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ASoC: qcom: audioreach: add compress offload 

Merge series from Srinivas Kandagatla <srinivas.kandagatla@linaro.org>:

This patchset adds compressed offload support to Qualcomm audioreach drivers.
Currently it supports AAC, MP3 and FALC along with gapless.

Tested this on SM8450 and sc7280.
pull/877/head
Mark Brown 2023-06-20 15:24:06 +01:00
parent 29735f6fb0 c317d148a2
commit d4b2aee1be
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
6 changed files with 745 additions and 94 deletions
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246
sound/soc/qcom/qdsp6/audioreach.c
View File

@ -732,33 +732,32 @@ static int audioreach_codec_dma_set_media_format(struct q6apm_graph *graph,
return rc;
}
static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
struct audioreach_module *module, bool enable)
int audioreach_send_u32_param(struct q6apm_graph *graph, struct audioreach_module *module,
uint32_t param_id, uint32_t param_val)
{
struct apm_module_param_data *param_data;
struct param_id_sal_limiter_enable *limiter_enable;
int payload_size;
struct gpr_pkt *pkt;
int rc;
uint32_t *param;
int rc, payload_size;
void *p;
payload_size = sizeof(*limiter_enable) + APM_MODULE_PARAM_DATA_SIZE;
payload_size = sizeof(uint32_t) + APM_MODULE_PARAM_DATA_SIZE;
p = audioreach_alloc_apm_cmd_pkt(payload_size, APM_CMD_SET_CFG, 0);
if (IS_ERR(p))
return -ENOMEM;
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;
pkt = p;
p = p + GPR_HDR_SIZE + APM_CMD_HDR_SIZE;
param_data = p;
param_data->module_instance_id = module->instance_id;
param_data->error_code = 0;
param_data->param_id = PARAM_ID_SAL_LIMITER_ENABLE;
param_data->param_size = sizeof(*limiter_enable);
p = p + APM_MODULE_PARAM_DATA_SIZE;
limiter_enable = p;
param_data->param_id = param_id;
param_data->param_size = sizeof(uint32_t);
limiter_enable->enable_lim = enable;
p = p + APM_MODULE_PARAM_DATA_SIZE;
param = p;
*param = param_val;
rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
@ -766,77 +765,34 @@ static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
return rc;
}
EXPORT_SYMBOL_GPL(audioreach_send_u32_param);
static int audioreach_sal_limiter_enable(struct q6apm_graph *graph,
struct audioreach_module *module, bool enable)
{
return audioreach_send_u32_param(graph, module, PARAM_ID_SAL_LIMITER_ENABLE, enable);
}
static int audioreach_sal_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
{
struct apm_module_param_data *param_data;
struct param_id_sal_output_config *media_format;
int payload_size;
struct gpr_pkt *pkt;
int rc;
void *p;
payload_size = sizeof(*media_format) + APM_MODULE_PARAM_DATA_SIZE;
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;
param_data = p;
param_data->module_instance_id = module->instance_id;
param_data->error_code = 0;
param_data->param_id = PARAM_ID_SAL_OUTPUT_CFG;
param_data->param_size = sizeof(*media_format);
p = p + APM_MODULE_PARAM_DATA_SIZE;
media_format = p;
media_format->bits_per_sample = cfg->bit_width;
rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
kfree(pkt);
return rc;
return audioreach_send_u32_param(graph, module, PARAM_ID_SAL_OUTPUT_CFG, cfg->bit_width);
}
static int audioreach_module_enable(struct q6apm_graph *graph,
struct audioreach_module *module,
bool enable)
{
struct apm_module_param_data *param_data;
struct param_id_module_enable *param;
int payload_size;
struct gpr_pkt *pkt;
int rc;
void *p;
return audioreach_send_u32_param(graph, module, PARAM_ID_MODULE_ENABLE, enable);
}
payload_size = sizeof(*param) + APM_MODULE_PARAM_DATA_SIZE;
