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refine hls, extract ts cache for ts cache audio and flush once

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This commit is contained in:
winlin 2013-12-02 22:09:10 +08:00
parent 1c2f5052dd
commit b5bba29768
2 changed files with 281 additions and 202 deletions
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434
trunk/src/core/srs_core_hls.cpp
View file

@ -856,13 +856,8 @@ int SrsM3u8Muxer::create_dir()
return ret;
}
SrsHls::SrsHls()
SrsTSCache::SrsTSCache()
{
hls_enabled = false;
codec = new SrsCodec();
sample = new SrsCodecSample();
jitter = new SrsRtmpJitter();
aac_jitter = new SrsHlsAacJitter();
ab = new SrsCodecBuffer();
@ -870,15 +865,10 @@ SrsHls::SrsHls()
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
muxer = new SrsM3u8Muxer();
}
SrsHls::~SrsHls()
SrsTSCache::~SrsTSCache()
{
srs_freep(codec);
srs_freep(sample);
srs_freep(jitter);
srs_freep(aac_jitter);
ab->free();
@ -889,8 +879,237 @@ SrsHls::~SrsHls()
srs_freep(af);
srs_freep(vf);
}
int SrsTSCache::write_audio(SrsCodec* codec, SrsM3u8Muxer* muxer, int64_t pts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// start buffer, set the af
if (ab->size == 0) {
pts = aac_jitter->on_buffer_start(pts, sample->sound_rate);
af->dts = af->pts = audio_buffer_start_pts = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
}
// write audio to cache.
if ((ret = cache_audio(codec, sample)) != ERROR_SUCCESS) {
return ret;
}
// flush if buffer exceed max size.
if (ab->size > SRS_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
return ret;
}
}
// TODO: config it.
// in ms, audio delay to flush the audios.
int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
// flush if audio delay exceed
if (pts - audio_buffer_start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
return ret;
}
}
return ret;
}
int SrsTSCache::write_video(SrsCodec* codec, SrsM3u8Muxer* muxer, int64_t dts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// write video to cache.
if ((ret = cache_video(codec, sample)) != ERROR_SUCCESS) {
return ret;
}
vf->dts = dts;
vf->pts = vf->dts + sample->cts * 90;
vf->pid = TS_VIDEO_PID;
vf->sid = TS_VIDEO_AVC;
vf->key = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
// flush video when got one
if ((ret = muxer->flush_video(af, ab, vf, vb)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush video failed. ret=%d", ret);
return ret;
}
return ret;
}
int SrsTSCache::flush_audio(SrsM3u8Muxer* muxer)
{
int ret = ERROR_SUCCESS;
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
return ret;
}
return ret;
}
int SrsTSCache::cache_audio(SrsCodec* codec, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
for (int i = 0; i < sample->nb_buffers; i++) {
SrsCodecBuffer* buf = &sample->buffers[i];
int32_t size = buf->size;
if (!buf->bytes || size <= 0 || size > 0x1fff) {
ret = ERROR_HLS_AAC_FRAME_LENGTH;
srs_error("invalid aac frame length=%d, ret=%d", size, ret);
return ret;
}
// the frame length is the AAC raw data plus the adts header size.
int32_t frame_length = size + 7;
// AAC-ADTS
// 6.2 Audio Data Transport Stream, ADTS
// in aac-iso-13818-7.pdf, page 26.
// fixed 7bytes header
static u_int8_t adts_header[7] = {0xff, 0xf1, 0x00, 0x00, 0x00, 0x0f, 0xfc};
/*
// adts_fixed_header
// 2B, 16bits
int16_t syncword; //12bits, '1111 1111 1111'
int8_t ID; //1bit, '0'
int8_t layer; //2bits, '00'
int8_t protection_absent; //1bit, can be '1'
// 12bits
int8_t profile; //2bit, 7.1 Profiles, page 40
TSAacSampleFrequency sampling_frequency_index; //4bits, Table 35, page 46
int8_t private_bit; //1bit, can be '0'
int8_t channel_configuration; //3bits, Table 8
int8_t original_or_copy; //1bit, can be '0'
int8_t home; //1bit, can be '0'
// adts_variable_header
// 28bits
int8_t copyright_identification_bit; //1bit, can be '0'
int8_t copyright_identification_start; //1bit, can be '0'
int16_t frame_length; //13bits
int16_t adts_buffer_fullness; //11bits, 7FF signals that the bitstream is a variable rate bitstream.
