[RFC PATCH 2/3] ASoC: qcom: q6asm: add support to flac config

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

