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[86.30.250.44]) by smtp.gmail.com with ESMTPSA id y189sm20566099wmd.27.2020.07.27.02.38.22 (version=TLS1_3 cipher=TLS_AES_256_GCM_SHA384 bits=256/256); Mon, 27 Jul 2020 02:38:23 -0700 (PDT) From: Srinivas Kandagatla To: broonie@kernel.org Subject: [PATCH v3 08/10] ASoC: q6asm-dai: prepare set params to accept profile change Date: Mon, 27 Jul 2020 10:38:04 +0100 Message-Id: <20200727093806.17089-9-srinivas.kandagatla@linaro.org> X-Mailer: git-send-email 2.21.0 In-Reply-To: <20200727093806.17089-1-srinivas.kandagatla@linaro.org> References: <20200727093806.17089-1-srinivas.kandagatla@linaro.org> MIME-Version: 1.0 Content-Transfer-Encoding: 8bit Cc: pierre-louis.bossart@linux.intel.com, alsa-devel@alsa-project.org, ckeepax@opensource.cirrus.com, tiwai@suse.com, lgirdwood@gmail.com, linux-kernel@vger.kernel.org, vkoul@kernel.org, Srinivas Kandagatla X-BeenThere: alsa-devel@alsa-project.org X-Mailman-Version: 2.1.15 Precedence: list List-Id: "Alsa-devel mailing list for ALSA developers - http://www.alsa-project.org" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Errors-To: alsa-devel-bounces@alsa-project.org Sender: "Alsa-devel" rearrange code so that it will be easy to change the codec profile at runtime. This means moving exiting set_params to an internal wrapper which can be called when codec profile changes. This is also preparing the code for easy to use in gapless cases. Signed-off-by: Srinivas Kandagatla --- sound/soc/qcom/qdsp6/q6asm-dai.c | 149 ++++++++++++++++++------------- 1 file changed, 85 insertions(+), 64 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6asm-dai.c b/sound/soc/qcom/qdsp6/q6asm-dai.c index a7fe4147c0a2..420aaaa67788 100644 --- a/sound/soc/qcom/qdsp6/q6asm-dai.c +++ b/sound/soc/qcom/qdsp6/q6asm-dai.c @@ -53,7 +53,7 @@ enum stream_state { struct q6asm_dai_rtd { struct snd_pcm_substream *substream; struct snd_compr_stream *cstream; - struct snd_compr_params codec_param; + struct snd_codec codec; struct snd_dma_buffer dma_buffer; spinlock_t lock; phys_addr_t phys; @@ -644,15 +644,13 @@ static int q6asm_dai_compr_free(struct snd_soc_component *component, return 0; } -static int q6asm_dai_compr_set_params(struct snd_soc_component *component, - struct snd_compr_stream *stream, - struct snd_compr_params *params) +static int __q6asm_dai_compr_set_codec_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_codec *codec, + int stream_id) { struct snd_compr_runtime *runtime = stream->runtime; struct q6asm_dai_rtd *prtd = runtime->private_data; - struct snd_soc_pcm_runtime *rtd = stream->private_data; - int dir = stream->direction; - struct q6asm_dai_data *pdata; struct q6asm_flac_cfg flac_cfg; struct q6asm_wma_cfg wma_cfg; struct q6asm_alac_cfg alac_cfg; @@ -666,53 +664,18 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, struct snd_dec_alac *alac; struct snd_dec_ape *ape; - codec_options = &(prtd->codec_param.codec.options); - - - memcpy(&prtd->codec_param, params, sizeof(*params)); - - pdata = snd_soc_component_get_drvdata(component); - if (!pdata) - return -EINVAL; - - if (!prtd || !prtd->audio_client) { - dev_err(dev, "private data null or audio client freed\n"); - return -EINVAL; - } - - prtd->periods = runtime->fragments; - prtd->pcm_count = runtime->fragment_size; - prtd->pcm_size = runtime->fragments * runtime->fragment_size; - prtd->bits_per_sample = 16; - if (dir == SND_COMPRESS_PLAYBACK) { - ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, - params->codec.id, params->codec.profile, - prtd->bits_per_sample, true); - - if (ret < 0) { - dev_err(dev, "q6asm_open_write failed\n"); - q6asm_audio_client_free(prtd->audio_client); - prtd->audio_client = NULL; - return ret; - } - } + codec_options = &(prtd->codec.options); - prtd->session_id = q6asm_get_session_id(prtd->audio_client); - ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, - prtd->session_id, dir); - if (ret) { - dev_err(dev, "Stream reg failed ret:%d\n", ret); - return ret; - } + memcpy(&prtd->codec, codec, sizeof(*codec)); - switch (params->codec.id) { + switch (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.ch_cfg = codec->ch_in; + flac_cfg.sample_rate = 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; @@ -721,7 +684,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, flac_cfg.min_frame_size = flac->min_frame_size; ret = q6asm_stream_media_format_block_flac(prtd->audio_client, - prtd->stream_id, + stream_id, &flac_cfg); if (ret < 0) { dev_err(dev, "FLAC CMD Format block failed:%d\n", ret); @@ -734,10 +697,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&wma_cfg, 0x0, sizeof(struct q6asm_wma_cfg)); - wma_cfg.sample_rate = params->codec.sample_rate; - wma_cfg.num_channels = params->codec.ch_in; - wma_cfg.bytes_per_sec = params->codec.bit_rate / 8; - wma_cfg.block_align = params->codec.align; + wma_cfg.sample_rate = codec->sample_rate; + wma_cfg.num_channels = codec->ch_in; + wma_cfg.bytes_per_sec = codec->bit_rate / 8; + wma_cfg.block_align = codec->align; wma_cfg.bits_per_sample = prtd->bits_per_sample; wma_cfg.enc_options = wma->encoder_option; wma_cfg.adv_enc_options = wma->adv_encoder_option; @@ -751,7 +714,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, return -EINVAL; /* check the codec profile */ - switch (params->codec.profile) { + switch (codec->profile) { case SND_AUDIOPROFILE_WMA9: wma_cfg.fmtag = 0x161; wma_v9 = 1; @@ -775,17 +738,17 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, default: dev_err(dev, "Unknown WMA profile:%x\n", - params->codec.profile); + codec->profile); return -EIO; } if (wma_v9) ret = q6asm_stream_media_format_block_wma_v9( - prtd->audio_client, prtd->stream_id, + prtd->audio_client, stream_id, &wma_cfg); else ret = q6asm_stream_media_format_block_wma_v10( - prtd->audio_client, prtd->stream_id, + prtd->audio_client, stream_id, &wma_cfg); if (ret < 0) { dev_err(dev, "WMA9 CMD failed:%d\n", ret); @@ -797,10 +760,10 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&alac_cfg, 0x0, sizeof(alac_cfg)); alac = &codec_options->alac_d; - alac_cfg.sample_rate = params->codec.sample_rate; - alac_cfg.avg_bit_rate = params->codec.bit_rate; + alac_cfg.sample_rate = codec->sample_rate; + alac_cfg.avg_bit_rate = codec->bit_rate; alac_cfg.bit_depth = prtd->bits_per_sample; - alac_cfg.num_channels = params->codec.ch_in; + alac_cfg.num_channels = codec->ch_in; alac_cfg.frame_length = alac->frame_length; alac_cfg.pb = alac->pb; @@ -810,7 +773,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, alac_cfg.compatible_version = alac->compatible_version; alac_cfg.max_frame_bytes = alac->max_frame_bytes; - switch (params->codec.ch_in) { + switch (codec->ch_in) { case 1: alac_cfg.channel_layout_tag = ALAC_CH_LAYOUT_MONO; break; @@ -819,7 +782,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, break; } ret = q6asm_stream_media_format_block_alac(prtd->audio_client, - prtd->stream_id, + stream_id, &alac_cfg); if (ret < 0) { dev_err(dev, "ALAC CMD Format block failed:%d\n", ret); @@ -831,8 +794,8 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, memset(&ape_cfg, 0x0, sizeof(ape_cfg)); ape = &codec_options->ape_d; - ape_cfg.sample_rate = params->codec.sample_rate; - ape_cfg.num_channels = params->codec.ch_in; + ape_cfg.sample_rate = codec->sample_rate; + ape_cfg.num_channels = codec->ch_in; ape_cfg.bits_per_sample = prtd->bits_per_sample; ape_cfg.compatible_version = ape->compatible_version; @@ -844,7 +807,7 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, ape_cfg.seek_table_present = ape->seek_table_present; ret = q6asm_stream_media_format_block_ape(prtd->audio_client, - prtd->stream_id, + stream_id, &ape_cfg); if (ret < 0) { dev_err(dev, "APE CMD Format block failed:%d\n", ret); @@ -856,6 +819,64 @@ static int q6asm_dai_compr_set_params(struct snd_soc_component *component, break; } + return 0; +} + +static int q6asm_dai_compr_set_params(struct snd_soc_component *component, + struct snd_compr_stream *stream, + struct snd_compr_params *params) +{ + struct snd_compr_runtime *runtime = stream->runtime; + struct q6asm_dai_rtd *prtd = runtime->private_data; + struct snd_soc_pcm_runtime *rtd = stream->private_data; + int dir = stream->direction; + struct q6asm_dai_data *pdata; + struct device *dev = component->dev; + int ret; + + pdata = snd_soc_component_get_drvdata(component); + if (!pdata) + return -EINVAL; + + if (!prtd || !prtd->audio_client) { + dev_err(dev, "private data null or audio client freed\n"); + return -EINVAL; + } + + prtd->periods = runtime->fragments; + prtd->pcm_count = runtime->fragment_size; + prtd->pcm_size = runtime->fragments * runtime->fragment_size; + prtd->bits_per_sample = 16; + + if (dir == SND_COMPRESS_PLAYBACK) { + ret = q6asm_open_write(prtd->audio_client, prtd->stream_id, params->codec.id, + params->codec.profile, prtd->bits_per_sample, + true); + + if (ret < 0) { + dev_err(dev, "q6asm_open_write failed\n"); + q6asm_audio_client_free(prtd->audio_client); + prtd->audio_client = NULL; + return ret; + } + } + + prtd->session_id = q6asm_get_session_id(prtd->audio_client); + ret = q6routing_stream_open(rtd->dai_link->id, LEGACY_PCM_MODE, + prtd->session_id, dir); + if (ret) { + dev_err(dev, "Stream reg failed ret:%d\n", ret); + return ret; + } + + ret = __q6asm_dai_compr_set_codec_params(component, stream, + ¶ms->codec, + prtd->stream_id); + if (ret) { + dev_err(dev, "codec param setup failed ret:%d\n", ret); + return ret; + } + ret = q6asm_map_memory_regions(dir, prtd->audio_client, prtd->phys, (prtd->pcm_size / prtd->periods), prtd->periods); -- 2.21.0