Donate to e Foundation | Murena handsets with /e/OS | Own a part of Murena! Learn more

Commit 17153ab7 authored by Satish Babu Patakokila's avatar Satish Babu Patakokila Committed by Gerrit - the friendly Code Review server
Browse files

ASoc: msm: qdsp6v2: Add timestamp support for compress capture 



Add timestamp support for compress driver.
Modify asm driver to read the buffer from predefined offset.

CRs-fixed: 1072067
Change-Id: I1c46befc223285495b3c0650e6c3eaae81f58771
Signed-off-by: default avatarSatish Babu Patakokila <sbpata@codeaurora.org>
parent bc92ad08

include/sound/q6asm-v2.h

+9 −0
Original line number Diff line number Diff line
@@ -73,6 +73,11 @@ 
/* bit 4 represents META enable of encoded data buffer */

#define BUFFER_META_ENABLE	0x0010



/* bit 5 represents timestamp */

/* bit 5 - 0 -- ASM_DATA_EVENT_READ_DONE will have relative time-stamp*/

/* bit 5 - 1 -- ASM_DATA_EVENT_READ_DONE will have absolute time-stamp*/

#define ABSOLUTE_TIMESTAMP_ENABLE  0x0020



/* Enable Sample_Rate/Channel_Mode notification event from Decoder */

#define SR_CM_NOTIFY_ENABLE	0x0004
@@ -156,6 +161,7 @@ struct audio_aio_read_param { 
	phys_addr_t   paddr;

	uint32_t      len;

	uint32_t      uid;

	uint32_t      flags;/*meta data flags*/

};



struct audio_port_data {
@@ -246,6 +252,9 @@ int q6asm_open_read_v2(struct audio_client *ac, uint32_t format, 
int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,

		       uint16_t bits_per_sample);



int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,

			uint16_t bits_per_sample);



int q6asm_open_write(struct audio_client *ac, uint32_t format

		/*, uint16_t bits_per_sample*/);

include/uapi/sound/compress_params.h

+14 −1
Original line number Diff line number Diff line
@@ -70,6 +70,11 @@ 
#define Q6_DTS		0x00010D88

#define Q6_DTS_LBR	0x00010DBB



/* Timestamp flsg */

/* Bit-0 - 1 : Enable Timestamp mode */

/* Bit-0 - 0 : Disable Timestamp mode */

#define COMPRESSED_TIMESTAMP_FLAG 0x0001



/* Codecs are listed linearly to allow for extensibility */

#define SND_AUDIOCODEC_PCM                   ((__u32) 0x00000001)

#define SND_AUDIOCODEC_MP3                   ((__u32) 0x00000002)
@@ -480,7 +485,15 @@ struct snd_codec { 
	__u32 align;

	__u32 compr_passthr;

	union snd_codec_options options;

	__u32 reserved[3];

	__u32 flags;

	__u32 reserved[2];

} __attribute__((packed, aligned(4)));



struct snd_codec_metadata {

	__u32 length;

	__u32 offset;

	__u64 timestamp;

	__u32 reserved[4];

};



#endif

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

+41 −7
Original line number Diff line number Diff line
@@ -55,6 +55,10 @@ 
#define DSP_NUM_OUTPUT_FRAME_BUFFERED	2

#define FLAC_BLK_SIZE_LIMIT		65535



/* Timestamp mode payload offsets */

#define TS_LSW_OFFSET			6

#define TS_MSW_OFFSET			7



/* decoder parameter length */

#define DDP_DEC_MAX_NUM_PARAM		18
@@ -139,6 +143,8 @@ struct msm_compr_audio { 
	uint64_t bytes_read; /* from DSP */

	uint32_t bytes_read_offset; /* bytes read offset */



	uint32_t ts_header_offset; /* holds the timestamp header size */



	int32_t first_buffer;

	int32_t last_buffer;

	int32_t partial_drain_delay;
@@ -402,26 +408,31 @@ static int msm_compr_read_buffer(struct msm_compr_audio *prtd) 
		return -EINVAL;

	}



	buffer_length = prtd->codec_param.buffer.fragment_size;

	buffer_length = prtd->codec_param.buffer.fragment_size -

						 prtd->ts_header_offset;

	bytes_available = prtd->received_total - prtd->bytes_copied;

	buffer_sent = prtd->bytes_read - prtd->bytes_copied;

	if (buffer_sent + buffer_length > prtd->buffer_size) {

	if (buffer_sent + buffer_length + prtd->ts_header_offset

						> prtd->buffer_size) {

		pr_debug(" %s : Buffer is Full bytes_available: %llu\n",

				__func__, bytes_available);

		return 0;

	}



	param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset;

	param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset +

						prtd->ts_header_offset;

	param.len = buffer_length;

	param.uid = buffer_length;

	param.flags = prtd->codec_param.codec.flags;



	pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n",

			__func__, buffer_length, prtd->bytes_read);

	if (q6asm_async_read(prtd->audio_client, &param) < 0) {

		pr_err("%s:q6asm_async_read failed\n", __func__);

