total;
uint16_t bits_per_sample;
- uint16_t source; /* Encoding source bit mask */
- uint16_t session_id;
bool next_track;
enum stream_state state;
struct q6apm_graph *graph;
spinlock_t lock;
- uint32_t initial_samples_drop;
- uint32_t trailing_samples_drop;
bool notify_on_drain;
};
@@ -85,7 +81,7 @@ struct q6apm_dai_data {
long long sid;
};
-static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_capture = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
@@ -104,7 +100,7 @@ static struct snd_pcm_hardware q6apm_dai_hardware_capture = {
.fifo_size = 0,
};
-static struct snd_pcm_hardware q6apm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6apm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED |
SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
@@ -243,6 +239,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
cfg.num_channels = runtime->channels;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = SND_AUDIOCODEC_PCM;
+ audioreach_set_default_channel_mapping(cfg.channel_map, runtime->channels);
if (prtd->state) {
/* clear the previous setup if any */
@@ -331,7 +328,7 @@ static int q6apm_dai_open(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
- struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = snd_soc_substream_to_rtd(substream);
struct snd_soc_dai *cpu_dai = snd_soc_rtd_to_cpu(soc_prtd, 0);
struct device *dev = component->dev;
struct q6apm_dai_data *pdata;
@@ -669,6 +666,8 @@ static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
cfg.num_channels = 2;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = codec->id;
+ audioreach_set_default_channel_mapping(cfg.channel_map,
+ cfg.num_channels);
memcpy(&cfg.codec, codec, sizeof(*codec));
ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
@@ -720,14 +719,12 @@ static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
switch (metadata->key) {
case SNDRV_COMPRESS_ENCODER_PADDING:
- prtd->trailing_samples_drop = metadata->value[0];
q6apm_remove_trailing_silence(component->dev, prtd->graph,
- prtd->trailing_samples_drop);
+ metadata->value[0]);
break;
case SNDRV_COMPRESS_ENCODER_DELAY:
- prtd->initial_samples_drop = metadata->value[0];
q6apm_remove_initial_silence(component->dev, prtd->graph,
- prtd->initial_samples_drop);
+ metadata->value[0]);
break;
default:
ret = -EINVAL;
diff --git a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
index 68a38f63a2db..9c98a35ad099 100644
--- a/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
+++ b/sound/soc/qcom/qdsp6/q6apm-lpass-dais.c
@@ -25,13 +25,15 @@ struct q6apm_lpass_dai_data {
};
static int q6dma_set_channel_map(struct snd_soc_dai *dai,
- unsigned int tx_num, unsigned int *tx_ch_mask,
- unsigned int rx_num, unsigned int *rx_ch_mask)
+ unsigned int tx_num,
+ const unsigned int *tx_ch_mask,
+ unsigned int rx_num,
+ const unsigned int *rx_ch_mask)
{
struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
- int ch_mask;
+ int i;
switch (dai->id) {
case WSA_CODEC_DMA_TX_0:
@@ -56,7 +58,8 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
tx_num);
return -EINVAL;
}
- ch_mask = *tx_ch_mask;
+ for (i = 0; i < tx_num; i++)
+ cfg->channel_map[i] = tx_ch_mask[i];
break;
case WSA_CODEC_DMA_RX_0:
@@ -79,7 +82,8 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
rx_num);
return -EINVAL;
}
- ch_mask = *rx_ch_mask;
+ for (i = 0; i < rx_num; i++)
+ cfg->channel_map[i] = rx_ch_mask[i];
break;
default:
@@ -88,8 +92,6 @@ static int q6dma_set_channel_map(struct snd_soc_dai *dai,
return -EINVAL;
}
- cfg->active_channels_mask = ch_mask;
-
return 0;
}
@@ -104,6 +106,7 @@ static int q6hdmi_hw_params(struct snd_pcm_substream *substream,
cfg->bit_width = params_width(params);
cfg->sample_rate = params_rate(params);
cfg->num_channels = channels;
+ audioreach_set_default_channel_mapping(cfg->channel_map, channels);
switch (dai->id) {
case DISPLAY_PORT_RX_0:
@@ -128,10 +131,12 @@ static int q6dma_hw_params(struct snd_pcm_substream *substream,
{
struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
struct audioreach_module_config *cfg = &dai_data->module_config[dai->id];
+ int channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max;
cfg->bit_width = params_width(params);
