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-rw-r--r--include/sound/q6asm-v2.h23
-rw-r--r--sound/soc/msm/msm-dai-fe.c78
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c58
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h2
-rw-r--r--sound/soc/msm/qdsp6v2/q6asm.c376
5 files changed, 485 insertions, 52 deletions
diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h
index 285d32e249b8..9df3e77da05b 100644
--- a/include/sound/q6asm-v2.h
+++ b/include/sound/q6asm-v2.h
@@ -114,6 +114,7 @@ enum {
PCM_MEDIA_FORMAT_V2 = 0,
PCM_MEDIA_FORMAT_V3,
PCM_MEDIA_FORMAT_V4,
+ PCM_MEDIA_FORMAT_V5,
};
/* PCM format modes in DSP */
@@ -288,6 +289,9 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, bool ts_mode);
+int q6asm_open_read_v5(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample, bool ts_mode);
+
int q6asm_open_write(struct audio_client *ac, uint32_t format
/*, uint16_t bits_per_sample*/);
@@ -303,6 +307,8 @@ int q6asm_open_write_v3(struct audio_client *ac, uint32_t format,
int q6asm_open_write_v4(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample);
+int q6asm_open_write_v5(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample);
int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode);
@@ -450,6 +456,13 @@ int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac,
uint16_t endianness,
uint16_t mode);
+int q6asm_enc_cfg_blk_pcm_format_support_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode);
+
int q6asm_set_encdec_chan_map(struct audio_client *ac,
uint32_t num_channels);
@@ -545,6 +558,15 @@ int q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac,
uint16_t endianness,
uint16_t mode);
+int q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ bool use_default_chmap,
+ char *channel_map,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode);
+
int q6asm_media_format_block_aac(struct audio_client *ac,
struct asm_aac_cfg *cfg);
@@ -704,4 +726,5 @@ uint8_t q6asm_get_stream_id_from_token(uint32_t token);
int q6asm_adjust_session_clock(struct audio_client *ac,
uint32_t adjust_time_lsw,
uint32_t adjust_time_msw);
+int q6asm_get_svc_version(uint32_t service_id);
#endif /* __Q6_ASM_H__ */
diff --git a/sound/soc/msm/msm-dai-fe.c b/sound/soc/msm/msm-dai-fe.c
index 3326c993e129..c9178fb3b8a3 100644
--- a/sound/soc/msm/msm-dai-fe.c
+++ b/sound/soc/msm/msm-dai-fe.c
@@ -99,7 +99,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -113,7 +113,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -132,7 +132,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -146,7 +146,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -217,7 +217,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 6,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -231,7 +231,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -250,7 +250,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -270,7 +270,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -284,7 +284,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -303,7 +303,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -317,7 +317,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -336,7 +336,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -356,7 +356,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -370,7 +370,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -2155,7 +2155,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2166,7 +2166,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_RATE_KNOT),
.formats = SNDRV_PCM_FMTBIT_S16_LE,
