The patch ASoC: qcom: q6asm-dai: add support to flac decoder 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 69efab0fc638704e100c3dfb29eac971a4c1cc29 Mon Sep 17 00:00:00 2001 From: Vinod Koul <vkoul@xxxxxxxxxx> Date: Fri, 15 Nov 2019 15:57:05 +0530 Subject: [PATCH] ASoC: qcom: q6asm-dai: add support to flac decoder Qualcomm DSPs also support the flac decoder, so add support for FLAC decoder and convert the snd_dec_flac params to qdsp format. Co-developed-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx> Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla@xxxxxxxxxx> Signed-off-by: Vinod Koul <vkoul@xxxxxxxxxx> Link: https://lore.kernel.org/r/20191115102705.649976-4-vkoul@xxxxxxxxxx Signed-off-by: Mark Brown <broonie@xxxxxxxxxx> --- sound/soc/qcom/qdsp6/q6asm-dai.c | 35 +++++++++++++++++++++++++++++++- 1 file changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index f59353f510b8..8150c10f081e 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -626,8 +626,14 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, struct snd_soc_component *c = snd_soc_rtdcom_lookup(rtd, DRV_NAME); int dir = stream->direction; struct q6asm_dai_data *pdata; + struct q6asm_flac_cfg flac_cfg; struct device *dev = c->dev; int ret; + union snd_codec_options *codec_options; + struct snd_dec_flac *flac; + + codec_options = &(prtd->codec_param.codec.options); + memcpy(&prtd->codec_param, params, sizeof(*params)); @@ -664,6 +670,32 @@ static int q6asm_dai_compr_set_params(struct snd_compr_stream *stream, return ret; } + switch (params->codec.id) { + case SND_AUDIOCODEC_FLAC: + + memset(&flac_cfg, 0x0, sizeof(struct q6asm_flac_cfg)); + flac = &codec_options->flac_d; + + flac_cfg.ch_cfg = params->codec.ch_in; + flac_cfg.sample_rate = params->codec.sample_rate; + flac_cfg.stream_info_present = 1; + flac_cfg.sample_size = flac->sample_size; + flac_cfg.min_blk_size = flac->min_blk_size; + flac_cfg.max_blk_size = flac->max_blk_size; + flac_cfg.max_frame_size = flac->max_frame_size; + flac_cfg.min_frame_size = flac->min_frame_size; + + ret = q6asm_stream_media_format_block_flac(prtd->audio_client, + &flac_cfg); + if (ret < 0) { + dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); + return -EIO; + } + break; + default: + break; + } + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, (prtd->pcm_size / prtd->periods), prtd->periods); @@ -759,8 +791,9 @@ static int q6asm_dai_compr_get_caps(struct snd_compr_stream *stream, caps->max_fragment_size = COMPR_PLAYBACK_MAX_FRAGMENT_SIZE; caps->min_fragments = COMPR_PLAYBACK_MIN_NUM_FRAGMENTS; caps->max_fragments = COMPR_PLAYBACK_MAX_NUM_FRAGMENTS; - caps->num_codecs = 1; + caps->num_codecs = 2; caps->codecs[0] = SND_AUDIOCODEC_MP3; + caps->codecs[1] = SND_AUDIOCODEC_FLAC; return 0; } -- 2.20.1 _______________________________________________ Alsa-devel mailing list Alsa-devel@xxxxxxxxxxxxxxxx https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