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;
param_data = p;
param_data->module_instance_id = module->instance_id;
param_data->error_code = 0;
param_data->param_id = PARAM_ID_MODULE_ENABLE;
param_data->param_size = sizeof(*param);
p = p + APM_MODULE_PARAM_DATA_SIZE;
param = p;
param->enable = enable;
rc = q6apm_send_cmd_sync(graph->apm, pkt, 0);
kfree(pkt);
return rc;
static int audioreach_gapless_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
{
return audioreach_send_u32_param(graph, module, PARAM_ID_EARLY_EOS_DELAY,
EARLY_EOS_DELAY_MS);
}
static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
@ -886,6 +842,99 @@ static int audioreach_mfc_set_media_format(struct q6apm_graph *graph,
return rc;
}
static int audioreach_set_compr_media_format(struct media_format *media_fmt_hdr,
void *p, struct audioreach_module_config *mcfg)
{
struct payload_media_fmt_aac_t *aac_cfg;
struct payload_media_fmt_pcm *mp3_cfg;
struct payload_media_fmt_flac_t *flac_cfg;
switch (mcfg->fmt) {
case SND_AUDIOCODEC_MP3:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_MP3;
media_fmt_hdr->payload_size = 0;
p = p + sizeof(*media_fmt_hdr);
mp3_cfg = p;
mp3_cfg->sample_rate = mcfg->sample_rate;
mp3_cfg->bit_width = mcfg->bit_width;
mp3_cfg->alignment = PCM_LSB_ALIGNED;
mp3_cfg->bits_per_sample = mcfg->bit_width;
mp3_cfg->q_factor = mcfg->bit_width - 1;
mp3_cfg->endianness = PCM_LITTLE_ENDIAN;
mp3_cfg->num_channels = mcfg->num_channels;
if (mcfg->num_channels == 1) {
mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
} else if (mcfg->num_channels == 2) {
mp3_cfg->channel_mapping[0] = PCM_CHANNEL_L;
mp3_cfg->channel_mapping[1] = PCM_CHANNEL_R;
}
break;
case SND_AUDIOCODEC_AAC:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_AAC;
media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_aac_t);
p = p + sizeof(*media_fmt_hdr);
aac_cfg = p;
aac_cfg->aac_fmt_flag = 0;
aac_cfg->audio_obj_type = 5;
aac_cfg->num_channels = mcfg->num_channels;
aac_cfg->total_size_of_PCE_bits = 0;
aac_cfg->sample_rate = mcfg->sample_rate;
break;
case SND_AUDIOCODEC_FLAC:
media_fmt_hdr->data_format = DATA_FORMAT_RAW_COMPRESSED;
media_fmt_hdr->fmt_id = MEDIA_FMT_ID_FLAC;
media_fmt_hdr->payload_size = sizeof(struct payload_media_fmt_flac_t);
p = p + sizeof(*media_fmt_hdr);
flac_cfg = p;
flac_cfg->sample_size = mcfg->codec.options.flac_d.sample_size;
flac_cfg->num_channels = mcfg->num_channels;
flac_cfg->min_blk_size = mcfg->codec.options.flac_d.min_blk_size;
flac_cfg->max_blk_size = mcfg->codec.options.flac_d.max_blk_size;
flac_cfg->sample_rate = mcfg->sample_rate;
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;
default:
return -EINVAL;
}
return 0;
}
int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg)
{
struct media_format *header;
struct gpr_pkt *pkt;
int iid, payload_size, rc;
void *p;
payload_size = sizeof(struct apm_sh_module_media_fmt_cmd);
iid = q6apm_graph_get_rx_shmem_module_iid(graph);
pkt = audioreach_alloc_cmd_pkt(payload_size, DATA_CMD_WR_SH_MEM_EP_MEDIA_FORMAT,
0, graph->port->id, iid);
if (IS_ERR(pkt))
return -ENOMEM;
p = (void *)pkt + GPR_HDR_SIZE;
header = p;
rc = audioreach_set_compr_media_format(header, p, mcfg);
if (rc) {
kfree(pkt);
return rc;
}
rc = gpr_send_port_pkt(graph->port, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(audioreach_compr_set_param);
static int audioreach_i2s_set_media_format(struct q6apm_graph *graph,
struct audioreach_module *module,
struct audioreach_module_config *cfg)
@ -1089,25 +1138,33 @@ static int audioreach_shmem_set_media_format(struct q6apm_graph *graph,
p = p + APM_MODULE_PARAM_DATA_SIZE;