int8_t number_of_raw_data_blocks_in_frame; //2bits, 0 indicating 1 raw_data_block()
*/
// profile, 2bits
adts_header[2] = (codec->aac_profile << 6) & 0xc0;
// sampling_frequency_index 4bits
adts_header[2] |= (codec->aac_sample_rate << 2) & 0x3c;
// channel_configuration 3bits
adts_header[2] |= (codec->aac_channels >> 2) & 0x01;
adts_header[3] = (codec->aac_channels << 6) & 0xc0;
// frame_length 13bits
adts_header[3] |= (frame_length >> 11) & 0x03;
adts_header[4] = (frame_length >> 3) & 0xff;
adts_header[5] = ((frame_length << 5) & 0xe0);
// adts_buffer_fullness; //11bits
adts_header[5] |= 0x1f;
// copy to audio buffer
ab->append(adts_header, sizeof(adts_header));
ab->append(buf->bytes, buf->size);
}
return ret;
}
int SrsTSCache::cache_video(SrsCodec* codec, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
static u_int8_t aud_nal[] = { 0x00, 0x00, 0x00, 0x01, 0x09, 0xf0 };
vb->append(aud_nal, sizeof(aud_nal));
bool sps_pps_sent = false;
for (int i = 0; i < sample->nb_buffers; i++) {
SrsCodecBuffer* buf = &sample->buffers[i];
int32_t size = buf->size;
if (!buf->bytes || size <= 0) {
ret = ERROR_HLS_AVC_SAMPLE_SIZE;
srs_error("invalid avc sample length=%d, ret=%d", size, ret);
return ret;
}
// 5bits, 7.3.1 NAL unit syntax,
// H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
u_int8_t nal_unit_type;
nal_unit_type = *buf->bytes;
nal_unit_type &= 0x1f;
// Table 7-1 – NAL unit type codes, page 61
// 1: Coded slice
if (nal_unit_type == 1) {
sps_pps_sent = false;
}
// 5: Coded slice of an IDR picture.
// insert sps/pps before IDR or key frame is ok.
if (nal_unit_type == 5 && !sps_pps_sent) {
//if (vf->key && !sps_pps_sent) {
sps_pps_sent = true;
// ngx_rtmp_hls_append_sps_pps
if (codec->sequenceParameterSetLength > 0) {
// AnnexB prefix
vb->append(aud_nal, 4);
// sps
vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
}
if (codec->pictureParameterSetLength > 0) {
// AnnexB prefix
vb->append(aud_nal, 4);
// pps
vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
}
}
// sample start prefix, '00 00 00 01' or '00 00 01'
u_int8_t* p = aud_nal + 1;
u_int8_t* end = p + 3;
// first AnnexB prefix is long (4 bytes)
if (i == 0) {
p = aud_nal;
}
vb->append(p, end - p);
// sample data
vb->append(buf->bytes, buf->size);
}
return ret;
}
SrsHls::SrsHls()
{
hls_enabled = false;
codec = new SrsCodec();
sample = new SrsCodecSample();
jitter = new SrsRtmpJitter();
muxer = new SrsM3u8Muxer();
ts_cache = new SrsTSCache();
}
SrsHls::~SrsHls()
{
srs_freep(codec);
srs_freep(sample);
srs_freep(jitter);
srs_freep(muxer);
srs_freep(ts_cache);
}
int SrsHls::on_publish(SrsRequest* req)
@ -953,7 +1172,7 @@ void SrsHls::on_unpublish()
int ret = ERROR_SUCCESS;
// close muxer when unpublish.
ret = muxer->flush_audio(af, ab);
ret = ts_cache->flush_audio(muxer);
ret += muxer->segment_close();
if (ret != ERROR_SUCCESS) {
srs_error("ignore m3u8 muxer flush/close audio failed. ret=%d", ret);
@ -1028,6 +1247,7 @@ int SrsHls::on_audio(SrsSharedPtrMessage* audio)
sample->clear();
if ((ret = codec->audio_aac_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
srs_error("codec demux audio failed. ret=%d", ret);
return ret;
}
@ -1042,44 +1262,18 @@ int SrsHls::on_audio(SrsSharedPtrMessage* audio)
int64_t corrected_time = 0;
if ((ret = jitter->correct(audio, 0, 0, &corrected_time)) != ERROR_SUCCESS) {
srs_error("rtmp jitter correct audio failed. ret=%d", ret);
return ret;
}
// the pts calc from rtmp/flv header.
int64_t pts = corrected_time * 90;
// start buffer, set the af
if (ab->size == 0) {
pts = aac_jitter->on_buffer_start(pts, sample->sound_rate);
af->dts = af->pts = audio_buffer_start_pts = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
}
// write audio to cache.
if ((ret = cache_audio()) != ERROR_SUCCESS) {
if ((ret = ts_cache->write_audio(codec, muxer, pts, sample)) != ERROR_SUCCESS) {
srs_error("ts cache write audio failed. ret=%d", ret);
return ret;
}
// flush if buffer exceed max size.
if (ab->size > SRS_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
return ret;
}
}
// TODO: config it.
