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Diffstat (limited to 'sound/soc/qcom/qdsp6/q6asm.c')
-rw-r--r--sound/soc/qcom/qdsp6/q6asm.c435
1 files changed, 371 insertions, 64 deletions
diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c
index 36e0eab13a98..195780f75d05 100644
--- a/sound/soc/qcom/qdsp6/q6asm.c
+++ b/sound/soc/qcom/qdsp6/q6asm.c
@@ -25,6 +25,7 @@
#define ASM_STREAM_CMD_FLUSH 0x00010BCE
#define ASM_SESSION_CMD_PAUSE 0x00010BD3
#define ASM_DATA_CMD_EOS 0x00010BDB
+#define ASM_DATA_EVENT_RENDERED_EOS 0x00010C1C
#define ASM_NULL_POPP_TOPOLOGY 0x00010C68
#define ASM_STREAM_CMD_FLUSH_READBUFS 0x00010C09
#define ASM_STREAM_CMD_SET_ENCDEC_PARAM 0x00010C10
@@ -39,6 +40,8 @@
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
+#define ASM_MEDIA_FMT_WMA_V9 0x00010DA8
+#define ASM_MEDIA_FMT_WMA_V10 0x00010DA7
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@@ -46,6 +49,10 @@
#define ASM_STREAM_CMD_OPEN_READ_V3 0x00010DB4
#define ASM_DATA_EVENT_READ_DONE_V2 0x00010D9A
#define ASM_STREAM_CMD_OPEN_READWRITE_V2 0x00010D8D
+#define ASM_MEDIA_FMT_ALAC 0x00012f31
+#define ASM_MEDIA_FMT_APE 0x00012f32
+#define ASM_DATA_CMD_REMOVE_INITIAL_SILENCE 0x00010D67
+#define ASM_DATA_CMD_REMOVE_TRAILING_SILENCE 0x00010D68
#define ASM_LEGACY_STREAM_SESSION 0
@@ -104,6 +111,63 @@ struct asm_flac_fmt_blk_v2 {
u16 reserved;
} __packed;
+struct asm_wmastdv9_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 reserved;
+} __packed;
+
+struct asm_wmaprov10_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 fmtag;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 bytes_per_sec;
+ u16 blk_align;
+ u16 bits_per_sample;
+ u32 channel_mask;
+ u16 enc_options;
+ u16 advanced_enc_options1;
+ u32 advanced_enc_options2;
+} __packed;
+
+struct asm_alac_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u32 frame_length;
+ u8 compatible_version;
+ u8 bit_depth;
+ u8 pb;
+ u8 mb;
+ u8 kb;
+ u8 num_channels;
+ u16 max_run;
+ u32 max_frame_bytes;
+ u32 avg_bit_rate;
+ u32 sample_rate;
+ u32 channel_layout_tag;
+} __packed;
+
+struct asm_ape_fmt_blk_v2 {
+ struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+ u16 compatible_version;
+ u16 compression_level;
+ u32 format_flags;
+ u32 blocks_per_frame;
+ u32 final_frame_blocks;
+ u32 total_frames;
+ u16 bits_per_sample;
+ u16 num_channels;
+ u32 sample_rate;
+ u32 seek_table_present;
+} __packed;
+
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@@ -208,7 +272,6 @@ struct audio_client {
wait_queue_head_t cmd_wait;
struct aprv2_ibasic_rsp_result_t result;
int perf_mode;
- int stream_id;
struct q6asm *q6asm;
struct device *dev;
};
@@ -249,7 +312,7 @@ static int q6asm_apr_send_session_pkt(struct q6asm *a, struct audio_client *ac,
5 * HZ);
if (!rc) {
- dev_err(a->dev, "CMD timeout\n");
+ dev_err(a->dev, "CMD %x timeout\n", hdr->opcode);
rc = -ETIMEDOUT;
} else if (ac->result.status > 0) {
dev_err(a->dev, "DSP returned error[%x]\n",
@@ -428,7 +491,7 @@ static int __q6asm_memory_map_regions(struct audio_client *ac, int dir,
*
* @dir: direction of audio stream
* @ac: audio client instanace
- * @phys: physcial address that needs mapping.
+ * @phys: physical address that needs mapping.