header = p;
header->data_format = DATA_FORMAT_FIXED_POINT;
header->fmt_id = MEDIA_FMT_ID_PCM;
header->payload_size = payload_size - sizeof(*header);
if (mcfg->fmt == SND_AUDIOCODEC_PCM) {
header->data_format = DATA_FORMAT_FIXED_POINT;
header->fmt_id = MEDIA_FMT_ID_PCM;
header->payload_size = payload_size - sizeof(*header);
p = p + sizeof(*header);
cfg = p;
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->q_factor = mcfg->bit_width - 1;
cfg->endianness = PCM_LITTLE_ENDIAN;
cfg->num_channels = mcfg->num_channels;
p = p + sizeof(*header);
cfg = p;
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->q_factor = mcfg->bit_width - 1;
cfg->endianness = PCM_LITTLE_ENDIAN;
cfg->num_channels = mcfg->num_channels;
if (mcfg->num_channels == 1) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
} else if (num_channels == 2) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
cfg->channel_mapping[1] = PCM_CHANNEL_R;
if (mcfg->num_channels == 1)
cfg->channel_mapping[0] = PCM_CHANNEL_L;
else if (num_channels == 2) {
cfg->channel_mapping[0] = PCM_CHANNEL_L;
cfg->channel_mapping[1] = PCM_CHANNEL_R;
}
} else {
rc = audioreach_set_compr_media_format(header, p, mcfg);
if (rc) {
kfree(pkt);
return rc;
}
}
rc = audioreach_graph_send_cmd_sync(graph, pkt, 0);
@ -1192,6 +1249,8 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
case MODULE_ID_PCM_DEC:
case MODULE_ID_PCM_ENC:
case MODULE_ID_PCM_CNV:
case MODULE_ID_PLACEHOLDER_DECODER:
case MODULE_ID_PLACEHOLDER_ENCODER:
rc = audioreach_pcm_set_media_format(graph, module, cfg);
break;
case MODULE_ID_DISPLAY_PORT_SINK:
@ -1219,6 +1278,9 @@ int audioreach_set_media_format(struct q6apm_graph *graph, struct audioreach_mod
case MODULE_ID_MFC:
rc = audioreach_mfc_set_media_format(graph, module, cfg);
break;
case MODULE_ID_GAPLESS:
rc = audioreach_gapless_set_media_format(graph, module, cfg);
break;
default:
rc = 0;
}

51
sound/soc/qcom/qdsp6/audioreach.h
View File

@ -15,6 +15,8 @@ struct q6apm_graph;
#define MODULE_ID_PCM_CNV 0x07001003
#define MODULE_ID_PCM_ENC 0x07001004
#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
@ -22,6 +24,10 @@ struct q6apm_graph;
#define MODULE_ID_I2S_SINK 0x0700100A
#define MODULE_ID_I2S_SOURCE 0x0700100B
#define MODULE_ID_DATA_LOGGING 0x0700101A
#define MODULE_ID_AAC_DEC 0x0700101F
#define MODULE_ID_FLAC_DEC 0x0700102F
#define MODULE_ID_MP3_DECODE 0x0700103B
#define MODULE_ID_GAPLESS 0x0700104D
#define MODULE_ID_DISPLAY_PORT_SINK 0x07001069
#define APM_CMD_GET_SPF_STATE 0x01001021
@ -143,12 +149,15 @@ struct param_id_enc_bitrate_param {
} __packed;
#define DATA_FORMAT_FIXED_POINT 1
#define DATA_FORMAT_GENERIC_COMPRESSED 5
#define DATA_FORMAT_RAW_COMPRESSED 6
#define PCM_LSB_ALIGNED 1
#define PCM_MSB_ALIGNED 2
#define PCM_LITTLE_ENDIAN 1
#define PCM_BIT_ENDIAN 2
#define MEDIA_FMT_ID_PCM 0x09001000
#define MEDIA_FMT_ID_MP3 0x09001009
#define PCM_CHANNEL_L 1
#define PCM_CHANNEL_R 2
#define SAMPLE_RATE_48K 48000
@ -226,6 +235,28 @@ struct apm_media_format {
uint32_t payload_size;
} __packed;
#define MEDIA_FMT_ID_FLAC 0x09001004
struct payload_media_fmt_flac_t {
uint16_t num_channels;
uint16_t sample_size;
uint16_t min_blk_size;
uint16_t max_blk_size;
uint32_t sample_rate;
uint32_t min_frame_size;
uint32_t max_frame_size;
} __packed;
#define MEDIA_FMT_ID_AAC 0x09001001
struct payload_media_fmt_aac_t {
uint16_t aac_fmt_flag;
uint16_t audio_obj_type;
uint16_t num_channels;
uint16_t total_size_of_PCE_bits;
uint32_t sample_rate;
} __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
@ -522,6 +553,8 @@ struct param_id_sal_limiter_enable {
} __packed;