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -2372,7 +2372,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2392,7 +2392,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2412,7 +2412,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2432,7 +2432,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2452,7 +2452,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2472,7 +2472,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2492,7 +2492,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2506,7 +2506,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -2574,7 +2574,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
@@ -2593,7 +2593,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
@@ -2612,7 +2612,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
@@ -2632,7 +2632,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2646,7 +2646,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -2665,7 +2665,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2679,7 +2679,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 48000,
},
@@ -2698,7 +2698,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2717,7 +2717,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2736,7 +2736,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2755,7 +2755,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2774,7 +2774,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_3LE |
SNDRV_PCM_FMTBIT_S32_LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 384000,
},
@@ -2793,7 +2793,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
@@ -2812,7 +2812,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
@@ -2831,7 +2831,7 @@ static struct snd_soc_dai_driver msm_fe_dais[] = {
SNDRV_PCM_FMTBIT_S24_LE |
SNDRV_PCM_FMTBIT_S24_3LE),
.channels_min = 1,
- .channels_max = 8,
+ .channels_max = 32,
.rate_min = 8000,
.rate_max = 192000,
},
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
index d54c357247f8..a0364bbdfeb9 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.c
@@ -35,6 +35,7 @@
#include <linux/of_device.h>
#include <sound/tlv.h>
#include <sound/pcm_params.h>
+#include <sound/q6core.h>
#include "msm-pcm-q6-v2.h"
#include "msm-pcm-routing-v2.h"
@@ -384,8 +385,13 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
return -ENOMEM;
}
} else {
- ret = q6asm_open_write_v4(prtd->audio_client,
- fmt_type, bits_per_sample);
+ if (q6asm_get_svc_version(APR_SVC_ASM) >=
+ ADSP_ASM_API_VERSION_V2)
+ ret = q6asm_open_write_v5(prtd->audio_client,
+ fmt_type, bits_per_sample);
+ else
+ ret = q6asm_open_write_v4(prtd->audio_client,
+ fmt_type, bits_per_sample);
if (ret < 0) {
pr_err("%s: q6asm_open_write_v4 failed (%d)\n",
@@ -425,7 +431,16 @@ static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
runtime->channels, !prtd->set_channel_map,
prtd->channel_map, bits_per_sample);
} else {
- ret = q6asm_media_format_block_multi_ch_pcm_v4(
+ if (q6asm_get_svc_version(APR_SVC_ASM) >=
+ ADSP_ASM_API_VERSION_V2)
+ ret = q6asm_media_format_block_multi_ch_pcm_v5(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, !prtd->set_channel_map,
+ prtd->channel_map, bits_per_sample,
+ sample_word_size, ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ else
+ ret = q6asm_media_format_block_multi_ch_pcm_v4(
prtd->audio_client, runtime->rate,
runtime->channels, !prtd->set_channel_map,
prtd->channel_map, bits_per_sample,
@@ -489,8 +504,15 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
__func__, params_channels(params),
prtd->audio_client->perf_mode);
- ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
- bits_per_sample, false);
+ if (q6asm_get_svc_version(APR_SVC_ASM) >=
+ ADSP_ASM_API_VERSION_V2)
+ ret = q6asm_open_read_v5(prtd->audio_client,
+ FORMAT_LINEAR_PCM,