* @period_sz: audio period size
* @periods: number of periods
*
@@ -450,7 +513,7 @@ int q6asm_map_memory_regions(unsigned int dir, struct audio_client *ac,
return 0;
}
- buf = kzalloc(((sizeof(struct audio_buffer)) * periods), GFP_ATOMIC);
+ buf = kcalloc(periods, sizeof(*buf), GFP_ATOMIC);
if (!buf) {
spin_unlock_irqrestore(&ac->lock, flags);
return -ENOMEM;
@@ -561,9 +624,6 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
case ASM_SESSION_CMD_SUSPEND:
client_event = ASM_CLIENT_EVENT_CMD_SUSPEND_DONE;
break;
- case ASM_DATA_CMD_EOS:
- client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
- break;
case ASM_STREAM_CMD_FLUSH:
client_event = ASM_CLIENT_EVENT_CMD_FLUSH_DONE;
break;
@@ -581,6 +641,8 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
case ASM_STREAM_CMD_OPEN_READWRITE_V2:
case ASM_STREAM_CMD_SET_ENCDEC_PARAM:
case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2:
+ case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
+ case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
if (result->status != 0) {
dev_err(ac->dev,
"cmd = 0x%x returned error = 0x%x\n",
@@ -612,6 +674,7 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
if (ac->io_mode & ASM_SYNC_IO_MODE) {
phys_addr_t phys;
unsigned long flags;
+ int token = hdr->token & ASM_WRITE_TOKEN_MASK;
spin_lock_irqsave(&ac->lock, flags);
@@ -623,12 +686,12 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
goto done;
}
- phys = port->buf[hdr->token].phys;
+ phys = port->buf[token].phys;
if (lower_32_bits(phys) != result->opcode ||
upper_32_bits(phys) != result->status) {
dev_err(ac->dev, "Expected addr %pa\n",
- &port->buf[hdr->token].phys);
+ &port->buf[token].phys);
spin_unlock_irqrestore(&ac->lock, flags);
ret = -EINVAL;
goto done;
@@ -667,6 +730,9 @@ static int32_t q6asm_stream_callback(struct apr_device *adev,
}
break;
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ client_event = ASM_CLIENT_EVENT_CMD_EOS_DONE;
+ break;
}
if (ac->cb)
@@ -766,21 +832,21 @@ EXPORT_SYMBOL_GPL(q6asm_get_session_id);
* @dev: Pointer to asm child device.
* @cb: event callback.
* @priv: private data associated with this client.
- * @stream_id: stream id
+ * @session_id: session id
* @perf_mode: performace mode for this client
*
* Return: Will be an error pointer on error or a valid audio client
* on success.
*/
struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
- void *priv, int stream_id,
+ void *priv, int session_id,
int perf_mode)
{
struct q6asm *a = dev_get_drvdata(dev->parent);
struct audio_client *ac;
unsigned long flags;
- ac = q6asm_get_audio_client(a, stream_id + 1);
+ ac = q6asm_get_audio_client(a, session_id + 1);
if (ac) {
dev_err(dev, "Audio Client already active\n");
return ac;
@@ -791,17 +857,15 @@ struct audio_client *q6asm_audio_client_alloc(struct device *dev, q6asm_cb cb,
return ERR_PTR(-ENOMEM);
spin_lock_irqsave(&a->slock, flags);
- a->session[stream_id + 1] = ac;
+ a->session[session_id + 1] = ac;
spin_unlock_irqrestore(&a->slock, flags);
- ac->session = stream_id + 1;
+ ac->session = session_id + 1;
ac->cb = cb;
ac->dev = dev;
ac->q6asm = a;
ac->priv = priv;
ac->io_mode = ASM_SYNC_IO_MODE;
ac->perf_mode = perf_mode;
- /* DSP expects stream id from 1 */
- ac->stream_id = 1;
ac->adev = a->adev;
kref_init(&ac->refcount);
@@ -829,7 +893,7 @@ static int q6asm_ac_send_cmd_sync(struct audio_client *ac, struct apr_pkt *pkt)
rc = wait_event_timeout(ac->cmd_wait,
(ac->result.opcode == hdr->opcode), 5 * HZ);
if (!rc) {
- dev_err(ac->dev, "CMD timeout\n");
+ dev_err(ac->dev, "CMD %x timeout\n", hdr->opcode);
rc = -ETIMEDOUT;
goto err;
}
@@ -850,15 +914,18 @@ err:
/**
* q6asm_open_write() - Open audio client for writing
- *
* @ac: audio client pointer
