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ASoC: qcom: q6asm: add support to flac config

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Qualcomm DSPs expect flac config to be set for flac decoders, so add the
API to program the flac config to the DSP

Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@linaro.org>
Signed-off-by: Vinod Koul <vkoul@kernel.org>
Link: https://lore.kernel.org/r/20191115102705.649976-3-vkoul@kernel.org
Signed-off-by: Mark Brown <broonie@kernel.org>
This commit is contained in:
Srinivas Kandagatla 2019-11-15 15:57:04 +05:30 committed by Mark Brown
parent d2522335c9
commit 51d2584a98
No known key found for this signature in database
GPG Key ID: 24D68B725D5487D0
2 changed files with 70 additions and 0 deletions
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55
sound/soc/qcom/qdsp6/q6asm.c
View File

@ -38,6 +38,7 @@
#define ASM_SESSION_CMD_RUN_V2 0x00010DAA
#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 0x00010DA5
#define ASM_MEDIA_FMT_MP3 0x00010BE9
#define ASM_MEDIA_FMT_FLAC 0x00010C16
#define ASM_DATA_CMD_WRITE_V2 0x00010DAB
#define ASM_DATA_CMD_READ_V2 0x00010DAC
#define ASM_SESSION_CMD_SUSPEND 0x00010DEC
@ -89,6 +90,20 @@ struct asm_multi_channel_pcm_fmt_blk_v2 {
u8 channel_mapping[PCM_MAX_NUM_CHANNEL];
} __packed;
struct asm_flac_fmt_blk_v2 {
struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
u16 is_stream_info_present;
u16 num_channels;
u16 min_blk_size;
u16 max_blk_size;
u16 md5_sum[8];
u32 sample_rate;
u32 min_frame_size;
u32 max_frame_size;
u16 sample_size;
u16 reserved;
} __packed;
struct asm_stream_cmd_set_encdec_param {
u32 param_id;
u32 param_size;
@ -876,6 +891,9 @@ int q6asm_open_write(struct audio_client *ac, uint32_t format,
case FORMAT_LINEAR_PCM:
open->dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
break;
case SND_AUDIOCODEC_FLAC:
open->dec_fmt_id = ASM_MEDIA_FMT_FLAC;
break;
default:
dev_err(ac->dev, "Invalid format 0x%x\n", format);
rc = -EINVAL;
@ -1021,6 +1039,42 @@ err:
}
EXPORT_SYMBOL_GPL(q6asm_media_format_block_multi_ch_pcm);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct q6asm_flac_cfg *cfg)
{
struct asm_flac_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, ac->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->is_stream_info_present = cfg->stream_info_present;
fmt->num_channels = cfg->ch_cfg;
fmt->min_blk_size = cfg->min_blk_size;
fmt->max_blk_size = cfg->max_blk_size;
fmt->sample_rate = cfg->sample_rate;
fmt->min_frame_size = cfg->min_frame_size;
fmt->max_frame_size = cfg->max_frame_size;
fmt->sample_size = cfg->sample_size;
rc = q6asm_ac_send_cmd_sync(ac, pkt);
kfree(pkt);
return rc;
}
EXPORT_SYMBOL_GPL(q6asm_stream_media_format_block_flac);
/**
* q6asm_enc_cfg_blk_pcm_format_support() - setup pcm configuration for capture
*
@ -1075,6 +1129,7 @@ err:
}
EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support);
/**
* q6asm_read() - read data of period size from audio client
*

15
sound/soc/qcom/qdsp6/q6asm.h
View File

@ -32,6 +32,19 @@ enum {
#define NO_TIMESTAMP 0xFF00
#define FORMAT_LINEAR_PCM 0x0000
struct q6asm_flac_cfg {
u32 sample_rate;
u32 ext_sample_rate;
u32 min_frame_size;
u32 max_frame_size;
u16 stream_info_present;
u16 min_blk_size;
u16 max_blk_size;
u16 ch_cfg;
u16 sample_size;
u16 md5_sum;
};
typedef void (*q6asm_cb) (uint32_t opcode, uint32_t token,
void *payload, void *priv);
struct audio_client;
@ -54,6 +67,8 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
uint32_t rate, uint32_t channels,
u8 channel_map[PCM_MAX_NUM_CHANNEL],
uint16_t bits_per_sample);
int q6asm_stream_media_format_block_flac(struct audio_client *ac,
struct q6asm_flac_cfg *cfg);
int q6asm_run(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 flags, uint32_t msw_ts,