cfg->sample_rate = params_rate(params);
- cfg->num_channels = hw_param_interval_c(params, SNDRV_PCM_HW_PARAM_CHANNELS)->max;
+ cfg->num_channels = channels;
+ audioreach_set_default_channel_mapping(cfg->channel_map, channels);
return 0;
}
@@ -141,14 +146,17 @@ static void q6apm_lpass_dai_shutdown(struct snd_pcm_substream *substream, struct
struct q6apm_lpass_dai_data *dai_data = dev_get_drvdata(dai->dev);
int rc;
- if (!dai_data->is_port_started[dai->id])
- return;
- rc = q6apm_graph_stop(dai_data->graph[dai->id]);
- if (rc < 0)
- dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+ if (dai_data->is_port_started[dai->id]) {
+ rc = q6apm_graph_stop(dai_data->graph[dai->id]);
+ dai_data->is_port_started[dai->id] = false;
+ if (rc < 0)
+ dev_err(dai->dev, "fail to close APM port (%d)\n", rc);
+ }
- q6apm_graph_close(dai_data->graph[dai->id]);
- dai_data->is_port_started[dai->id] = false;
+ if (dai_data->graph[dai->id]) {
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
}
static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
@@ -163,8 +171,10 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
q6apm_graph_stop(dai_data->graph[dai->id]);
dai_data->is_port_started[dai->id] = false;
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ }
}
/**
@@ -183,26 +193,29 @@ static int q6apm_lpass_dai_prepare(struct snd_pcm_substream *substream, struct s
cfg->direction = substream->stream;
rc = q6apm_graph_media_format_pcm(dai_data->graph[dai->id], cfg);
-
if (rc) {
dev_err(dai->dev, "Failed to set media format %d\n", rc);
- return rc;
+ goto err;
}
rc = q6apm_graph_prepare(dai_data->graph[dai->id]);
if (rc) {
dev_err(dai->dev, "Failed to prepare Graph %d\n", rc);
- return rc;
+ goto err;
}
rc = q6apm_graph_start(dai_data->graph[dai->id]);
if (rc < 0) {
dev_err(dai->dev, "fail to start APM port %x\n", dai->id);
- return rc;
+ goto err;
}
dai_data->is_port_started[dai->id] = true;
return 0;
+err:
+ q6apm_graph_close(dai_data->graph[dai->id]);
+ dai_data->graph[dai->id] = NULL;
+ return rc;
}
static int q6apm_lpass_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai)
diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c
index aeb6a9d479ab..045100c94352 100644
--- a/sound/soc/qcom/qdsp6/q6asm-dai.c
+++ b/sound/soc/qcom/qdsp6/q6asm-dai.c
@@ -103,7 +103,7 @@ static const struct snd_pcm_hardware q6asm_dai_hardware_capture = {
.fifo_size = 0,
};
-static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
+static const struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_BATCH |
SNDRV_PCM_INFO_BLOCK_TRANSFER |
SNDRV_PCM_INFO_MMAP_VALID |
@@ -128,8 +128,13 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
#define Q6ASM_FEDAI_DRIVER(num) { \
.playback = { \
.stream_name = "MultiMedia"#num" Playback", \
- .rates = (SNDRV_PCM_RATE_8000_192000| \
- SNDRV_PCM_RATE_KNOT), \
+ .rates = (SNDRV_PCM_RATE_8000_48000 | \
+ SNDRV_PCM_RATE_12000 | \
+ SNDRV_PCM_RATE_24000 | \
+ SNDRV_PCM_RATE_88200 | \
+ SNDRV_PCM_RATE_96000 | \
+ SNDRV_PCM_RATE_176400 | \
+ SNDRV_PCM_RATE_192000), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
@@ -139,8 +144,9 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
}, \
.capture = { \
.stream_name = "MultiMedia"#num" Capture", \
- .rates = (SNDRV_PCM_RATE_8000_48000| \
- SNDRV_PCM_RATE_KNOT), \
+ .rates = (SNDRV_PCM_RATE_8000_48000 | \
+ SNDRV_PCM_RATE_12000 | \
+ SNDRV_PCM_RATE_24000), \
.formats = (SNDRV_PCM_FMTBIT_S16_LE | \
SNDRV_PCM_FMTBIT_S24_LE), \
.channels_min = 1, \
@@ -152,18 +158,6 @@ static struct snd_pcm_hardware q6asm_dai_hardware_playback = {
.id = MSM_FRONTEND_DAI_MULTIMEDIA##num, \
}
-/* Conventional and unconventional sample rate supported */
-static unsigned int supported_sample_rates[] = {
- 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
- 88200, 96000, 176400, 192000
-};
-
-static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
- .count = ARRAY_SIZE(supported_sample_rates),
- .list = supported_sample_rates,
- .mask = 0,
-};
-
static const struct snd_compr_codec_caps q6asm_compr_caps = {
.num_descriptors = 1,
.descriptor[0].max_ch = 2,
@@ -390,11 +384,6 @@ static int q6asm_dai_open(struct snd_soc_component *component,