+ * @stream_id: stream id of q6asm session
* @format: audio sample format
+ * @codec_profile: compressed format profile
* @bits_per_sample: bits per sample
+ * @is_gapless: flag to indicate if this is a gapless stream
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_open_write(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+int q6asm_open_write(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, u32 codec_profile,
+ uint16_t bits_per_sample, bool is_gapless)
{
struct asm_stream_cmd_open_write_v3 *open;
struct apr_pkt *pkt;
@@ -873,11 +940,13 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
pkt = p;
open = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_WRITE_V3;
open->mode_flags = 0x00;
open->mode_flags |= ASM_LEGACY_STREAM_SESSION;
+ if (is_gapless)
+ open->mode_flags |= BIT(ASM_SHIFT_GAPLESS_MODE_FLAG);
/* source endpoint : matrix */
open->sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
@@ -894,6 +963,30 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
+ case SND_AUDIOCODEC_WMA:
+ switch (codec_profile) {
+ case SND_AUDIOPROFILE_WMA9:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V9;
+ break;
+ case SND_AUDIOPROFILE_WMA10:
+ case SND_AUDIOPROFILE_WMA9_PRO:
+ case SND_AUDIOPROFILE_WMA9_LOSSLESS:
+ case SND_AUDIOPROFILE_WMA10_LOSSLESS:
+ open->dec_fmt_id = ASM_MEDIA_FMT_WMA_V10;
+ break;
+ default:
+ dev_err(ac->dev, "Invalid codec profile 0x%x\n",
+ codec_profile);
+ rc = -EINVAL;
+ goto err;
+ }
+ break;
+ case SND_AUDIOCODEC_ALAC:
+ open->dec_fmt_id = ASM_MEDIA_FMT_ALAC;
+ break;
+ case SND_AUDIOCODEC_APE:
+ open->dec_fmt_id = ASM_MEDIA_FMT_APE;
+ break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@@ -912,8 +1005,9 @@ err:
}
EXPORT_SYMBOL_GPL(q6asm_open_write);
-static int __q6asm_run(struct audio_client *ac, uint32_t flags,
- uint32_t msw_ts, uint32_t lsw_ts, bool wait)
+static int __q6asm_run(struct audio_client *ac, uint32_t stream_id,
+ uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts,
+ bool wait)
{
struct asm_session_cmd_run_v2 *run;
struct apr_pkt *pkt;
@@ -928,7 +1022,7 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags,
pkt = p;
run = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_SESSION_CMD_RUN_V2;
run->flags = flags;
@@ -950,16 +1044,17 @@ static int __q6asm_run(struct audio_client *ac, uint32_t flags,
* q6asm_run() - start the audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id of q6asm session
* @flags: flags associated with write
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_run(struct audio_client *ac, uint32_t flags,
+int q6asm_run(struct audio_client *ac, uint32_t stream_id, uint32_t flags,
uint32_t msw_ts, uint32_t lsw_ts)
{
- return __q6asm_run(ac, flags, msw_ts, lsw_ts, true);
+ return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, true);
}
EXPORT_SYMBOL_GPL(q6asm_run);
@@ -967,16 +1062,17 @@ EXPORT_SYMBOL_GPL(q6asm_run);
* q6asm_run_nowait() - start the audio client withou blocking
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @flags: flags associated with write
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_run_nowait(struct audio_client *ac, uint32_t flags,
- uint32_t msw_ts, uint32_t lsw_ts)
+int q6asm_run_nowait(struct audio_client *ac, uint32_t stream_id,
+ uint32_t flags, uint32_t msw_ts, uint32_t lsw_ts)
{
- return __q6asm_run(ac, flags, msw_ts, lsw_ts, false);
+ return __q6asm_run(ac, stream_id, flags, msw_ts, lsw_ts, false);
}
EXPORT_SYMBOL_GPL(q6asm_run_nowait);
@@ -984,6 +1080,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait);
* q6asm_media_format_block_multi_ch_pcm() - setup pcm configuration
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @rate: audio sample rate
* @channels: number of audio channels.