+ bits_per_sample, false);
+ else
+ ret = q6asm_open_read_v4(prtd->audio_client,
+ FORMAT_LINEAR_PCM,
+ bits_per_sample, false);
if (ret < 0) {
pr_err("%s: q6asm_open_read failed\n", __func__);
q6asm_audio_client_free(prtd->audio_client);
@@ -557,13 +579,25 @@ static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
pr_debug("%s: Samp_rate = %d Channel = %d bit width = %d, word size = %d\n",
__func__, prtd->samp_rate, prtd->channel_mode,
bits_per_sample, sample_word_size);
- ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,
- prtd->samp_rate,
- prtd->channel_mode,
- bits_per_sample,
- sample_word_size,
- ASM_LITTLE_ENDIAN,
- DEFAULT_QF);
+ if (q6asm_get_svc_version(APR_SVC_ASM) >=
+ ADSP_ASM_API_VERSION_V2)
+ ret = q6asm_enc_cfg_blk_pcm_format_support_v5(
+ prtd->audio_client,
+ prtd->samp_rate,
+ prtd->channel_mode,
+ bits_per_sample,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ else
+ ret = q6asm_enc_cfg_blk_pcm_format_support_v4(
+ prtd->audio_client,
+ prtd->samp_rate,
+ prtd->channel_mode,
+ bits_per_sample,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
if (ret < 0)
pr_debug("%s: cmd cfg pcm was block failed", __func__);
diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
index f9c0c03492e1..afe314455fbf 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-q6-v2.h
@@ -104,7 +104,7 @@ struct msm_audio {
int mmap_flag;
atomic_t pending_buffer;
bool set_channel_map;
- char channel_map[8];
+ char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL_V2];
int cmd_interrupt;
bool meta_data_mode;
uint32_t volume;
diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c
index 9e14b341448a..e9592cf1eb5a 100644
--- a/sound/soc/msm/qdsp6v2/q6asm.c
+++ b/sound/soc/msm/qdsp6v2/q6asm.c
@@ -195,6 +195,39 @@ static int is_adsp_raise_event(uint32_t cmd)
return -EINVAL;
}
+int q6asm_get_svc_version(uint32_t service_id)
+{
+ int ret = 0;
+ static int asm_cached_version;
+ size_t ver_size;
+ struct avcs_fwk_ver_info *ver_info = NULL;
+
+ if (service_id == AVCS_SERVICE_ID_ALL) {
+ pr_err("%s: Invalid service id: %d", __func__,
+ AVCS_SERVICE_ID_ALL);
+ return -EINVAL;
+ }
+
+ if (asm_cached_version != 0)
+ return asm_cached_version;
+
+ ver_size = sizeof(struct avcs_get_fwk_version) +
+ sizeof(struct avs_svc_api_info);
+ ver_info = kzalloc(ver_size, GFP_KERNEL);
+ if (ver_info == NULL)
+ return -ENOMEM;
+
+ ret = q6core_get_service_version(service_id, ver_info, ver_size);
+ if (ret < 0)
+ goto done;
+
+ ret = ver_info->services[0].api_version;
+ asm_cached_version = ret;
+done:
+ kfree(ver_info);
+ return ret;
+}
+
static inline void q6asm_set_flag_in_token(union asm_token_struct *asm_token,
int flag, int flag_offset)
{
@@ -228,6 +261,9 @@ static inline uint32_t q6asm_get_pcm_format_id(uint32_t media_format_block_ver)
uint32_t pcm_format_id;
switch (media_format_block_ver) {
+ case PCM_MEDIA_FORMAT_V5:
+ pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V5;
+ break;
case PCM_MEDIA_FORMAT_V4:
pcm_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4;
break;
@@ -2869,6 +2905,23 @@ int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
}
EXPORT_SYMBOL(q6asm_open_read_v4);
+/*
+ * asm_open_read_v5 - Opens audio capture session
+ *
+ * @ac: Client session handle
+ * @format: encoder format
+ * @bits_per_sample: bit width of capture session
+ * @ts_mode: timestamp mode
+ */
+int q6asm_open_read_v5(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample, bool ts_mode)
+{
+ return __q6asm_open_read(ac, format, bits_per_sample,
+ PCM_MEDIA_FORMAT_V5 /*media fmt block ver*/,
+ ts_mode);
+}
+EXPORT_SYMBOL(q6asm_open_read_v5);
+
