asoc: msm-compress: Add support for up to 32 channels
Compressed streams now do support up to 32 channels.
Use new ADSP API for 32 channels if ADSP supports this.
If not fall back to 8 channel API.
Change-Id: Ief6fe189a6a34f15cfb96bfb9cb7d97b03633b95
Signed-off-by: Dieter Luecking <dieterl@codeaurora.org>
Signed-off-by: Mangesh Kunchamwar <mangeshk@codeaurora.org>
diff --git a/asoc/msm-compress-q6-v2.c b/asoc/msm-compress-q6-v2.c
index 5dc1ad2..cf2f264 100644
--- a/asoc/msm-compress-q6-v2.c
+++ b/asoc/msm-compress-q6-v2.c
@@ -41,6 +41,7 @@
#include <dsp/msm_audio_ion.h>
#include <dsp/apr_audio-v2.h>
#include <dsp/q6asm-v2.h>
+#include <dsp/q6core.h>
#include <dsp/msm-audio-effects-q6-v2.h>
#include "msm-pcm-routing-v2.h"
#include "msm-qti-pp-config.h"
@@ -204,7 +205,7 @@
struct msm_compr_ch_map {
bool set_ch_map;
- char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL];
+ char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL_V8];
};
static int msm_compr_send_dec_params(struct snd_compr_stream *cstream,
@@ -1013,16 +1014,32 @@
sample_word_size = 16;
break;
}
- ret = q6asm_media_format_block_pcm_format_support_v4(
- prtd->audio_client,
- prtd->sample_rate,
- prtd->num_channels,
- bit_width, stream_id,
- use_default_chmap,
- chmap,
- sample_word_size,
- ASM_LITTLE_ENDIAN,
- DEFAULT_QF);
+
+ if (q6core_get_avcs_api_version_per_service(
+ APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=
+ ADSP_ASM_API_VERSION_V2) {
+ ret = q6asm_media_format_block_pcm_format_support_v5(
+ prtd->audio_client,
+ prtd->sample_rate,
+ prtd->num_channels,
+ bit_width, stream_id,
+ use_default_chmap,
+ chmap,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ } else {
+ ret = q6asm_media_format_block_pcm_format_support_v4(
+ prtd->audio_client,
+ prtd->sample_rate,
+ prtd->num_channels,
+ bit_width, stream_id,
+ use_default_chmap,
+ chmap,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ }
if (ret < 0)
pr_err("%s: CMD Format block failed\n", __func__);
@@ -1337,7 +1354,16 @@
} else {
pr_debug("%s: stream_id %d bits_per_sample %d\n",
__func__, ac->stream_id, bits_per_sample);
- ret = q6asm_stream_open_write_v4(ac,
+
+ if (q6core_get_avcs_api_version_per_service(
+ APRV2_IDS_SERVICE_ID_ADSP_ASM_V) >=
+ ADSP_ASM_API_VERSION_V2)
+ ret = q6asm_stream_open_write_v5(ac,
+ prtd->codec, bits_per_sample,
+ ac->stream_id,
+ prtd->gapless_state.use_dsp_gapless_mode);
+ else
+ ret = q6asm_stream_open_write_v4(ac,
prtd->codec, bits_per_sample,
ac->stream_id,
prtd->gapless_state.use_dsp_gapless_mode);
@@ -3595,7 +3621,7 @@
if (pdata->ch_map[fe_id]) {
pdata->ch_map[fe_id]->set_ch_map = true;
- for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL_V8; i++)
pdata->ch_map[fe_id]->channel_map[i] =
(char)(ucontrol->value.integer.value[i]);
} else {
@@ -3624,7 +3650,7 @@
goto end;
}
if (pdata->ch_map[fe_id]) {
- for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL_V8; i++)
ucontrol->value.integer.value[i] =
pdata->ch_map[fe_id]->channel_map[i];
}
@@ -3958,9 +3984,10 @@
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
- uinfo->count = 8;
+ uinfo->count = PCM_FORMAT_MAX_NUM_CHANNEL_V8;
uinfo->value.integer.min = 0;
- uinfo->value.integer.max = 0xFFFFFFFF;
+ /* See PCM_CHANNEL_RSD=34 in apr_audio-v2.h */
+ uinfo->value.integer.max = 34;
return 0;
}