// in ms, audio delay to flush the audios.
int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
// flush if audio delay exceed
if (pts - audio_buffer_start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(af, ab)) != ERROR_SUCCESS) {
return ret;
}
}
return ret;
}
@ -1096,6 +1290,7 @@ int SrsHls::on_video(SrsSharedPtrMessage* video)
sample->clear();
if ((ret = codec->video_avc_demux(video->payload, video->size, sample)) != ERROR_SUCCESS) {
srs_error("codec demux video failed. ret=%d", ret);
return ret;
}
@ -1104,166 +1299,23 @@ int SrsHls::on_video(SrsSharedPtrMessage* video)
}
// ignore sequence header
if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame && sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame
&& sample->avc_packet_type == SrsCodecVideoAVCTypeSequenceHeader) {
return ret;
}
int64_t corrected_time = 0;
if ((ret = jitter->correct(video, 0, 0, &corrected_time)) != ERROR_SUCCESS) {
srs_error("rtmp jitter correct video failed. ret=%d", ret);
return ret;
}
// write video to cache.
if ((ret = cache_video()) != ERROR_SUCCESS) {
int64_t dts = corrected_time * 90;
if ((ret = ts_cache->write_video(codec, muxer, dts, sample)) != ERROR_SUCCESS) {
srs_error("ts cache write video failed. ret=%d", ret);
return ret;
}
vf->dts = corrected_time * 90;
vf->pts = vf->dts + sample->cts * 90;
vf->pid = TS_VIDEO_PID;
vf->sid = TS_VIDEO_AVC;
vf->key = sample->frame_type == SrsCodecVideoAVCFrameKeyFrame;
// flush video when got one
if ((ret = muxer->flush_video(af, ab, vf, vb)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush video failed. ret=%d", ret);
return ret;
}
return ret;
}
int SrsHls::cache_audio()
{
int ret = ERROR_SUCCESS;
for (int i = 0; i < sample->nb_buffers; i++) {
SrsCodecBuffer* buf = &sample->buffers[i];
int32_t size = buf->size;
if (!buf->bytes || size <= 0 || size > 0x1fff) {
ret = ERROR_HLS_AAC_FRAME_LENGTH;
srs_error("invalid aac frame length=%d, ret=%d", size, ret);
return ret;
}
// the frame length is the AAC raw data plus the adts header size.
int32_t frame_length = size + 7;
// AAC-ADTS
// 6.2 Audio Data Transport Stream, ADTS
// in aac-iso-13818-7.pdf, page 26.
// fixed 7bytes header
static u_int8_t adts_header[7] = {0xff, 0xf1, 0x00, 0x00, 0x00, 0x0f, 0xfc};
/*
// adts_fixed_header
// 2B, 16bits
int16_t syncword; //12bits, '1111 1111 1111'
int8_t ID; //1bit, '0'
int8_t layer; //2bits, '00'
int8_t protection_absent; //1bit, can be '1'
// 12bits
int8_t profile; //2bit, 7.1 Profiles, page 40
TSAacSampleFrequency sampling_frequency_index; //4bits, Table 35, page 46
int8_t private_bit; //1bit, can be '0'
int8_t channel_configuration; //3bits, Table 8
int8_t original_or_copy; //1bit, can be '0'
int8_t home; //1bit, can be '0'
// adts_variable_header
// 28bits
int8_t copyright_identification_bit; //1bit, can be '0'
int8_t copyright_identification_start; //1bit, can be '0'
int16_t frame_length; //13bits
int16_t adts_buffer_fullness; //11bits, 7FF signals that the bitstream is a variable rate bitstream.