int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format,
uint32_t passthrough_flag)
{
@@ -3172,6 +3225,22 @@ int q6asm_open_write_v4(struct audio_client *ac, uint32_t format,
}
EXPORT_SYMBOL(q6asm_open_write_v4);
+/*
+ * q6asm_open_write_v5 - Opens audio playback session
+ *
+ * @ac: Client session handle
+ * @format: decoder format
+ * @bits_per_sample: bit width of playback session
+ */
+int q6asm_open_write_v5(struct audio_client *ac, uint32_t format,
+ uint16_t bits_per_sample)
+{
+ return __q6asm_open_write(ac, format, bits_per_sample,
+ ac->stream_id, false /*gapless*/,
+ PCM_MEDIA_FORMAT_V5 /*pcm_format_block_ver*/);
+}
+EXPORT_SYMBOL(q6asm_open_write_v5);
+
int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
uint16_t bits_per_sample, int32_t stream_id,
bool is_gapless_mode)
@@ -4239,6 +4308,108 @@ fail_cmd:
}
/*
+ * q6asm_enc_cfg_blk_pcm_v5 - sends encoder configuration parameters
+ *
+ * @ac: Client session handle
+ * @rate: sample rate
+ * @channels: number of channels
+ * @bits_per_sample: bit width of encoder session
+ * @use_default_chmap: true if default channel map to be used
+ * @use_back_flavor: to configure back left and right channel
+ * @channel_map: input channel map
+ * @sample_word_size: Size in bits of the word that holds a sample of a channel
+ * @endianness: endianness of the pcm data
+ * @mode: Mode to provide additional info about the pcm input data
+ */
+static int q6asm_enc_cfg_blk_pcm_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ uint16_t bits_per_sample, bool use_default_chmap,
+ bool use_back_flavor, u8 *channel_map,
+ uint16_t sample_word_size, uint16_t endianness,
+ uint16_t mode)
+{
+ struct asm_multi_channel_pcm_enc_cfg_v5 enc_cfg;
+ struct asm_enc_cfg_blk_param_v2 enc_fg_blk;
+ u8 *channel_mapping;
+ u32 frames_per_buf = 0;
+ int rc;
+
+ if (!use_default_chmap && (channel_map == NULL)) {
+ pr_err("%s: No valid chan map and can't use default\n",
+ __func__);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+
+ pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
+ ac->session, rate, channels,
+ bits_per_sample, sample_word_size);
+
+ memset(&enc_cfg, 0, sizeof(enc_cfg));
+ 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(enc_cfg) - sizeof(enc_cfg.hdr) -
+ sizeof(enc_cfg.encdec);
+ enc_cfg.encblk.frames_per_buf = frames_per_buf;
+ enc_cfg.encblk.enc_cfg_blk_size = enc_cfg.encdec.param_size -
+ sizeof(enc_fg_blk);
+ enc_cfg.num_channels = channels;
+ enc_cfg.bits_per_sample = bits_per_sample;
+ enc_cfg.sample_rate = rate;
+ enc_cfg.is_signed = 1;
+ enc_cfg.sample_word_size = sample_word_size;
+ enc_cfg.endianness = endianness;
+ enc_cfg.mode = mode;
+ channel_mapping = enc_cfg.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL_V2);
+
+ if (use_default_chmap) {
+ pr_debug("%s: setting default channel map for %d channels",
+ __func__, channels);
+ if (q6asm_map_channels(channel_mapping, channels,
+ use_back_flavor)) {
+ pr_err("%s: map channels failed %d\n",
+ __func__, channels);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ } else {
+ pr_debug("%s: Using pre-defined channel map", __func__);
+ memcpy(channel_mapping, channel_map,
+ PCM_FORMAT_MAX_NUM_CHANNEL_V2);
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &enc_cfg);
+ if (rc < 0) {
+ pr_err("%s: Command open failed %d\n", __func__, rc);
+ goto fail_cmd;
+ }
+ rc = wait_event_timeout(ac->cmd_wait,
+ (atomic_read(&ac->cmd_state) >= 0), 5*HZ);
+ if (!rc) {
+ pr_err("%s: timeout opcode[0x%x]\n",
+ __func__, enc_cfg.hdr.opcode);
+ 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_pcm_v5);
+
+/*
