Merge "ASoC: msm: qdsp6v2: Add FLAC in compress offload path" (d907e5eb) · Commits · e / devices / android_kernel_sony_msm8994

include/sound/apr_audio-v2.h

+82 −0

Original line number	Diff line number	Diff line
		@@ -2571,6 +2571,19 @@ 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;
		@@ -2914,6 +2927,73 @@ 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. */
		@@ -3318,6 +3398,8 @@ 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
		@@ -358,6 +359,9 @@ 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;
		@@ -1663,6 +1708,8 @@ 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);
		@@ -1854,6 +1901,7 @@ 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
		@@ -32,6 +32,7 @@
		/* 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 {
		@@ -1408,6 +1409,7 @@ 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) \|\|