#define PARAM_ID_MFC_OUTPUT_MEDIA_FORMAT 0x08001024
#define PARAM_ID_EARLY_EOS_DELAY 0x0800114C
#define EARLY_EOS_DELAY_MS 150
struct param_id_mfc_media_format {
uint32_t sample_rate;
@ -530,6 +563,10 @@ struct param_id_mfc_media_format {
uint16_t channel_mapping[];
} __packed;
struct param_id_gapless_early_eos_delay_t {
uint32_t early_eos_delay_ms;
} __packed;
struct media_format {
uint32_t data_format;
uint32_t fmt_id;
@ -608,6 +645,15 @@ struct param_id_vol_ctrl_master_gain {
} __packed;
#define PARAM_ID_REMOVE_INITIAL_SILENCE 0x0800114B
#define PARAM_ID_REMOVE_TRAILING_SILENCE 0x0800115D
#define PARAM_ID_REAL_MODULE_ID 0x0800100B
struct param_id_placeholder_real_module_id {
uint32_t real_module_id;
} __packed;
/* Graph */
struct audioreach_connection {
/* Connections */
@ -716,6 +762,7 @@ struct audioreach_module_config {
u32 channel_allocation;
u32 sd_line_mask;
int fmt;
struct snd_codec codec;
u8 channel_map[AR_PCM_MAX_NUM_CHANNEL];
};
@ -752,4 +799,8 @@ int audioreach_set_media_format(struct q6apm_graph *graph,
int audioreach_shared_memory_send_eos(struct q6apm_graph *graph);
int audioreach_gain_set_vol_ctrl(struct q6apm *apm,
struct audioreach_module *module, int vol);
int audioreach_send_u32_param(struct q6apm_graph *graph, struct audioreach_module *module,
uint32_t param_id, uint32_t param_val);
int audioreach_compr_set_param(struct q6apm_graph *graph, struct audioreach_module_config *mcfg);
#endif /* __AUDIOREACH_H__ */

445
sound/soc/qcom/qdsp6/q6apm-dai.c
View File

@ -28,8 +28,27 @@
#define CAPTURE_MIN_PERIOD_SIZE 320
#define BUFFER_BYTES_MAX (PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE)
#define BUFFER_BYTES_MIN (PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE)
#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
#define SID_MASK_DEFAULT 0xF
static const struct snd_compr_codec_caps q6apm_compr_caps = {
.num_descriptors = 1,
.descriptor[0].max_ch = 2,
.descriptor[0].sample_rates = { 8000, 11025, 12000, 16000, 22050,
24000, 32000, 44100, 48000, 88200,
96000, 176400, 192000 },
.descriptor[0].num_sample_rates = 13,
.descriptor[0].bit_rate[0] = 320,
.descriptor[0].bit_rate[1] = 128,
.descriptor[0].num_bitrates = 2,
.descriptor[0].profiles = 0,
.descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO,
.descriptor[0].formats = 0,
};
enum stream_state {
Q6APM_STREAM_IDLE = 0,
Q6APM_STREAM_STOPPED,
@ -39,6 +58,7 @@ enum stream_state {
struct q6apm_dai_rtd {
struct snd_pcm_substream *substream;
struct snd_compr_stream *cstream;
struct snd_codec codec;
struct snd_compr_params codec_param;
struct snd_dma_buffer dma_buffer;
phys_addr_t phys;
@ -52,9 +72,13 @@ struct q6apm_dai_rtd {
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;
};
struct q6apm_dai_data {
@ -132,6 +156,69 @@ static void event_handler(uint32_t opcode, uint32_t token, uint32_t *payload, vo
}
}
static void event_handler_compr(uint32_t opcode, uint32_t token,
uint32_t *payload, void *priv)
{
struct q6apm_dai_rtd *prtd = priv;
struct snd_compr_stream *substream = prtd->cstream;
unsigned long flags;
uint32_t wflags = 0;
uint64_t avail;
uint32_t bytes_written, bytes_to_write;
bool is_last_buffer = false;
switch (opcode) {
case APM_CLIENT_EVENT_CMD_EOS_DONE:
spin_lock_irqsave(&prtd->lock, flags);
if (prtd->notify_on_drain) {
snd_compr_drain_notify(prtd->cstream);
prtd->notify_on_drain = false;
} else {
prtd->state = Q6APM_STREAM_STOPPED;
}
spin_unlock_irqrestore(&prtd->lock, flags);
break;
case APM_CLIENT_EVENT_DATA_WRITE_DONE:
spin_lock_irqsave(&prtd->lock, flags);
bytes_written = token >> APM_WRITE_TOKEN_LEN_SHIFT;
prtd->copied_total += bytes_written;