* @channel_map: channel map pointer
@@ -992,6 +1089,7 @@ EXPORT_SYMBOL_GPL(q6asm_run_nowait);
* Return: Will be an negative value on error or zero on success
*/
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
+ uint32_t stream_id,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample)
@@ -1010,7 +1108,7 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1039,8 +1137,8 @@ err:
}
EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
-
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+ uint32_t stream_id,
struct q6asm_flac_cfg *cfg)
{
struct asm_flac_fmt_blk_v2 *fmt;
@@ -1056,7 +1154,7 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
pkt = p;
fmt = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
@@ -1075,10 +1173,220 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
+
+int q6asm_stream_media_format_block_wma_v9(struct audio_client *ac,
+ uint32_t stream_id,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmastdv9_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->reserved = 0;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v9);
+
+int q6asm_stream_media_format_block_wma_v10(struct audio_client *ac,
+ uint32_t stream_id,
+ struct q6asm_wma_cfg *cfg)
+{
+ struct asm_wmaprov10_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+ fmt->fmtag = cfg->fmtag;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->bytes_per_sec = cfg->bytes_per_sec;
+ fmt->blk_align = cfg->block_align;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->channel_mask = cfg->channel_mask;
+ fmt->enc_options = cfg->enc_options;
+ fmt->advanced_enc_options1 = cfg->adv_enc_options;
+ fmt->advanced_enc_options2 = cfg->adv_enc_options2;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_wma_v10);
+
+int q6asm_stream_media_format_block_alac(struct audio_client *ac,
+ uint32_t stream_id,
+ struct q6asm_alac_cfg *cfg)
+{
+ struct asm_alac_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->frame_length = cfg->frame_length;
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->bit_depth = cfg->bit_depth;
+ fmt->num_channels = cfg->num_channels;
+ fmt->max_run = cfg->max_run;
+ fmt->max_frame_bytes = cfg->max_frame_bytes;
+ fmt->avg_bit_rate = cfg->avg_bit_rate;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->channel_layout_tag = cfg->channel_layout_tag;
+ fmt->pb = cfg->pb;
+ fmt->mb = cfg->mb;
+ fmt->kb = cfg->kb;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_alac);
+
+int q6asm_stream_media_format_block_ape(struct audio_client *ac,
+ uint32_t stream_id,
+ struct q6asm_ape_cfg *cfg)
+{
+ struct asm_ape_fmt_blk_v2 *fmt;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(*fmt);
+ p = kzalloc(pkt_size, GFP_KERNEL);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ fmt = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+ pkt->hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt->fmt_blk.fmt_blk_size = sizeof(*fmt) - sizeof(fmt->fmt_blk);
+
+ fmt->compatible_version = cfg->compatible_version;
+ fmt->compression_level = cfg->compression_level;
+ fmt->format_flags = cfg->format_flags;
+ fmt->blocks_per_frame = cfg->blocks_per_frame;
+ fmt->final_frame_blocks = cfg->final_frame_blocks;
+ fmt->total_frames = cfg->total_frames;
+ fmt->bits_per_sample = cfg->bits_per_sample;
+ fmt->num_channels = cfg->num_channels;
+ fmt->sample_rate = cfg->sample_rate;
+ fmt->seek_table_present = cfg->seek_table_present;
+
+ rc = q6asm_ac_send_cmd_sync(ac, pkt);
+ kfree(pkt);
+
+ return rc;
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_ape);
+
+static int q6asm_stream_remove_silence(struct audio_client *ac, uint32_t stream_id,
+ uint32_t cmd,
+ uint32_t num_samples)
+{
+ uint32_t *samples;
+ struct apr_pkt *pkt;
+ void *p;
+ int rc, pkt_size;
+
+ pkt_size = APR_HDR_SIZE + sizeof(uint32_t);
+ p = kzalloc(pkt_size, GFP_ATOMIC);
+ if (!p)
+ return -ENOMEM;
+
+ pkt = p;
+ samples = p + APR_HDR_SIZE;
+
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
+
+ pkt->hdr.opcode = cmd;
+ *samples = num_samples;
+ rc = apr_send_pkt(ac->adev, pkt);
+ if (rc == pkt_size)
+ rc = 0;
+
+ kfree(pkt);
+
+ return rc;
+}
+
+int q6asm_stream_remove_initial_silence(struct audio_client *ac,
+ uint32_t stream_id,
+ uint32_t initial_samples)
+{
+ return q6asm_stream_remove_silence(ac, stream_id,
+ ASM_DATA_CMD_REMOVE_INITIAL_SILENCE,
+ initial_samples);
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_remove_initial_silence);
+
+int q6asm_stream_remove_trailing_silence(struct audio_client *ac, uint32_t stream_id,
+ uint32_t trailing_samples)
+{
+ return q6asm_stream_remove_silence(ac, stream_id,
+ ASM_DATA_CMD_REMOVE_TRAILING_SILENCE,
+ trailing_samples);
+}
+EXPORT_SYMBOL_GPL(q6asm_stream_remove_trailing_silence);
+
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @rate: audio sample rate
* @channels: number of audio channels.
