ASoC: msm: qdsp6v2: Add FLAC in compress offload path

Add FLAC format in compress offload driver, and asm
module

Change-Id: I818ace8397e761b1acff7f9b2eab6e0103ed78c8
Signed-off-by: Apurupa Pattapu <apurupa@codeaurora.org>

Change-Id: I40310136eb980442eed4bd3b0739ed9ffeb8e033

include/sound/apr_audio-v2.h

@@ -2634,6 +2634,19 @@ struct asm_amrwbplus_cfg {
 	u32  amr_lsf_idx;
 } __packed;
 
+struct asm_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;
+};
+
 struct asm_softpause_params {
 	u32 enable;
 	u32 period;
@@ -2981,6 +2994,73 @@ struct asm_aac_enc_cfg_v2 {
 
 } __packed;
 
+struct asm_flac_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+
+	u16 is_stream_info_present;
+/* Specifies whether stream information is present in the FLAC format
+ * block.
+ *
+ * Supported values:
+ * - 0 -- Stream information is not present in this message
+ * - 1 -- Stream information is present in this message
+ *
+ * When set to 1, the FLAC bitstream was successfully parsed by the
+ * client, and other fields in the FLAC format block can be read by the
+ * decoder to get metadata stream information.
+ */
+
+	u16 num_channels;
+/* Number of channels for decoding.
+ * Supported values: 1 to 2
+ */
+
+	u16 min_blk_size;
+/* Minimum block size (in samples) used in the stream. It must be less
+ * than or equal to max_blk_size.
+ */
+
+	u16 max_blk_size;
+/* Maximum block size (in samples) used in the stream. If the
+ * minimum block size equals the maximum block size, a fixed block
+ * size stream is implied.
+ */
+
+	u16 md5_sum[8];
+/* MD5 signature array of the unencoded audio data. This allows the
+ * decoder to determine if an error exists in the audio data, even when
+ * the error does not result in an invalid bitstream.
+ */
+
+	u32 sample_rate;
+/* Number of samples per second.
+ * Supported values: 8000 to 48000 Hz
+ */
+
+	u32 min_frame_size;
+/* Minimum frame size used in the stream.
+ * Supported values:
+ * - > 0 bytes
+ * - 0 -- The value is unknown
+ */
+
+	u32 max_frame_size;
+/* Maximum frame size used in the stream.
+ * Supported values:
+ * -- > 0 bytes
+ * -- 0 . The value is unknown
+ */
+
+	u16 sample_size;
+/* Bits per sample.Supported values: 8, 16 */
+
+	u16 reserved;
+/* Clients must set this field to zero
+ */
+
+} __packed;
+
 #define ASM_MEDIA_FMT_AMRNB_FS                  0x00010BEB
 
 /* Enumeration for 4.75 kbps AMR-NB Encoding mode. */
@@ -3385,6 +3465,7 @@ struct asm_amrwbplus_fmt_blk_v2 {
 #define ASM_MEDIA_FMT_EAC3_DEC                   0x00010C3C
 #define ASM_MEDIA_FMT_DTS                    0x00010D88
 #define ASM_MEDIA_FMT_MP2                    0x00010DE9
+#define ASM_MEDIA_FMT_FLAC                   0x00010C16
 
 /* Media format ID for adaptive transform acoustic coding. This
  * ID is used by the #ASM_STREAM_CMD_OPEN_WRITE_COMPRESSED command

include/sound/q6asm-v2.h

@@ -45,6 +45,7 @@
 #define FORMAT_AC3          0x0013
 #define FORMAT_EAC3         0x0014
 #define FORMAT_MP2          0x0015
+#define FORMAT_FLAC         0x0016
 
 #define ENCDEC_SBCBITRATE   0x0001
 #define ENCDEC_IMMEDIATE_DECODE 0x0002
@@ -373,6 +374,9 @@ int q6asm_media_format_block_wmapro(struct audio_client *ac,
 int q6asm_media_format_block_amrwbplus(struct audio_client *ac,
 			struct asm_amrwbplus_cfg *cfg);
 