snd_compr_fragment_elapsed(substream);
if (prtd->state != Q6APM_STREAM_RUNNING) {
spin_unlock_irqrestore(&prtd->lock, flags);
break;
}
avail = prtd->bytes_received - prtd->bytes_sent;
if (avail > prtd->pcm_count) {
bytes_to_write = prtd->pcm_count;
} else {
if (substream->partial_drain || prtd->notify_on_drain)
is_last_buffer = true;
bytes_to_write = avail;
}
if (bytes_to_write) {
if (substream->partial_drain && is_last_buffer)
wflags |= APM_LAST_BUFFER_FLAG;
q6apm_write_async(prtd->graph,
bytes_to_write, 0, 0, wflags);
prtd->bytes_sent += bytes_to_write;
if (prtd->notify_on_drain && is_last_buffer)
audioreach_shared_memory_send_eos(prtd->graph);
}
spin_unlock_irqrestore(&prtd->lock, flags);
break;
default:
break;
}
}
static int q6apm_dai_prepare(struct snd_soc_component *component,
struct snd_pcm_substream *substream)
{
@ -155,6 +242,7 @@ static int q6apm_dai_prepare(struct snd_soc_component *component,
cfg.sample_rate = runtime->rate;
cfg.num_channels = runtime->channels;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = SND_AUDIOCODEC_PCM;
if (prtd->state) {
/* clear the previous setup if any */
@ -386,6 +474,362 @@ static int q6apm_dai_pcm_new(struct snd_soc_component *component, struct snd_soc
return snd_pcm_set_fixed_buffer_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, component->dev, size);
}
static int q6apm_dai_compr_open(struct snd_soc_component *component,
struct snd_compr_stream *stream)
{
struct snd_soc_pcm_runtime *rtd = stream->private_data;
struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd;
struct q6apm_dai_data *pdata;
struct device *dev = component->dev;
int ret, size;
int graph_id;
graph_id = cpu_dai->driver->id;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata)
return -EINVAL;
prtd = kzalloc(sizeof(*prtd), GFP_KERNEL);
if (prtd == NULL)
return -ENOMEM;
prtd->cstream = stream;
prtd->graph = q6apm_graph_open(dev, (q6apm_cb)event_handler_compr, prtd, graph_id);
if (IS_ERR(prtd->graph)) {
ret = PTR_ERR(prtd->graph);
kfree(prtd);
return ret;
}
runtime->private_data = prtd;
runtime->dma_bytes = BUFFER_BYTES_MAX;
size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE * COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
ret = snd_dma_alloc_pages(SNDRV_DMA_TYPE_DEV, dev, size, &prtd->dma_buffer);
if (ret)
return ret;
if (pdata->sid < 0)
prtd->phys = prtd->dma_buffer.addr;
else
prtd->phys = prtd->dma_buffer.addr | (pdata->sid << 32);
snd_compr_set_runtime_buffer(stream, &prtd->dma_buffer);
spin_lock_init(&prtd->lock);
q6apm_enable_compress_module(dev, prtd->graph, true);
return 0;
}
static int q6apm_dai_compr_free(struct snd_soc_component *component,
struct snd_compr_stream *stream)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
q6apm_graph_stop(prtd->graph);
q6apm_unmap_memory_regions(prtd->graph, SNDRV_PCM_STREAM_PLAYBACK);
q6apm_graph_close(prtd->graph);
snd_dma_free_pages(&prtd->dma_buffer);
prtd->graph = NULL;
kfree(prtd);
runtime->private_data = NULL;
return 0;
}
static int q6apm_dai_compr_get_caps(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_caps *caps)
{
caps->direction = SND_COMPRESS_PLAYBACK;
caps->min_fragment_size = COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
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->codecs[0] = SND_AUDIOCODEC_MP3;
caps->codecs[1] = SND_AUDIOCODEC_AAC;
caps->codecs[2] = SND_AUDIOCODEC_FLAC;
return 0;
}
static int q6apm_dai_compr_get_codec_caps(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_codec_caps *codec)
{
switch (codec->codec) {
case SND_AUDIOCODEC_MP3:
*codec = q6apm_compr_caps;