* @bits_per_sample: bits per sample
@@ -1086,7 +1394,9 @@ EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
* Return: Will be an negative value on error or zero on success
*/
int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
- uint32_t rate, uint32_t channels, uint16_t bits_per_sample)
+ uint32_t stream_id, uint32_t rate,
+ uint32_t channels,
+ uint16_t bits_per_sample)
{
struct asm_multi_channel_pcm_enc_cfg_v2 *enc_cfg;
struct apr_pkt *pkt;
@@ -1102,7 +1412,7 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
pkt = p;
enc_cfg = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
enc_cfg->encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
@@ -1134,10 +1444,11 @@ EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
* q6asm_read() - read data of period size from audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_read(struct audio_client *ac)
+int q6asm_read(struct audio_client *ac, uint32_t stream_id)
{
struct asm_data_cmd_read_v2 *read;
struct audio_port_data *port;
@@ -1158,7 +1469,7 @@ int q6asm_read(struct audio_client *ac)
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_CAPTURE];
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf];
pkt->hdr.opcode = ASM_DATA_CMD_READ_V2;
read->buf_addr_lsw = lower_32_bits(ab->phys);
@@ -1186,7 +1497,7 @@ int q6asm_read(struct audio_client *ac)
}
EXPORT_SYMBOL_GPL(q6asm_read);
-static int __q6asm_open_read(struct audio_client *ac,
+static int __q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
uint32_t format, uint16_t bits_per_sample)
{
struct asm_stream_cmd_open_read_v3 *open;
@@ -1202,7 +1513,7 @@ static int __q6asm_open_read(struct audio_client *ac,
pkt = p;
open = p + APR_HDR_SIZE;
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, true, stream_id);
pkt->hdr.opcode = ASM_STREAM_CMD_OPEN_READ_V3;
/* Stream prio : High, provide meta info with encoded frames */
open->src_endpointype = ASM_END_POINT_DEVICE_MATRIX;
@@ -1233,15 +1544,16 @@ static int __q6asm_open_read(struct audio_client *ac,
* q6asm_open_read() - Open audio client for reading
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @format: audio sample format
* @bits_per_sample: bits per sample
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_open_read(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample)
+int q6asm_open_read(struct audio_client *ac, uint32_t stream_id,
+ uint32_t format, uint16_t bits_per_sample)
{
- return __q6asm_open_read(ac, format, bits_per_sample);
+ return __q6asm_open_read(ac, stream_id, format, bits_per_sample);
}
EXPORT_SYMBOL_GPL(q6asm_open_read);
@@ -1249,6 +1561,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read);
* q6asm_write_async() - non blocking write
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @len: length in bytes
* @msw_ts: timestamp msw
* @lsw_ts: timestamp lsw
@@ -1256,8 +1569,8 @@ EXPORT_SYMBOL_GPL(q6asm_open_read);
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
- uint32_t lsw_ts, uint32_t wflags)
+int q6asm_write_async(struct audio_client *ac, uint32_t stream_id, uint32_t len,
+ uint32_t msw_ts, uint32_t lsw_ts, uint32_t wflags)
{
struct asm_data_cmd_write_v2 *write;
struct audio_port_data *port;
