diff --git a/asoc/msm-compress-q6-v2.c b/asoc/msm-compress-q6-v2.c
index c4fecdefa2b3..c0d8614559f1 100644
--- a/asoc/msm-compress-q6-v2.c
+++ b/asoc/msm-compress-q6-v2.c
@@ -1432,6 +1432,7 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
int dir = OUT, ret = 0;
struct audio_client *ac = prtd->audio_client;
uint32_t stream_index;
+ uint32_t enc_cfg_id = ENC_CFG_ID_NONE;
switch (prtd->codec_param.codec.format) {
case SNDRV_PCM_FORMAT_S24_LE:
@@ -1450,6 +1451,9 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
default:
bits_per_sample = 16;
sample_word_size = 16;
+ if (prtd->codec == FORMAT_BESPOKE)
+ enc_cfg_id =
+ prtd->codec_param.codec.options.generic.reserved[0];
break;
}
@@ -1457,11 +1461,11 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
__func__, ac->stream_id, bits_per_sample);
if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) {
- ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
- bits_per_sample, true);
+ ret = q6asm_open_read_v4(prtd->audio_client, prtd->codec,
+ bits_per_sample, true, enc_cfg_id);
} else {
- ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
- bits_per_sample, false);
+ ret = q6asm_open_read_v4(prtd->audio_client, prtd->codec,
+ bits_per_sample, false, enc_cfg_id);
}
if (ret < 0) {
pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret);
@@ -1521,10 +1525,20 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",
__func__, prtd->sample_rate, prtd->num_channels,
bits_per_sample, sample_word_size);
- ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,
+ if (prtd->codec == FORMAT_BESPOKE) {
+ /*
+ * For BESPOKE codec, encoder specific config params are
+ * included as part of generic.
+ */
+ ret = q6asm_enc_cfg_blk_custom(prtd->audio_client, prtd->sample_rate,
+ prtd->num_channels, prtd->codec,
+ (void *)&prtd->codec_param.codec.options.generic);
+ } else {
+ ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,
prtd->sample_rate, prtd->num_channels,
bits_per_sample, sample_word_size,
ASM_LITTLE_ENDIAN, DEFAULT_QF);
+ }
return ret;
}
@@ -2043,6 +2057,12 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
break;
}
+ case SND_AUDIOCODEC_BESPOKE: {
+ pr_debug("%s: SND_AUDIOCODEC_BESPOKE\n", __func__);
+ prtd->codec = FORMAT_BESPOKE;
+ break;
+ }
+
default:
pr_err("codec not supported, id =%d\n", params->codec.id);
return -EINVAL;
diff --git a/asoc/msm-pcm-q6-v2.c b/asoc/msm-pcm-q6-v2.c
index b82128e7fabe..8ae85b743686 100644
--- a/asoc/msm-pcm-q6-v2.c
+++ b/asoc/msm-pcm-q6-v2.c
@@ -490,7 +490,7 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
prtd->audio_client->perf_mode);
ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
- bits_per_sample, false);
+ bits_per_sample, false, ENC_CFG_ID_NONE);
if (ret < 0) {
pr_err("%s: q6asm_open_read failed\n", __func__);
q6asm_audio_client_free(prtd->audio_client);
diff --git a/dsp/q6asm.c b/dsp/q6asm.c
index a12347966f5a..38ddc88b0f24 100644
--- a/dsp/q6asm.c
+++ b/dsp/q6asm.c
@@ -46,6 +46,8 @@
#define FALSE 0x00
#define SESSION_MAX 8
+#define ENC_FRAMES_PER_BUFFER 0x01
+
enum {
ASM_TOPOLOGY_CAL = 0,
ASM_CUSTOM_TOP_CAL,
@@ -2812,7 +2814,7 @@ EXPORT_SYMBOL(q6asm_set_soft_volume_module_instance_ids);
