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: Satish Babu Patakokila <sbpata@codeaurora.org>
This commit is contained in:
Satish Babu Patakokila
parent
04947ea3d8
commit
160813b82f
4 changed files with 84 additions and 12 deletions
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@ -73,6 +73,11 @@
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/* bit 4 represents META enable of encoded data buffer */
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#define BUFFER_META_ENABLE 0x0010
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/* bit 5 represents timestamp */
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/* bit 5 - 0 -- ASM_DATA_EVENT_READ_DONE will have relative time-stamp*/
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/* bit 5 - 1 -- ASM_DATA_EVENT_READ_DONE will have absolute time-stamp*/
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#define ABSOLUTE_TIMESTAMP_ENABLE 0x0020
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/* Enable Sample_Rate/Channel_Mode notification event from Decoder */
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#define SR_CM_NOTIFY_ENABLE 0x0004
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@ -175,6 +180,7 @@ struct audio_aio_read_param {
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phys_addr_t paddr;
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uint32_t len;
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uint32_t uid;
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uint32_t flags;/*meta data flags*/
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};
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struct audio_port_data {
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@ -70,6 +70,11 @@
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#define Q6_DTS 0x00010D88
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#define Q6_DTS_LBR 0x00010DBB
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/* Timestamp flsg */
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/* Bit-0 - 1 : Enable Timestamp mode */
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/* Bit-0 - 0 : Disable Timestamp mode */
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#define COMPRESSED_TIMESTAMP_FLAG 0x0001
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/* Codecs are listed linearly to allow for extensibility */
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#define SND_AUDIOCODEC_PCM ((__u32) 0x00000001)
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#define SND_AUDIOCODEC_MP3 ((__u32) 0x00000002)
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@ -480,7 +485,24 @@ struct snd_codec {
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__u32 align;
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__u32 compr_passthr;
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union snd_codec_options options;
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__u32 reserved[3];
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__u32 flags;
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__u32 reserved[2];
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} __attribute__((packed, aligned(4)));
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/** struct snd_codec_metadata
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* @length: Length of the encoded buffer.
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* @offset: Offset from the buffer address to the first byte of the first
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* encoded frame. All encoded frames are consecutive starting
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* from this offset.
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* @timestamp: Session time in microseconds of the first sample in the buffer.
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* @reserved: Reserved for future use.
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*/
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struct snd_codec_metadata {
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__u32 length;
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__u32 offset;
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__u64 timestamp;
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__u32 reserved[4];
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};
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#endif
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@ -55,6 +55,10 @@
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#define DSP_NUM_OUTPUT_FRAME_BUFFERED 2
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#define FLAC_BLK_SIZE_LIMIT 65535
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/* Timestamp mode payload offsets */
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#define TS_LSW_OFFSET 6
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#define TS_MSW_OFFSET 7
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/* decoder parameter length */
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#define DDP_DEC_MAX_NUM_PARAM 18
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@ -130,7 +134,9 @@ struct msm_compr_audio {
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uint64_t received_total; /* bytes received from DSP */
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uint64_t bytes_copied; /* to userspace */
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uint64_t bytes_read; /* from DSP */
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uint32_t bytes_read_offset; /* bytes read offset*/
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uint32_t bytes_read_offset; /* bytes read offset */
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uint32_t ts_header_offset; /* holds the timestamp header offset */
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int32_t first_buffer;
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int32_t last_buffer;
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@ -380,19 +386,23 @@ static int msm_compr_read_buffer(struct msm_compr_audio *prtd)
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return -EINVAL;
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}
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buffer_length = prtd->codec_param.buffer.fragment_size;
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buffer_length = prtd->codec_param.buffer.fragment_size -
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prtd->ts_header_offset;
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bytes_available = prtd->received_total - prtd->bytes_copied;
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buffer_sent = prtd->bytes_read - prtd->bytes_copied;
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if (buffer_sent + buffer_length > prtd->buffer_size) {
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if (buffer_sent + buffer_length + prtd->ts_header_offset
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> prtd->buffer_size) {
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pr_debug(" %s : Buffer is Full bytes_available: %llu\n",
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__func__, bytes_available);
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return 0;
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}
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memset(¶m, 0x0, sizeof(struct audio_aio_read_param));
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param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset;
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param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset +
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prtd->ts_header_offset;
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param.len = buffer_length;
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param.uid = buffer_length;
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param.flags = prtd->codec_param.codec.flags;
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pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n",
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__func__, buffer_length, prtd->bytes_read);
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@ -423,6 +433,7 @@ static void compr_event_handler(uint32_t opcode,
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uint32_t stream_index;
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unsigned long flags;
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uint64_t read_size;
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uint32_t *buff_addr;
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if (!prtd) {
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pr_err("%s: prtd is NULL\n", __func__);
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@ -514,6 +525,23 @@ static void compr_event_handler(uint32_t opcode,
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pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n",