int8_t number_of_raw_data_blocks_in_frame; //2bits, 0 indicating 1 raw_data_block()
*/
// profile, 2bits
adts_header[2] = (codec->aac_profile << 6) & 0xc0;
// sampling_frequency_index 4bits
adts_header[2] |= (codec->aac_sample_rate << 2) & 0x3c;
// channel_configuration 3bits
adts_header[2] |= (codec->aac_channels >> 2) & 0x01;
adts_header[3] = (codec->aac_channels << 6) & 0xc0;
// frame_length 13bits
adts_header[3] |= (frame_length >> 11) & 0x03;
adts_header[4] = (frame_length >> 3) & 0xff;
adts_header[5] = ((frame_length << 5) & 0xe0);
// adts_buffer_fullness; //11bits
adts_header[5] |= 0x1f;
// copy to audio buffer
ab->append(adts_header, sizeof(adts_header));
ab->append(buf->bytes, buf->size);
}
return ret;
}
int SrsHls::cache_video()
{
int ret = ERROR_SUCCESS;
static u_int8_t aud_nal[] = { 0x00, 0x00, 0x00, 0x01, 0x09, 0xf0 };
vb->append(aud_nal, sizeof(aud_nal));
bool sps_pps_sent = false;
for (int i = 0; i < sample->nb_buffers; i++) {
SrsCodecBuffer* buf = &sample->buffers[i];
int32_t size = buf->size;
if (!buf->bytes || size <= 0) {
ret = ERROR_HLS_AVC_SAMPLE_SIZE;
srs_error("invalid avc sample length=%d, ret=%d", size, ret);
return ret;
}
// 5bits, 7.3.1 NAL unit syntax,
// H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
u_int8_t nal_unit_type;
nal_unit_type = *buf->bytes;
nal_unit_type &= 0x1f;
// Table 7-1 – NAL unit type codes, page 61
// 1: Coded slice
if (nal_unit_type == 1) {
sps_pps_sent = false;
}
// 5: Coded slice of an IDR picture.
// insert sps/pps before IDR or key frame is ok.
if (nal_unit_type == 5 && !sps_pps_sent) {
//if (vf->key && !sps_pps_sent) {
sps_pps_sent = true;
// ngx_rtmp_hls_append_sps_pps
if (codec->sequenceParameterSetLength > 0) {
// AnnexB prefix
vb->append(aud_nal, 4);
// sps
vb->append(codec->sequenceParameterSetNALUnit, codec->sequenceParameterSetLength);
}
if (codec->pictureParameterSetLength > 0) {
// AnnexB prefix
vb->append(aud_nal, 4);
// pps
vb->append(codec->pictureParameterSetNALUnit, codec->pictureParameterSetLength);
}
}
// sample start prefix, '00 00 00 01' or '00 00 01'
u_int8_t* p = aud_nal + 1;
u_int8_t* end = p + 3;
// first AnnexB prefix is long (4 bytes)
if (i == 0) {
p = aud_nal;
}
vb->append(p, end - p);
// sample data
vb->append(buf->bytes, buf->size);
}
return ret;
}

49
trunk/src/core/srs_core_hls.hpp
View file

@ -159,6 +159,43 @@ private:
virtual int create_dir();
};
/**
* ts need to cache some audio then flush
*/
class SrsTSCache
{
private:
// current frame and buffer
SrsMpegtsFrame* af;
SrsCodecBuffer* ab;
SrsMpegtsFrame* vf;
SrsCodecBuffer* vb;
private:
// the audio cache buffer start pts, to flush audio if full.
int64_t audio_buffer_start_pts;
// time jitter for aac
SrsHlsAacJitter* aac_jitter;
public:
SrsTSCache();
virtual ~SrsTSCache();
public:
/**
* write audio to cache, if need to flush, flush to muxer.
*/
virtual int write_audio(SrsCodec* codec, SrsM3u8Muxer* muxer, int64_t pts, SrsCodecSample* sample);
/**
* write video to muxer.
*/
virtual int write_video(SrsCodec* codec, SrsM3u8Muxer* muxer, int64_t dts, SrsCodecSample* sample);
/**
* flush audio in cache to muxer.
*/
virtual int flush_audio(SrsM3u8Muxer* muxer);
private:
virtual int cache_audio(SrsCodec* codec, SrsCodecSample* sample);
virtual int cache_video(SrsCodec* codec, SrsCodecSample* sample);
};
/**
* write m3u8 hls.
*/
@ -166,19 +203,12 @@ class SrsHls
{
private:
SrsM3u8Muxer* muxer;
// current frame and buffer
SrsMpegtsFrame* af;
SrsCodecBuffer* ab;
SrsMpegtsFrame* vf;
SrsCodecBuffer* vb;
// the audio cache buffer start pts, to flush audio if full.
int64_t audio_buffer_start_pts;
SrsTSCache* ts_cache;
private:
bool hls_enabled;
SrsCodec* codec;
SrsCodecSample* sample;
SrsRtmpJitter* jitter;
SrsHlsAacJitter* aac_jitter;
public:
SrsHls();
virtual ~SrsHls();
@ -188,9 +218,6 @@ public:
virtual int on_meta_data(SrsOnMetaDataPacket* metadata);
virtual int on_audio(SrsSharedPtrMessage* audio);
virtual int on_video(SrsSharedPtrMessage* video);
private:
virtual int cache_audio();
virtual int cache_video();
};
#endif