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+			struct asm_flac_cfg *cfg, int stream_id);
+
 int q6asm_ds1_set_endp_params(struct audio_client *ac,
 				int param_id, int param_value);
 

include/uapi/sound/compress_params.h

@@ -347,6 +347,13 @@ struct snd_dec_ddp {
 	__u32 params_id[18];
 	__u32 params_value[18];
 };
+struct snd_dec_flac {
+	__u16 sample_size;
+	__u16 min_blk_size;
+	__u16 max_blk_size;
+	__u16 min_frame_size;
+	__u16 max_frame_size;
+};
 union snd_codec_options {
 	struct snd_enc_wma wma;
 	struct snd_enc_vorbis vorbis;
@@ -355,6 +362,7 @@ union snd_codec_options {
 	struct snd_enc_generic generic;
 	struct snd_dec_dts dts;
 	struct snd_dec_ddp ddp;
+	struct snd_dec_flac flac_dec;
 };
 
 /** struct snd_codec_desc - description of codec capabilities

sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c

@@ -510,7 +510,7 @@ static void populate_codec_list(struct msm_compr_audio *prtd)
 			COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
 	prtd->compr_cap.max_fragments =
 			COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
-	prtd->compr_cap.num_codecs = 8;
+	prtd->compr_cap.num_codecs = 9;
 	prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
 	prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
 	prtd->compr_cap.codecs[2] = SND_AUDIOCODEC_AC3;
@@ -519,6 +519,7 @@ static void populate_codec_list(struct msm_compr_audio *prtd)
 	prtd->compr_cap.codecs[5] = SND_AUDIOCODEC_PCM;
 	prtd->compr_cap.codecs[6] = SND_AUDIOCODEC_WMA;
 	prtd->compr_cap.codecs[7] = SND_AUDIOCODEC_WMA_PRO;
+	prtd->compr_cap.codecs[8] = SND_AUDIOCODEC_FLAC;
 }
 
 static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream,
@@ -529,6 +530,7 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream,
 	struct asm_aac_cfg aac_cfg;
 	struct asm_wma_cfg wma_cfg;
 	struct asm_wmapro_cfg wma_pro_cfg;
+	struct asm_flac_cfg flac_cfg;
 	int ret = 0;
 	uint16_t bit_width = 16;
 
@@ -621,6 +623,30 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream,
 		break;
 	case FORMAT_MP2:
 		pr_info("%s: SND_AUDIOCODEC_MP2\n", __func__);
+		break;
+	case FORMAT_FLAC:
+		pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__);
+		memset(&flac_cfg, 0x0, sizeof(struct asm_flac_cfg));
+		flac_cfg.ch_cfg = prtd->num_channels;
+		flac_cfg.sample_rate = prtd->sample_rate;
+		flac_cfg.stream_info_present = 1;
+		flac_cfg.sample_size =
+			prtd->codec_param.codec.options.flac_dec.sample_size;
+		flac_cfg.min_blk_size =
+			prtd->codec_param.codec.options.flac_dec.min_blk_size;
+		flac_cfg.max_blk_size =
+			prtd->codec_param.codec.options.flac_dec.max_blk_size;
+		flac_cfg.max_frame_size =
+			prtd->codec_param.codec.options.flac_dec.max_frame_size;
+		flac_cfg.min_frame_size =
+			prtd->codec_param.codec.options.flac_dec.min_frame_size;
+
+		ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+							&flac_cfg, stream_id);
+		if (ret < 0)
+			pr_err("%s: CMD Format block failed ret %d\n",
+				__func__, ret);
+
 		break;
 	default:
 		pr_info("%s, unsupported format, skip", __func__);
@@ -650,6 +676,11 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream)
 		.rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
 	};
 
+	if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE)
+		bits_per_sample = 24;
+	if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE)
+		bits_per_sample = 32;
+
 	if (prtd->compr_passthr != LEGACY_PCM) {
 		ret = q6asm_open_write_compressed(ac, prtd->codec,
 						  prtd->compr_passthr);
@@ -991,9 +1022,18 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
 	case SNDRV_PCM_RATE_48000:
 		prtd->sample_rate = 48000;
 		break;
+	case SNDRV_PCM_RATE_64000:
+		prtd->sample_rate = 64000;
+		break;
+	case SNDRV_PCM_RATE_88200:
+		prtd->sample_rate = 88200;
+		break;
 	case SNDRV_PCM_RATE_96000:
 		prtd->sample_rate = 96000;
 		break;
+	case SNDRV_PCM_RATE_176400:
+		prtd->sample_rate = 176400;
+		break;
 	case SNDRV_PCM_RATE_192000:
 		prtd->sample_rate = 192000;
 		break;
@@ -1067,6 +1107,12 @@ static int msm_compr_set_params(struct snd_compr_stream *cstream,
 		break;
 	}
 
