From: Vinod Koul <vkoul@kernel.org> To: Takashi Iwai <tiwai@suse.com> Cc: linux-arm-msm@vger.kernel.org, Bjorn Andersson <bjorn.andersson@linaro.org>, Srinivas Kandagatla <srinivas.kandagatla@linaro.org>, Jaroslav Kysela <perex@perex.cz>, Patrick Lai <plai@codeaurora.org>, Banajit Goswami <bgoswami@codeaurora.org>, Liam Girdwood <lgirdwood@gmail.com>, Mark Brown <broonie@kernel.org>, Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>, alsa-devel@alsa-project.org, linux-kernel@vger.kernel.org, Vinod Koul <vkoul@kernel.org> Subject: [RFC PATCH 2/3] ASoC: qcom: q6asm: add support to flac config Date: Fri, 15 Nov 2019 15:57:04 +0530 [thread overview] Message-ID: <20191115102705.649976-3-vkoul@kernel.org> (raw) In-Reply-To: <20191115102705.649976-1-vkoul@kernel.org> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> 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> --- 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, -- 2.23.0
WARNING: multiple messages have this Message-ID (diff)
From: Vinod Koul <vkoul@kernel.org> To: Takashi Iwai <tiwai@suse.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>, alsa-devel@alsa-project.org, Banajit Goswami <bgoswami@codeaurora.org>, linux-arm-msm@vger.kernel.org, Patrick Lai <plai@codeaurora.org>, Liam Girdwood <lgirdwood@gmail.com>, Vinod Koul <vkoul@kernel.org>, Mark Brown <broonie@kernel.org>, Srinivas Kandagatla <srinivas.kandagatla@linaro.org>, Bjorn Andersson <bjorn.andersson@linaro.org>, linux-kernel@vger.kernel.org Subject: [alsa-devel] [RFC PATCH 2/3] ASoC: qcom: q6asm: add support to flac config Date: Fri, 15 Nov 2019 15:57:04 +0530 [thread overview] Message-ID: <20191115102705.649976-3-vkoul@kernel.org> (raw) In-Reply-To: <20191115102705.649976-1-vkoul@kernel.org> From: Srinivas Kandagatla <srinivas.kandagatla@linaro.org> 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> --- 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, -- 2.23.0 _______________________________________________ Alsa-devel mailing list Alsa-devel@alsa-project.org https://mailman.alsa-project.org/mailman/listinfo/alsa-devel
next prev parent reply other threads:[~2019-11-15 10:27 UTC|newest] Thread overview: 19+ messages / expand[flat|nested] mbox.gz Atom feed top 2019-11-15 10:27 [RFC PATCH 0/3] ALSA: compress: Add support for FLAC Vinod Koul 2019-11-15 10:27 ` [alsa-devel] " Vinod Koul 2019-11-15 10:27 ` [RFC PATCH 1/3] ALSA: compress: add flac decoder params Vinod Koul 2019-11-15 10:27 ` [alsa-devel] " Vinod Koul 2019-11-18 13:00 ` [alsa-devel] Applied "ALSA: compress: add flac decoder params" to the asoc tree Mark Brown 2019-11-15 10:27 ` Vinod Koul [this message] 2019-11-15 10:27 ` [alsa-devel] [RFC PATCH 2/3] ASoC: qcom: q6asm: add support to flac config Vinod Koul 2019-11-18 13:00 ` [alsa-devel] Applied "ASoC: qcom: q6asm: add support to flac config" to the asoc tree Mark Brown 2019-11-15 10:27 ` [RFC PATCH 3/3] ASoC: qcom: q6asm-dai: add support to flac decoder Vinod Koul 2019-11-15 10:27 ` [alsa-devel] " Vinod Koul 2019-11-18 13:00 ` [alsa-devel] Applied "ASoC: qcom: q6asm-dai: add support to flac decoder" to the asoc tree Mark Brown 2019-11-15 13:21 ` [RFC PATCH 0/3] ALSA: compress: Add support for FLAC Takashi Iwai 2019-11-15 13:21 ` [alsa-devel] " Takashi Iwai 2019-11-15 13:36 ` Vinod Koul 2019-11-15 13:36 ` [alsa-devel] " Vinod Koul 2019-11-15 14:55 ` Pierre-Louis Bossart 2019-11-15 14:55 ` Pierre-Louis Bossart 2019-11-19 4:15 ` Vinod Koul 2019-11-19 4:15 ` Vinod Koul
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