* q6asm_enc_cfg_blk_pcm_v4 - sends encoder configuration parameters
*
* @ac: Client session handle
@@ -4516,6 +4687,18 @@ fail_cmd:
return rc;
}
+static int __q6asm_enc_cfg_blk_pcm_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode)
+{
+ return q6asm_enc_cfg_blk_pcm_v5(ac, rate, channels,
+ bits_per_sample, true, false, NULL,
+ sample_word_size, endianness, mode);
+}
+
static int __q6asm_enc_cfg_blk_pcm_v4(struct audio_client *ac,
uint32_t rate, uint32_t channels,
uint16_t bits_per_sample,
@@ -4602,6 +4785,31 @@ int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac,
}
EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_format_support_v4);
+/*
+ * q6asm_enc_cfg_blk_pcm_format_support_v5 - sends encoder configuration
+ * parameters
+ *
+ * @ac: Client session handle
+ * @rate: sample rate
+ * @channels: number of channels
+ * @bits_per_sample: bit width of encoder session
+ * @sample_word_size: Size in bits of the word that holds a sample of a channel
+ * @endianness: endianness of the pcm data
+ * @mode: Mode to provide additional info about the pcm input data
+ */
+int q6asm_enc_cfg_blk_pcm_format_support_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode)
+{
+ return __q6asm_enc_cfg_blk_pcm_v5(ac, rate, channels,
+ bits_per_sample, sample_word_size,
+ endianness, mode);
+}
+EXPORT_SYMBOL(q6asm_enc_cfg_blk_pcm_format_support_v5);
+
int q6asm_enc_cfg_blk_pcm_native(struct audio_client *ac,
uint32_t rate, uint32_t channels)
{
@@ -4727,6 +4935,67 @@ static int q6asm_map_channels(u8 *channel_mapping, uint32_t channels,
lchannel_mapping[5] = PCM_CHANNEL_RB;
lchannel_mapping[6] = PCM_CHANNEL_LS;
lchannel_mapping[7] = PCM_CHANNEL_RS;
+ } else if (channels == 10) {
+ lchannel_mapping[0] = PCM_CHANNEL_FL;
+ lchannel_mapping[1] = PCM_CHANNEL_FR;
+ lchannel_mapping[2] = PCM_CHANNEL_LFE;
+ lchannel_mapping[3] = PCM_CHANNEL_FC;
+ lchannel_mapping[4] = PCM_CHANNEL_LS;
+ lchannel_mapping[5] = PCM_CHANNEL_RS;
+ lchannel_mapping[6] = PCM_CHANNEL_LB;
+ lchannel_mapping[7] = PCM_CHANNEL_RB;
+ lchannel_mapping[8] = PCM_CHANNEL_CS;
+ lchannel_mapping[9] = PCM_CHANNELS;
+ } else if (channels == 16) {
+ lchannel_mapping[0] = PCM_CHANNEL_FL;
+ lchannel_mapping[1] = PCM_CHANNEL_FR;
+ lchannel_mapping[2] = PCM_CHANNEL_LFE;
+ lchannel_mapping[3] = PCM_CHANNEL_FC;
+ lchannel_mapping[4] = PCM_CHANNEL_LS;
+ lchannel_mapping[5] = PCM_CHANNEL_RS;
+ lchannel_mapping[6] = PCM_CHANNEL_LB;
+ lchannel_mapping[7] = PCM_CHANNEL_RB;
+ lchannel_mapping[8] = PCM_CHANNEL_CS;
+ lchannel_mapping[9] = PCM_CHANNELS;
+ lchannel_mapping[10] = PCM_CHANNEL_CVH;
+ lchannel_mapping[11] = PCM_CHANNEL_MS;
+ lchannel_mapping[12] = PCM_CHANNEL_FLC;
+ lchannel_mapping[13] = PCM_CHANNEL_FRC;
+ lchannel_mapping[14] = PCM_CHANNEL_RLC;
+ lchannel_mapping[15] = PCM_CHANNEL_RRC;
+ } else if (channels == 32) {
+ lchannel_mapping[0] = PCM_CHANNEL_FL;
+ lchannel_mapping[1] = PCM_CHANNEL_FR;
+ lchannel_mapping[2] = PCM_CHANNEL_LFE;
+ lchannel_mapping[3] = PCM_CHANNEL_FC;
+ lchannel_mapping[4] = PCM_CHANNEL_LS;
+ lchannel_mapping[5] = PCM_CHANNEL_RS;
+ lchannel_mapping[6] = PCM_CHANNEL_LB;
+ lchannel_mapping[7] = PCM_CHANNEL_RB;
+ lchannel_mapping[8] = PCM_CHANNEL_CS;
+ lchannel_mapping[9] = PCM_CHANNELS;
+ lchannel_mapping[10] = PCM_CHANNEL_CVH;
+ lchannel_mapping[11] = PCM_CHANNEL_MS;
+ lchannel_mapping[12] = PCM_CHANNEL_FLC;
+ lchannel_mapping[13] = PCM_CHANNEL_FRC;
+ lchannel_mapping[14] = PCM_CHANNEL_RLC;
+ lchannel_mapping[15] = PCM_CHANNEL_RRC;
+ lchannel_mapping[16] = PCM_CHANNEL_LFE2;
+ lchannel_mapping[17] = PCM_CHANNEL_SL;
+ lchannel_mapping[18] = PCM_CHANNEL_SR;
+ lchannel_mapping[19] = PCM_CHANNEL_TFL;
