The patch ASoC: qcom: q6asm: add support to flac config has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.5 All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From badfe2c0666b1b65ad443aca74540bf6d976ec83 Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx> Date: Fri, 15 Nov 2019 15:57:04 +0530 Subject: [PATCH] ASoC: qcom: q6asm: add support to flac config 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@xxxxxxxxxx> Signed-off-by: Vinod Koul <vkoul@xxxxxxxxxx> Link: https://lore.kernel.org/r/20191115102705.649976-3-vkoul@xxxxxxxxxx Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- sound/soc/qcom/qdsp6/q6asm.c | 55 ++++++++++++++++++++++++++++++++++++ sound/soc/qcom/qdsp6/q6asm.h | 15 ++++++++++ 2 files changed, 70 insertions(+) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index e8141a33a55e..36e0eab13a98 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -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 @@ int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac, } 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 @@ int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac, } EXPORT_SYMBOL_GPL(q6asm_enc_cfg_blk_pcm_format_support); + /** * q6asm_read() - read data of period size from audio client * diff --git a/sound/soc/qcom/qdsp6/q6asm.h b/sound/soc/qcom/qdsp6/q6asm.h index 9f5fb573e4a0..6764f55f7078 100644 --- a/sound/soc/qcom/qdsp6/q6asm.h +++ b/sound/soc/qcom/qdsp6/q6asm.h @@ -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, -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