	} else {

		prtd->bytes_read += buffer_length;

		prtd->bytes_read_offset += buffer_length;

		prtd->bytes_read += buffer_length + prtd->ts_header_offset;

		prtd->bytes_read_offset += buffer_length +

							prtd->ts_header_offset;

		if (prtd->bytes_read_offset >= prtd->buffer_size)

			prtd->bytes_read_offset -= prtd->buffer_size;

	}
@@ -442,6 +453,7 @@ static void compr_event_handler(uint32_t opcode, 
	uint32_t stream_index;

	unsigned long flags;

	uint64_t read_size;

	uint32_t *buff_addr;



	if (!prtd) {

		pr_err("%s: prtd is NULL\n", __func__);
@@ -525,6 +537,23 @@ static void compr_event_handler(uint32_t opcode, 


		pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n",

				 prtd->byte_offset, payload[4]);



		if (prtd->ts_header_offset) {

			/* Update the header for received buffer */

			buff_addr = prtd->buffer + prtd->byte_offset;

			/* Write the length of the buffer */

			*buff_addr = prtd->codec_param.buffer.fragment_size

						 - prtd->ts_header_offset;

			buff_addr++;

			/* Write the offset */

			*buff_addr = prtd->ts_header_offset;

			buff_addr++;

			/* Write the TS LSW */

			*buff_addr = payload[TS_LSW_OFFSET];

			buff_addr++;

			/* Write the TS MSW */

			*buff_addr = payload[TS_MSW_OFFSET];

		}

		/* Always assume read_size is same as fragment_size */

		read_size = prtd->codec_param.buffer.fragment_size;

		prtd->byte_offset += read_size;
@@ -1232,7 +1261,7 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
	pr_debug("%s: stream_id %d bits_per_sample %d\n",

			__func__, ac->stream_id, bits_per_sample);



	ret = q6asm_open_read_v3(prtd->audio_client, FORMAT_LINEAR_PCM,

	ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,

		bits_per_sample);

	if (ret < 0) {

		pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret);
@@ -1283,6 +1312,11 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream) 
	prtd->buffer_paddr   = ac->port[dir].buf[0].phys;

	prtd->buffer_size    = runtime->fragments * runtime->fragment_size;



	/* Bit-0 of flags represent timestamp mode */

	if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG)

		prtd->ts_header_offset = sizeof(struct snd_codec_metadata);

	else

		prtd->ts_header_offset = 0;



	pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",

			__func__, prtd->sample_rate, prtd->num_channels,

sound/soc/msm/qdsp6v2/q6asm.c

+26 −5
Original line number Diff line number Diff line
@@ -2201,7 +2201,7 @@ static void q6asm_add_mmaphdr(struct audio_client *ac, struct apr_hdr *hdr, 


static int __q6asm_open_read(struct audio_client *ac,

			     uint32_t format, uint16_t bits_per_sample,

			     bool use_v3_format)

			     bool use_v3_format, bool ts_mode)

{

	int rc = 0x00;

	struct asm_stream_cmd_open_read_v3 open;
@@ -2244,6 +2244,8 @@ static int __q6asm_open_read(struct audio_client *ac, 
	switch (format) {

	case FORMAT_LINEAR_PCM:

		open.mode_flags |= 0x00;

		if (ts_mode)

			open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE;

		if (use_v3_format)

			open.enc_cfg_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3;

		else
@@ -2318,14 +2320,14 @@ int q6asm_open_read(struct audio_client *ac, 
		uint32_t format)

{

	return __q6asm_open_read(ac, format, 16,

				 false /*use_v3_format*/);

				 false /*use_v3_format*/, false/*ts_mode*/);

}



int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,

			uint16_t bits_per_sample)

{

	return __q6asm_open_read(ac, format, bits_per_sample,

				 false /*use_v3_format*/);

				 false /*use_v3_format*/, false/*ts_mode*/);

}



/*
@@ -2339,10 +2341,25 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format, 
			uint16_t bits_per_sample)

{

	return __q6asm_open_read(ac, format, bits_per_sample,

				 true /*use_v3_format*/);

				 true /*use_v3_format*/, false/*ts_mode*/);

}

EXPORT_SYMBOL(q6asm_open_read_v3);



/*

 * asm_open_read_v4 - Opens audio capture session

 *

 * @ac: Client session handle

 * @format: encoder format

 * @bits_per_sample: bit width of capture session

 */

int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,

			uint16_t bits_per_sample)

{

	return __q6asm_open_read(ac, format, bits_per_sample,

				true /*use_v3_format*/, true/*ts_mode*/);

}

EXPORT_SYMBOL(q6asm_open_read_v4);



int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format,

				uint32_t passthrough_flag)

{
@@ -6708,6 +6725,10 @@ int q6asm_async_read(struct audio_client *ac, 
		lbuf_phys_addr = (param->paddr - 64);

		dir = OUT;

	} else {

		if (param->flags & COMPRESSED_TIMESTAMP_FLAG)

			lbuf_phys_addr = param->paddr -

				 sizeof(struct snd_codec_metadata);

		else

			lbuf_phys_addr = param->paddr;

		dir = OUT;

	}