+	case SND_AUDIOCODEC_FLAC: {
+		pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__);
+		prtd->codec = FORMAT_FLAC;
+		break;
+	}
+
 	default:
 		pr_err("codec not supported, id =%d\n", params->codec.id);
 		return -EINVAL;
@@ -1766,6 +1812,8 @@ static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream,
 		break;
 	case SND_AUDIOCODEC_EAC3:
 		break;
+	case SND_AUDIOCODEC_FLAC:
+		break;
 	default:
 		pr_err("%s: Unsupported audio codec %d\n",
 			__func__, codec->codec);
@@ -1956,6 +2004,7 @@ static int msm_compr_send_dec_params(struct snd_compr_stream *cstream,
 	switch (prtd->codec) {
 	case FORMAT_MP3:
 	case FORMAT_MPEG4_AAC:
+	case FORMAT_FLAC:
 		pr_debug("%s: no runtime parameters for codec: %d\n", __func__,
 			 prtd->codec);
 		break;

sound/soc/msm/qdsp6v2/q6adm.c

@@ -34,6 +34,7 @@
 /* 2 is to account for module & param ID in payload */
 #define ADM_GET_PARAMETER_LENGTH  (4096 - APR_HDR_SIZE - 2 * sizeof(uint32_t))
 
+#define ULL_SUPPORTED_BITS_PER_SAMPLE 16
 #define ULL_SUPPORTED_SAMPLE_RATE 48000
 
 
@@ -1442,6 +1443,7 @@ int adm_open(int port_id, int path, int rate, int channel_mode, int topology,
 			if (perf_mode == ULTRA_LOW_LATENCY_PCM_MODE) {
 				open.topology_id = NULL_COPP_TOPOLOGY;
 				rate = ULL_SUPPORTED_SAMPLE_RATE;
+				open.bit_width = ULL_SUPPORTED_BITS_PER_SAMPLE;
 			} else if (perf_mode == LOW_LATENCY_PCM_MODE) {
 				if ((open.topology_id ==
 					DOLBY_ADM_COPP_TOPOLOGY_ID) ||

sound/soc/msm/qdsp6v2/q6asm.c

@@ -2081,6 +2081,9 @@ static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
 	case FORMAT_MP2:
 		open.dec_fmt_id = ASM_MEDIA_FMT_MP2;
 		break;
+	case FORMAT_FLAC:
+		open.dec_fmt_id = ASM_MEDIA_FMT_FLAC;
+		break;
 	default:
 		pr_err("%s: Invalid format 0x%x\n", __func__, format);
 		goto fail_cmd;
@@ -3289,6 +3292,49 @@ int q6asm_media_format_block_amrwbplus(struct audio_client *ac,
 	return -EINVAL;
 }
 
+int q6asm_stream_media_format_block_flac(struct audio_client *ac,
+				struct asm_flac_cfg *cfg, int stream_id)
+{
+	struct asm_flac_fmt_blk_v2 fmt;
+	int rc = 0;
+
+	pr_debug("%s :session[%d]rate[%d]ch[%d]size[%d]\n", __func__,
+		ac->session, cfg->sample_rate, cfg->ch_cfg, cfg->sample_size);
+
+	q6asm_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE);
+	atomic_set(&ac->cmd_state, 1);
+
+	fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+						sizeof(fmt.fmtblk);
+
+	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 = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+	if (rc < 0) {
+		pr_err("%s :Comamnd media format update failed %d\n",
+				__func__, rc);
+		goto fail_cmd;
+	}
+	rc = wait_event_timeout(ac->cmd_wait,
+				(atomic_read(&ac->cmd_state) == 0), 5*HZ);
+	if (!rc) {
+		pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
+		rc = -ETIMEDOUT;
+		goto fail_cmd;
+	}
+	return 0;
+fail_cmd:
+	return rc;
+}
+
 static int __q6asm_ds1_set_endp_params(struct audio_client *ac, int param_id,
 				int param_value, int stream_id)
 {