else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
runtime->hw = q6asm_dai_hardware_capture;
- ret = snd_pcm_hw_constraint_list(runtime, 0,
- SNDRV_PCM_HW_PARAM_RATE,
- &constraints_sample_rates);
- if (ret < 0)
- dev_info(dev, "snd_pcm_hw_constraint_list failed\n");
/* Ensure that buffer size is a multiple of period size */
ret = snd_pcm_hw_constraint_integer(runtime,
SNDRV_PCM_HW_PARAM_PERIODS);
diff --git a/sound/soc/qcom/qdsp6/q6dsp-common.c b/sound/soc/qcom/qdsp6/q6dsp-common.c
index 95585dea2b36..f74585d88bd6 100644
--- a/sound/soc/qcom/qdsp6/q6dsp-common.c
+++ b/sound/soc/qcom/qdsp6/q6dsp-common.c
@@ -98,4 +98,6 @@ int q6dsp_get_channel_allocation(int channels)
return channel_allocation;
}
EXPORT_SYMBOL_GPL(q6dsp_get_channel_allocation);
+
+MODULE_DESCRIPTION("ASoC MSM QDSP6 helper functions");
MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/qcom/qdsp6/q6routing.c b/sound/soc/qcom/qdsp6/q6routing.c
index 81fde0681f95..90228699ba7d 100644
--- a/sound/soc/qcom/qdsp6/q6routing.c
+++ b/sound/soc/qcom/qdsp6/q6routing.c
@@ -1161,7 +1161,7 @@ static struct platform_driver q6pcm_routing_platform_driver = {
.of_match_table = of_match_ptr(q6pcm_routing_device_id),
},
.probe = q6pcm_routing_probe,
- .remove_new = q6pcm_routing_remove,
+ .remove = q6pcm_routing_remove,
};
module_platform_driver(q6pcm_routing_platform_driver);
diff --git a/sound/soc/qcom/qdsp6/topology.c b/sound/soc/qcom/qdsp6/topology.c
index 70572c83e101..83319a928f29 100644
--- a/sound/soc/qcom/qdsp6/topology.c
+++ b/sound/soc/qcom/qdsp6/topology.c
@@ -1,6 +1,7 @@
// SPDX-License-Identifier: GPL-2.0
// Copyright (c) 2020, Linaro Limited
+#include <linux/cleanup.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <sound/pcm.h>
@@ -730,6 +731,29 @@ static int audioreach_widget_i2s_module_load(struct audioreach_module *mod,
return 0;
}
+static int audioreach_widget_dp_module_load(struct audioreach_module *mod,
+ struct snd_soc_tplg_vendor_array *mod_array)
+{
+ struct snd_soc_tplg_vendor_value_elem *mod_elem;
+ int tkn_count = 0;
+
+ mod_elem = mod_array->value;
+
+ while (tkn_count <= (le32_to_cpu(mod_array->num_elems) - 1)) {
+ switch (le32_to_cpu(mod_elem->token)) {
+ case AR_TKN_U32_MODULE_FMT_DATA:
+ mod->data_format = le32_to_cpu(mod_elem->value);
+ break;
+ default:
+ break;
+ }
+ tkn_count++;
+ mod_elem++;
+ }
+
+ return 0;
+}
+
static int audioreach_widget_load_buffer(struct snd_soc_component *component,
int index, struct snd_soc_dapm_widget *w,
struct snd_soc_tplg_dapm_widget *tplg_w)
@@ -760,6 +784,9 @@ static int audioreach_widget_load_buffer(struct snd_soc_component *component,
case MODULE_ID_I2S_SOURCE:
audioreach_widget_i2s_module_load(mod, mod_array);
break;
+ case MODULE_ID_DISPLAY_PORT_SINK:
+ audioreach_widget_dp_module_load(mod, mod_array);
+ break;
default:
return -EINVAL;
}
@@ -1240,7 +1267,7 @@ static const struct snd_soc_tplg_kcontrol_ops audioreach_io_ops[] = {
audioreach_put_vol_ctrl_audio_mixer, snd_soc_info_volsw},
};
-static struct snd_soc_tplg_ops audioreach_tplg_ops = {
+static const struct snd_soc_tplg_ops audioreach_tplg_ops = {
.io_ops = audioreach_io_ops,
.io_ops_count = ARRAY_SIZE(audioreach_io_ops),
@@ -1262,18 +1289,19 @@ int audioreach_tplg_init(struct snd_soc_component *component)
struct snd_soc_card *card = component->card;
struct device *dev = component->dev;
const struct firmware *fw;
- char *tplg_fw_name;
int ret;
/* Inline with Qualcomm UCM configs and linux-firmware path */
- tplg_fw_name = kasprintf(GFP_KERNEL, "qcom/%s/%s-tplg.bin", card->driver_name, card->name);
+ char *tplg_fw_name __free(kfree) = kasprintf(GFP_KERNEL, "qcom/%s/%s-tplg.bin",
+ card->driver_name,
+ card->name);
if (!tplg_fw_name)
return -ENOMEM;
ret = request_firmware(&fw, tplg_fw_name, dev);
if (ret < 0) {
dev_err(dev, "tplg firmware loading %s failed %d\n", tplg_fw_name, ret);
- goto err;
+ return ret;
}
ret = snd_soc_tplg_component_load(component, &audioreach_tplg_ops, fw);
@@ -1283,8 +1311,6 @@ int audioreach_tplg_init(struct snd_soc_component *component)
}
release_firmware(fw);
-err:
- kfree(tplg_fw_name);
return ret;
}
generated by cgit (git 2.34.1) at 2025-08-01 00:45:49 +0000