static int __q6asm_open_read(struct audio_client *ac,
uint32_t format, uint16_t bits_per_sample,
uint32_t pcm_format_block_ver,
- bool ts_mode)
+ bool ts_mode, uint32_t enc_cfg_id)
{
int rc = 0x00;
struct asm_stream_cmd_open_read_v3 open;
@@ -2888,6 +2890,12 @@ static int __q6asm_open_read(struct audio_client *ac,
open.mode_flags |= BUFFER_META_ENABLE;
open.enc_cfg_id = ASM_MEDIA_FMT_AMRWB_FS;
break;
+ case FORMAT_BESPOKE:
+ open.mode_flags |= BUFFER_META_ENABLE;
+ open.enc_cfg_id = enc_cfg_id;
+ if (ts_mode)
+ open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE;
+ break;
default:
pr_err("%s: Invalid format 0x%x\n",
__func__, format);
@@ -2939,7 +2947,7 @@ int q6asm_open_read(struct audio_client *ac,
{
return __q6asm_open_read(ac, format, 16,
PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
- false/*ts_mode*/);
+ false/*ts_mode*/, ENC_CFG_ID_NONE);
}
EXPORT_SYMBOL(q6asm_open_read);
@@ -2948,7 +2956,7 @@ int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
- false/*ts_mode*/);
+ false/*ts_mode*/, ENC_CFG_ID_NONE);
}
/*
@@ -2963,7 +2971,7 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V3/*media fmt block ver*/,
- false/*ts_mode*/);
+ false/*ts_mode*/, ENC_CFG_ID_NONE);
}
EXPORT_SYMBOL(q6asm_open_read_v3);
@@ -2976,11 +2984,12 @@ EXPORT_SYMBOL(q6asm_open_read_v3);
* @ts_mode: timestamp mode
*/
int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample, bool ts_mode)
+ uint16_t bits_per_sample, bool ts_mode,
+ uint32_t enc_cfg_id)
{
return __q6asm_open_read(ac, format, bits_per_sample,
PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/,
- ts_mode);
+ ts_mode, enc_cfg_id);
}
EXPORT_SYMBOL(q6asm_open_read_v4);
@@ -4287,6 +4296,95 @@ int q6asm_stream_run_nowait(struct audio_client *ac, uint32_t flags,
return __q6asm_run_nowait(ac, flags, msw_ts, lsw_ts, stream_id);
}
+/**
+ * q6asm_enc_cfg_blk_custom -
+ * command to set encode cfg block for custom
+ *
+ * @ac: Audio client handle
+ * @sample_rate: Sample rate
+ * @channels: number of ASM channels
+ * @format: custom format flag
+ * @cfg: generic encoder config
+ *
+ * Returns 0 on success or error on failure
+ */
+int q6asm_enc_cfg_blk_custom(struct audio_client *ac,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t format, void *cfg)
+{
+ struct asm_custom_enc_cfg_t_v2 enc_cfg;
+ int rc = 0;
+ uint32_t custom_size;
+ struct snd_enc_generic *enc_generic = (struct snd_enc_generic *) cfg;
+
+ custom_size = enc_generic->reserved[1];
+
+ pr_debug("%s: session[%d] size[%d] res[2]=[%d] res[3]=[%d]\n",
+ __func__, ac->session, custom_size, enc_generic->reserved[2],
+ enc_generic->reserved[3]);
+
+ pr_debug("%s: res[4]=[%d] sr[%d] ch[%d] format[%d]\n",
+ __func__, enc_generic->reserved[4], sample_rate,
+ channels, format);
+
+ memset(&enc_cfg, 0, sizeof(struct asm_custom_enc_cfg_t_v2));
+ q6asm_add_hdr(ac, &enc_cfg.hdr, sizeof(enc_cfg), TRUE);
+ atomic_set(&ac->cmd_state, -1);
+
+ enc_cfg.hdr.opcode = ASM_STREAM_CMD_SET_ENCDEC_PARAM;
+ enc_cfg.encdec.param_id = ASM_PARAM_ID_ENCDEC_ENC_CFG_BLK_V2;
+ enc_cfg.encdec.param_size = sizeof(struct asm_custom_enc_cfg_t_v2) -
+ sizeof(struct asm_stream_cmd_set_encdec_param);