break;
default:
break;
}
return 0;
}
static int q6apm_dai_compr_pointer(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_tstamp *tstamp)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
spin_lock_irqsave(&prtd->lock, flags);
tstamp->copied_total = prtd->copied_total;
tstamp->byte_offset = prtd->copied_total % prtd->pcm_size;
spin_unlock_irqrestore(&prtd->lock, flags);
return 0;
}
static int q6apm_dai_compr_trigger(struct snd_soc_component *component,
struct snd_compr_stream *stream, int cmd)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
int ret = 0;
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
ret = q6apm_write_async(prtd->graph, prtd->pcm_count, 0, 0, NO_TIMESTAMP);
break;
case SNDRV_PCM_TRIGGER_STOP:
break;
case SNDRV_PCM_TRIGGER_SUSPEND:
case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
break;
case SND_COMPR_TRIGGER_NEXT_TRACK:
prtd->next_track = true;
break;
case SND_COMPR_TRIGGER_DRAIN:
case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
prtd->notify_on_drain = true;
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6apm_dai_compr_ack(struct snd_soc_component *component, struct snd_compr_stream *stream,
size_t count)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
unsigned long flags;
spin_lock_irqsave(&prtd->lock, flags);
prtd->bytes_received += count;
spin_unlock_irqrestore(&prtd->lock, flags);
return count;
}
static int q6apm_dai_compr_set_params(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_params *params)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
struct q6apm_dai_data *pdata;
struct audioreach_module_config cfg;
struct snd_codec *codec = &params->codec;
int dir = stream->direction;
int ret;
pdata = snd_soc_component_get_drvdata(component);
if (!pdata)
return -EINVAL;
prtd->periods = runtime->fragments;
prtd->pcm_count = runtime->fragment_size;
prtd->pcm_size = runtime->fragments * runtime->fragment_size;
prtd->bits_per_sample = 16;
prtd->pos = 0;
if (prtd->next_track != true) {
memcpy(&prtd->codec, codec, sizeof(*codec));
ret = q6apm_set_real_module_id(component->dev, prtd->graph, codec->id);
if (ret)
return ret;
cfg.direction = dir;
cfg.sample_rate = codec->sample_rate;
cfg.num_channels = 2;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = codec->id;
memcpy(&cfg.codec, codec, sizeof(*codec));
ret = q6apm_graph_media_format_shmem(prtd->graph, &cfg);
if (ret < 0)
return ret;
ret = q6apm_graph_media_format_pcm(prtd->graph, &cfg);
if (ret)
return ret;
ret = q6apm_map_memory_regions(prtd->graph, SNDRV_PCM_STREAM_PLAYBACK,
prtd->phys, (prtd->pcm_size / prtd->periods),
prtd->periods);
if (ret < 0)
return -ENOMEM;
ret = q6apm_graph_prepare(prtd->graph);
if (ret)
return ret;
ret = q6apm_graph_start(prtd->graph);
if (ret)
return ret;
} else {
cfg.direction = dir;
cfg.sample_rate = codec->sample_rate;
cfg.num_channels = 2;
cfg.bit_width = prtd->bits_per_sample;
cfg.fmt = codec->id;
memcpy(&cfg.codec, codec, sizeof(*codec));
ret = audioreach_compr_set_param(prtd->graph, &cfg);
if (ret < 0)
return ret;
}
prtd->state = Q6APM_STREAM_RUNNING;
return 0;
}
static int q6apm_dai_compr_set_metadata(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct snd_compr_metadata *metadata)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
int ret = 0;
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);
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);
break;
default:
ret = -EINVAL;
break;
}
return ret;
}
static int q6apm_dai_compr_mmap(struct snd_soc_component *component,
struct snd_compr_stream *stream,
struct vm_area_struct *vma)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