+ lchannel_mapping[20] = PCM_CHANNEL_TFR;
+ lchannel_mapping[21] = PCM_CHANNEL_TC;
+ lchannel_mapping[22] = PCM_CHANNEL_TBL;
+ lchannel_mapping[23] = PCM_CHANNEL_TBR;
+ lchannel_mapping[24] = PCM_CHANNEL_TSL;
+ lchannel_mapping[25] = PCM_CHANNEL_TSR;
+ lchannel_mapping[26] = PCM_CHANNEL_TBC;
+ lchannel_mapping[27] = PCM_CHANNEL_BFC;
+ lchannel_mapping[28] = PCM_CHANNEL_BFL;
+ lchannel_mapping[29] = PCM_CHANNEL_BFR;
+ lchannel_mapping[30] = PCM_CHANNEL_LW;
+ lchannel_mapping[31] = PCM_CHANNEL_RW;
} else {
pr_err("%s: ERROR.unsupported num_ch = %u\n",
__func__, channels);
@@ -5647,6 +5916,79 @@ fail_cmd:
return rc;
}
+static int __q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac,
+ uint32_t rate,
+ uint32_t channels,
+ bool use_default_chmap,
+ char *channel_map,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode)
+{
+ struct asm_multi_channel_pcm_fmt_blk_param_v5 fmt;
+ u8 *channel_mapping;
+ int rc;
+
+ pr_debug("%s: session[%d]rate[%d]ch[%d]bps[%d]wordsize[%d]\n", __func__,
+ ac->session, rate, channels,
+ bits_per_sample, sample_word_size);
+
+ memset(&fmt, 0, sizeof(fmt));
+ q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+ atomic_set(&ac->cmd_state, -1);
+
+ fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+ fmt.fmt_blk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+ sizeof(fmt.fmt_blk);
+ fmt.param.num_channels = channels;
+ fmt.param.bits_per_sample = bits_per_sample;
+ fmt.param.sample_rate = rate;
+ fmt.param.is_signed = 1;
+ fmt.param.sample_word_size = sample_word_size;
+ fmt.param.endianness = endianness;
+ fmt.param.mode = mode;
+ channel_mapping = fmt.param.channel_mapping;
+
+ memset(channel_mapping, 0, PCM_FORMAT_MAX_NUM_CHANNEL_V2);
+
+ if (use_default_chmap) {
+ if (q6asm_map_channels(channel_mapping, channels, false)) {
+ pr_err("%s: map channels failed %d\n",
+ __func__, channels);
+ rc = -EINVAL;
+ goto fail_cmd;
+ }
+ } else {
+ memcpy(channel_mapping, channel_map,
+ PCM_FORMAT_MAX_NUM_CHANNEL_V2);
+ }
+
+ rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+ if (rc < 0) {
+ pr_err("%s: Comamnd open failed %d\n", __func__, rc);
+ 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;
+}
+
int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
bool use_default_chmap, char *channel_map)
@@ -5727,6 +6069,40 @@ int q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac,
EXPORT_SYMBOL(q6asm_media_format_block_multi_ch_pcm_v4);
/*
+ * q6asm_media_format_block_multi_ch_pcm_v5 - sends pcm decoder configuration
+ * parameters
+ *
+ * @ac: Client session handle
+ * @rate: sample rate
+ * @channels: number of channels
+ * @bits_per_sample: bit width of encoder session
+ * @use_default_chmap: true if default channel map to be used
+ * @channel_map: input channel map
+ * @sample_word_size: Size in bits of the word that holds a sample of a channel
+ * @endianness: endianness of the pcm data
+ * @mode: Mode to provide additional info about the pcm input data
+ */
+int q6asm_media_format_block_multi_ch_pcm_v5(struct audio_client *ac,
+ uint32_t rate, uint32_t channels,
+ bool use_default_chmap,
+ char *channel_map,
+ uint16_t bits_per_sample,
+ uint16_t sample_word_size,
+ uint16_t endianness,
+ uint16_t mode)
+{
+ return __q6asm_media_format_block_multi_ch_pcm_v5(ac, rate, channels,
+ use_default_chmap,
+ channel_map,
+ bits_per_sample,
+ sample_word_size,
+ endianness,
+ mode);
+}
+EXPORT_SYMBOL(q6asm_media_format_block_multi_ch_pcm_v5);
+
+
+/*
* q6asm_media_format_block_gen_compr - set up generic compress format params
*
* @ac: Client session handle
generated by cgit v1.2.3 (git 2.25.1) at 2025-12-25 03:38:30 +0000