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Commit 1b3e5e64 authored by Apurupa Pattapu's avatar Apurupa Pattapu
Browse files

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: default avatarApurupa Pattapu <apurupa@codeaurora.org>
parent c575d3e6

include/sound/apr_audio-v2.h

+82 −0
Original line number Diff line number Diff line
@@ -2570,6 +2570,19 @@ 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;
@@ -2917,6 +2930,73 @@ 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. */
@@ -3321,6 +3401,8 @@ 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

+4 −0
Original line number Diff line number Diff line
@@ -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
@@ -360,6 +361,9 @@ 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

+8 −0
Original line number Diff line number Diff line
@@ -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

+48 −0
Original line number Diff line number Diff line
@@ -495,6 +495,7 @@ 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,
@@ -505,6 +506,7 @@ 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;
@@ -597,6 +599,30 @@ static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream, 
		break;

	case FORMAT_MP2:

		pr_debug("%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_debug("%s, unsupported format, skip", __func__);
@@ -619,6 +645,10 @@ static int msm_compr_configure_dsp(struct snd_compr_stream *cstream) 
		.step = SOFT_VOLUME_STEP,

		.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;



	pr_debug("%s: stream_id %d\n", __func__, ac->stream_id);

	ret = q6asm_stream_open_write_v2(ac,
@@ -909,9 +939,18 @@ 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;
@@ -972,6 +1011,12 @@ 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;
@@ -1657,6 +1702,8 @@ 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);
@@ -1848,6 +1895,7 @@ static int msm_compr_dec_params_put(struct snd_kcontrol *kcontrol, 
	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

+2 −0
Original line number Diff line number Diff line
@@ -33,6 +33,7 @@ 
/* 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



enum {
@@ -1479,6 +1480,7 @@ int adm_open(int port_id, int path, int rate, int channel_mode, 
		flags = ADM_ULTRA_LOW_LATENCY_DEVICE_SESSION;

		topology = NULL_COPP_TOPOLOGY;

		rate = ULL_SUPPORTED_SAMPLE_RATE;

		bit_width = ULL_SUPPORTED_BITS_PER_SAMPLE;

	} else if (perf_mode == LOW_LATENCY_PCM_MODE) {

		flags = ADM_LOW_LATENCY_DEVICE_SESSION;

		if ((topology == DOLBY_ADM_COPP_TOPOLOGY_ID) ||