struct device *dev = component->dev;
return dma_mmap_coherent(dev, vma, prtd->dma_buffer.area, prtd->dma_buffer.addr,
prtd->dma_buffer.bytes);
}
static int q6apm_compr_copy(struct snd_soc_component *component,
struct snd_compr_stream *stream, char __user *buf,
size_t count)
{
struct snd_compr_runtime *runtime = stream->runtime;
struct q6apm_dai_rtd *prtd = runtime->private_data;
void *dstn;
unsigned long flags;
size_t copy;
u32 wflags = 0;
u32 app_pointer;
u32 bytes_received;
uint32_t bytes_to_write;
int avail, bytes_in_flight = 0;
bytes_received = prtd->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)
bytes_received = ALIGN(prtd->bytes_received, prtd->pcm_count);
app_pointer = bytes_received/prtd->pcm_size;
app_pointer = bytes_received - (app_pointer * prtd->pcm_size);
dstn = prtd->dma_buffer.area + app_pointer;
if (count < prtd->pcm_size - app_pointer) {
if (copy_from_user(dstn, buf, count))
return -EFAULT;
} else {
copy = prtd->pcm_size - app_pointer;
if (copy_from_user(dstn, buf, copy))
return -EFAULT;
if (copy_from_user(prtd->dma_buffer.area, buf + copy, count - copy))
return -EFAULT;
}
spin_lock_irqsave(&prtd->lock, flags);
bytes_in_flight = prtd->bytes_received - prtd->copied_total;
if (prtd->next_track) {
prtd->next_track = false;
prtd->copied_total = ALIGN(prtd->copied_total, prtd->pcm_count);
prtd->bytes_sent = ALIGN(prtd->bytes_sent, prtd->pcm_count);
}
prtd->bytes_received = bytes_received + count;
/* Kick off the data to dsp if its starving!! */
if (prtd->state == Q6APM_STREAM_RUNNING && (bytes_in_flight == 0)) {
bytes_to_write = prtd->pcm_count;
avail = prtd->bytes_received - prtd->bytes_sent;
if (avail < prtd->pcm_count)
bytes_to_write = avail;
q6apm_write_async(prtd->graph, bytes_to_write, 0, 0, wflags);
prtd->bytes_sent += bytes_to_write;
}
spin_unlock_irqrestore(&prtd->lock, flags);
return count;
}
static const struct snd_compress_ops q6apm_dai_compress_ops = {
.open = q6apm_dai_compr_open,
.free = q6apm_dai_compr_free,
.get_caps = q6apm_dai_compr_get_caps,
.get_codec_caps = q6apm_dai_compr_get_codec_caps,
.pointer = q6apm_dai_compr_pointer,
.trigger = q6apm_dai_compr_trigger,
.ack = q6apm_dai_compr_ack,
.set_params = q6apm_dai_compr_set_params,
.set_metadata = q6apm_dai_compr_set_metadata,
.mmap = q6apm_dai_compr_mmap,
.copy = q6apm_compr_copy,
};
static const struct snd_soc_component_driver q6apm_fe_dai_component = {
.name = DRV_NAME,
.open = q6apm_dai_open,
@ -395,6 +839,7 @@ static const struct snd_soc_component_driver q6apm_fe_dai_component = {
.hw_params = q6apm_dai_hw_params,
.pointer = q6apm_dai_pointer,
.trigger = q6apm_dai_trigger,
.compress_ops = &q6apm_dai_compress_ops,
};
static int q6apm_dai_probe(struct platform_device *pdev)

68
sound/soc/qcom/qdsp6/q6apm.c
View File

@ -298,6 +298,71 @@ int q6apm_unmap_memory_regions(struct q6apm_graph *graph, unsigned int dir)
}
EXPORT_SYMBOL_GPL(q6apm_unmap_memory_regions);
int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
{
struct audioreach_module *module;
module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
if (!module)
return -ENODEV;
return audioreach_send_u32_param(graph, module, PARAM_ID_REMOVE_INITIAL_SILENCE, samples);
}
EXPORT_SYMBOL_GPL(q6apm_remove_initial_silence);
int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples)
{
struct audioreach_module *module;
module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
if (!module)
return -ENODEV;
return audioreach_send_u32_param(graph, module, PARAM_ID_REMOVE_TRAILING_SILENCE, samples);
}
EXPORT_SYMBOL_GPL(q6apm_remove_trailing_silence);