@@ -1278,10 +1591,10 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
spin_lock_irqsave(&ac->lock, flags);
port = &ac->port[SNDRV_PCM_STREAM_PLAYBACK];
- q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, ac->stream_id);
+ q6asm_add_hdr(ac, &pkt->hdr, pkt_size, false, stream_id);
ab = &port->buf[port->dsp_buf];
- pkt->hdr.token = port->dsp_buf;
+ pkt->hdr.token = port->dsp_buf | (len << ASM_WRITE_TOKEN_LEN_SHIFT);
pkt->hdr.opcode = ASM_DATA_CMD_WRITE_V2;
write->buf_addr_lsw = lower_32_bits(ab->phys);
write->buf_addr_msw = upper_32_bits(ab->phys);
@@ -1292,10 +1605,7 @@ int q6asm_write_async(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
write->mem_map_handle =
ac->port[SNDRV_PCM_STREAM_PLAYBACK].mem_map_handle;
- if (wflags == NO_TIMESTAMP)
- write->flags = (wflags & 0x800000FF);
- else
- write->flags = (0x80000000 | wflags);
+ write->flags = wflags;
port->dsp_buf++;
@@ -1314,7 +1624,7 @@ EXPORT_SYMBOL_GPL(q6asm_write_async);
static void q6asm_reset_buf_state(struct audio_client *ac)
{
- struct audio_port_data *port = NULL;
+ struct audio_port_data *port;
unsigned long flags;
spin_lock_irqsave(&ac->lock, flags);
@@ -1325,9 +1635,9 @@ static void q6asm_reset_buf_state(struct audio_client *ac)
spin_unlock_irqrestore(&ac->lock, flags);
}
-static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
+static int __q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd,
+ bool wait)
{
- int stream_id = ac->stream_id;
struct apr_pkt pkt;
int rc;
@@ -1374,13 +1684,14 @@ static int __q6asm_cmd(struct audio_client *ac, int cmd, bool wait)
* q6asm_cmd() - run cmd on audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_cmd(struct audio_client *ac, int cmd)
+int q6asm_cmd(struct audio_client *ac, uint32_t stream_id, int cmd)
{
- return __q6asm_cmd(ac, cmd, true);
+ return __q6asm_cmd(ac, stream_id, cmd, true);
}
EXPORT_SYMBOL_GPL(q6asm_cmd);
@@ -1388,13 +1699,14 @@ EXPORT_SYMBOL_GPL(q6asm_cmd);
* q6asm_cmd_nowait() - non blocking, run cmd on audio client
*
* @ac: audio client pointer
+ * @stream_id: stream id
* @cmd: command to run on audio client.
*
* Return: Will be an negative value on error or zero on success
*/
-int q6asm_cmd_nowait(struct audio_client *ac, int cmd)
+int q6asm_cmd_nowait(struct audio_client *ac, uint32_t stream_id, int cmd)
{
- return __q6asm_cmd(ac, cmd, false);
+ return __q6asm_cmd(ac, stream_id, cmd, false);
}
EXPORT_SYMBOL_GPL(q6asm_cmd_nowait);
@@ -1415,24 +1727,19 @@ static int q6asm_probe(struct apr_device *adev)
spin_lock_init(&q6asm->slock);
dev_set_drvdata(dev, q6asm);
- return of_platform_populate(dev->of_node, NULL, NULL, dev);
+ return devm_of_platform_populate(dev);
}
-static int q6asm_remove(struct apr_device *adev)
-{
- of_platform_depopulate(&adev->dev);
-
- return 0;
-}
+#ifdef CONFIG_OF
static const struct of_device_id q6asm_device_id[] = {
{ .compatible = "qcom,q6asm" },
{},
};
MODULE_DEVICE_TABLE(of, q6asm_device_id);
+#endif
static struct apr_driver qcom_q6asm_driver = {
.probe = q6asm_probe,
- .remove = q6asm_remove,
.callback = q6asm_srvc_callback,
.driver = {
.name = "qcom-q6asm",
Copyright © 1996 – 2023 Jason A. Donenfeld. All Rights Reverse Engineered.