+ enc_cfg.encblk.frames_per_buf = ENC_FRAMES_PER_BUFFER;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(struct asm_enc_cfg_blk_param_v2);
+
+ enc_cfg.num_channels = channels;
+ enc_cfg.sample_rate = sample_rate;
+
+ if (q6asm_map_channels(enc_cfg.channel_mapping, channels, false)) {
+ pr_err("%s: map channels failed %d\n",
+ __func__, channels);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ if (format == FORMAT_BESPOKE && custom_size &&
+ custom_size <= sizeof(enc_cfg.custom_data)) {
+ memcpy(enc_cfg.custom_data, &enc_generic->reserved[2],
+ custom_size);
+ enc_cfg.custom_size = custom_size;
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("%s: Comamnd %d failed %d\n",
+ __func__, ASM_STREAM_CMD_SET_ENCDEC_PARAM, rc);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) >= 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout. waited for FORMAT_UPDATE\n",
+ __func__);
+ rc = -ETIMEDOUT;
+ goto fail_cmd;
+ }
+ if (atomic_read(&ac->cmd_state) > 0) {
+ pr_err("%s: DSP returned error[%s]\n",
+ __func__, adsp_err_get_err_str(
+ atomic_read(&ac->cmd_state)));
+ rc = adsp_err_get_lnx_err_code(
+ atomic_read(&ac->cmd_state));
+ goto fail_cmd;
+ }
+ return 0;
+fail_cmd:
+ return rc;
+}
+EXPORT_SYMBOL(q6asm_enc_cfg_blk_custom);
+
/**
* q6asm_enc_cfg_blk_aac -
* command to set encode cfg block for aac
diff --git a/include/dsp/apr_audio-v2.h b/include/dsp/apr_audio-v2.h
index a282f78b33d6..5f62a2711d6e 100644
--- a/include/dsp/apr_audio-v2.h
+++ b/include/dsp/apr_audio-v2.h
@@ -4743,6 +4743,22 @@ struct asm_enc_cfg_blk_param_v2 {
} __packed;
+struct asm_custom_enc_cfg_t_v2 {
+ struct apr_hdr hdr;
+ struct asm_stream_cmd_set_encdec_param encdec;
+ struct asm_enc_cfg_blk_param_v2 encblk;
+ uint32_t sample_rate;
+
+ uint16_t num_channels;
+ uint16_t reserved;
+ /* num_ch == 1, then PCM_CHANNEL_C,
+ * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
+ */
+ uint8_t channel_mapping[8];
+ uint32_t custom_size;
+ uint8_t custom_data[15];
+} __packed;
+
/* @brief Dolby Digital Plus end point configuration structure
*/
struct asm_dec_ddp_endp_param_v2 {
diff --git a/include/dsp/q6asm-v2.h b/include/dsp/q6asm-v2.h
index ba35779a5788..e6f021db7e88 100644
--- a/include/dsp/q6asm-v2.h
+++ b/include/dsp/q6asm-v2.h
@@ -57,11 +57,14 @@
#define FORMAT_GEN_COMPR 0x001f
#define FORMAT_TRUEHD 0x0020
#define FORMAT_IEC61937 0x0021
+#define FORMAT_BESPOKE 0x0022
#define ENCDEC_SBCBITRATE 0x0001
#define ENCDEC_IMMEDIATE_DECODE 0x0002
#define ENCDEC_CFG_BLK 0x0003
+#define ENC_CFG_ID_NONE 0x0000
+
#define CMD_PAUSE 0x0001
#define CMD_FLUSH 0x0002
#define CMD_EOS 0x0003
@@ -290,7 +293,8 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
- uint16_t bits_per_sample, bool ts_mode);
+ uint16_t bits_per_sample, bool ts_mode,
+ uint32_t enc_cfg_id);
int q6asm_open_write(struct audio_client *ac, uint32_t format
/*, uint16_t bits_per_sample*/);
@@ -454,6 +458,10 @@ int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac,
uint16_t endianness,
uint16_t mode);
+int q6asm_enc_cfg_blk_custom(struct audio_client *ac,
+ uint32_t sample_rate, uint32_t channels,
+ uint32_t format, void *cfg);
+
int q6asm_set_encdec_chan_map(struct audio_client *ac,
uint32_t num_channels);