int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en)
{
struct audioreach_module *module;
module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
if (!module)
return -ENODEV;
return audioreach_send_u32_param(graph, module, PARAM_ID_MODULE_ENABLE, en);
}
EXPORT_SYMBOL_GPL(q6apm_enable_compress_module);
int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph,
uint32_t codec_id)
{
struct audioreach_module *module;
uint32_t module_id;
module = q6apm_find_module_by_mid(graph, MODULE_ID_PLACEHOLDER_DECODER);
if (!module)
return -ENODEV;
switch (codec_id) {
case SND_AUDIOCODEC_MP3:
module_id = MODULE_ID_MP3_DECODE;
break;
case SND_AUDIOCODEC_AAC:
module_id = MODULE_ID_AAC_DEC;
break;
case SND_AUDIOCODEC_FLAC:
module_id = MODULE_ID_FLAC_DEC;
break;
default:
return -EINVAL;
}
return audioreach_send_u32_param(graph, module, PARAM_ID_REAL_MODULE_ID,
module_id);
}
EXPORT_SYMBOL_GPL(q6apm_set_real_module_id);
int q6apm_graph_media_format_pcm(struct q6apm_graph *graph, struct audioreach_module_config *cfg)
{
struct audioreach_graph_info *info = graph->info;
@ -497,6 +562,9 @@ static int graph_callback(struct gpr_resp_pkt *data, void *priv, int op)
}
break;
case DATA_CMD_WR_SH_MEM_EP_EOS_RENDERED:
client_event = APM_CLIENT_EVENT_CMD_EOS_DONE;
if (graph->cb)
graph->cb(client_event, hdr->token, data->payload, graph->priv);
break;
case GPR_BASIC_RSP_RESULT:
switch (result->opcode) {

6
sound/soc/qcom/qdsp6/q6apm.h
View File

@ -45,6 +45,8 @@
#define APM_WRITE_TOKEN_LEN_SHIFT 16
#define APM_MAX_SESSIONS 8
#define APM_LAST_BUFFER_FLAG BIT(30)
#define NO_TIMESTAMP 0xFF00
struct q6apm {
struct device *dev;
@ -147,4 +149,8 @@ int q6apm_graph_get_rx_shmem_module_iid(struct q6apm_graph *graph);
bool q6apm_is_adsp_ready(void);
int q6apm_enable_compress_module(struct device *dev, struct q6apm_graph *graph, bool en);
int q6apm_remove_initial_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
int q6apm_remove_trailing_silence(struct device *dev, struct q6apm_graph *graph, uint32_t samples);
int q6apm_set_real_module_id(struct device *dev, struct q6apm_graph *graph, uint32_t codec_id);
#endif /* __APM_GRAPH_ */

23
sound/soc/qcom/sc7280.c
View File

@ -14,6 +14,7 @@
#include <sound/soc.h>
#include <sound/rt5682s.h>
#include <linux/soundwire/sdw.h>
#include <sound/pcm_params.h>
#include "../codecs/rt5682.h"
#include "../codecs/rt5682s.h"
@ -196,8 +197,10 @@ static int sc7280_snd_hw_params(struct snd_pcm_substream *substream,
struct sdw_stream_runtime *sruntime;
int i;
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, 48000, 48000);
if (!rtd->dai_link->no_pcm) {
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_RATE, 48000, 48000);
}
switch (cpu_dai->id) {
case LPASS_CDC_DMA_TX3:
@ -358,6 +361,20 @@ static const struct snd_soc_dapm_widget sc7280_snd_widgets[] = {
SND_SOC_DAPM_MIC("Headset Mic", NULL),
};
static int sc7280_snd_be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
struct snd_pcm_hw_params *params)
{
struct snd_interval *rate = hw_param_interval(params, 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;
channels->min = channels->max = 2;
snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S16_LE);
return 0;
}
static int sc7280_snd_platform_probe(struct platform_device *pdev)
{
struct snd_soc_card *card;
@ -387,6 +404,8 @@ static int sc7280_snd_platform_probe(struct platform_device *pdev)
for_each_card_prelinks(card, i, link) {
link->init = sc7280_init;
link->ops = &sc7280_ops;
if (link->no_pcm == 1)
link->be_hw_params_fixup = sc7280_snd_be_hw_params_fixup;
}
return devm_snd_soc_register_card(dev, card);