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prtd->byte_offset, payload[4]);
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if (prtd->ts_header_offset) {
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/* Update the header for received buffer */
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buff_addr = prtd->buffer + prtd->byte_offset;
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/* Write the length of the buffer */
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*buff_addr = prtd->codec_param.buffer.fragment_size
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- prtd->ts_header_offset;
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buff_addr++;
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/* Write the offset */
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*buff_addr = prtd->ts_header_offset;
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buff_addr++;
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/* Write the TS LSW */
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*buff_addr = payload[TS_LSW_OFFSET];
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buff_addr++;
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/* Write the TS MSW */
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*buff_addr = payload[TS_MSW_OFFSET];
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}
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/* Always assume read_size is same as fragment_size */
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read_size = prtd->codec_param.buffer.fragment_size;
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prtd->byte_offset += read_size;
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@ -1214,7 +1242,7 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
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pr_debug("%s: stream_id %d bits_per_sample %d\n",
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__func__, ac->stream_id, bits_per_sample);
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ret = q6asm_open_read_v3(prtd->audio_client, FORMAT_LINEAR_PCM,
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ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
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bits_per_sample);
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if (ret < 0) {
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pr_err("%s: q6asm_open_read failed:%d\n", __func__, ret);
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@ -1265,6 +1293,11 @@ static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
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prtd->buffer_paddr = ac->port[dir].buf[0].phys;
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prtd->buffer_size = runtime->fragments * runtime->fragment_size;
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/* Bit-0 of flags represent timestamp mode */
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if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG)
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prtd->ts_header_offset = sizeof(struct snd_codec_metadata);
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else
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prtd->ts_header_offset = 0;
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pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",
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__func__, prtd->sample_rate, prtd->num_channels,
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@ -2285,7 +2285,8 @@ static void q6asm_add_mmaphdr(struct audio_client *ac, struct apr_hdr *hdr,
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static int __q6asm_open_read(struct audio_client *ac,
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uint32_t format, uint16_t bits_per_sample,
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uint32_t pcm_format_block_ver)
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uint32_t pcm_format_block_ver,
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bool ts_mode)
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{
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int rc = 0x00;
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struct asm_stream_cmd_open_read_v3 open;
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@ -2329,6 +2330,8 @@ static int __q6asm_open_read(struct audio_client *ac,
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case FORMAT_LINEAR_PCM:
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open.mode_flags |= 0x00;
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open.enc_cfg_id = q6asm_get_pcm_format_id(pcm_format_block_ver);
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if (ts_mode)
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open.mode_flags |= ABSOLUTE_TIMESTAMP_ENABLE;
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break;
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case FORMAT_MPEG4_AAC:
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open.mode_flags |= BUFFER_META_ENABLE;
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@ -2391,14 +2394,16 @@ int q6asm_open_read(struct audio_client *ac,
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uint32_t format)
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{
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return __q6asm_open_read(ac, format, 16,
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PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/);
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PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
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false/*ts_mode*/);
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}
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int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,
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uint16_t bits_per_sample)
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{
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return __q6asm_open_read(ac, format, bits_per_sample,
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PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/);
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PCM_MEDIA_FORMAT_V2 /*media fmt block ver*/,
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false/*ts_mode*/);
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}
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/*
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@ -2412,7 +2417,8 @@ int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
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uint16_t bits_per_sample)
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{
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return __q6asm_open_read(ac, format, bits_per_sample,
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PCM_MEDIA_FORMAT_V3/*media fmt block ver*/);
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PCM_MEDIA_FORMAT_V3/*media fmt block ver*/,
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false/*ts_mode*/);
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}
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EXPORT_SYMBOL(q6asm_open_read_v3);
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@ -2427,7 +2433,8 @@ int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
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uint16_t bits_per_sample)
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{
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return __q6asm_open_read(ac, format, bits_per_sample,
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PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/);
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PCM_MEDIA_FORMAT_V4 /*media fmt block ver*/,
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true/*ts_mode*/);
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}
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EXPORT_SYMBOL(q6asm_open_read_v4);
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@ -7150,7 +7157,11 @@ int q6asm_async_read(struct audio_client *ac,
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lbuf_phys_addr = (param->paddr - 64);
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dir = OUT;
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} else {
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lbuf_phys_addr = param->paddr;
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if (param->flags & COMPRESSED_TIMESTAMP_FLAG)
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lbuf_phys_addr = param->paddr -
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sizeof(struct snd_codec_metadata);
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else
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lbuf_phys_addr = param->paddr;
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dir = OUT;
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}
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