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Code Issues Releases Wiki Activity

for #301, http ts stream support h.264+mp3. 2.0.106

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winlin 2015-01-25 16:42:22 +08:00
parent aaade0f04f
commit 2c42350489
10 changed files with 305 additions and 75 deletions
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1
README.md
View file

@ -521,6 +521,7 @@ Supported operating systems and hardware:
### SRS 2.0 history
* v2.0, 2015-01-25, for [#301](https://github.com/winlinvip/simple-rtmp-server/issues/301), http ts stream support h.264+mp3. 2.0.106
* v2.0, 2015-01-25, hotfix [#268](https://github.com/winlinvip/simple-rtmp-server/issues/268), refine the pcr start at 0, dts/pts plus delay. 2.0.105
* v2.0, 2015-01-25, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), refine pcr=dts-800ms and use dts/pts directly. 2.0.104
* v2.0, 2015-01-23, hotfix [#151](https://github.com/winlinvip/simple-rtmp-server/issues/151), use absolutely overflow to make jwplayer happy. 2.0.103

31
trunk/src/app/srs_app_hls.cpp
View file

@ -199,6 +199,13 @@ bool SrsHlsMuxer::is_segment_absolutely_overflow()
return current->duration >= 2 * hls_fragment;
}
int SrsHlsMuxer::update_acodec(SrsCodecAudio acodec)
{
srs_assert(current);
srs_assert(current->muxer);
return current->muxer->update_acodec(acodec);
}
int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
{
int ret = ERROR_SUCCESS;
@ -572,8 +579,6 @@ int SrsHlsCache::on_sequence_header(SrsHlsMuxer* muxer)
int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t pts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
audio_buffer_start_pts = pts;
// write audio to cache.
if ((ret = cache->cache_audio(codec, pts, sample)) != ERROR_SUCCESS) {
@ -591,7 +596,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// 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 (pts - cache->audio_buffer_start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
return ret;
}
@ -773,11 +778,25 @@ int SrsHls::on_audio(SrsSharedPtrMessage* __audio)
sample->clear();
if ((ret = codec->audio_aac_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
srs_error("hls codec demux audio failed. ret=%d", ret);
if (ret != ERROR_HLS_TRY_MP3) {
srs_error("hls aac demux audio failed. ret=%d", ret);
return ret;
}
if ((ret = codec->audio_mp3_demux(audio->payload, audio->size, sample)) != ERROR_SUCCESS) {
srs_error("hls mp3 demux audio failed. ret=%d", ret);
return ret;
}
}
SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
// ts support audio codec: aac/mp3
if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
return ret;
}
if (codec->audio_codec_id != SrsCodecAudioAAC) {
// when codec changed, write new header.
if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
srs_error("http: ts audio write header failed. ret=%d", ret);
return ret;
}

6
trunk/src/app/srs_app_hls.hpp
View file

@ -37,6 +37,8 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <string>
#include <vector>
#include <srs_kernel_codec.hpp>
class SrsSharedPtrMessage;
class SrsCodecSample;
class SrsMpegtsFrame;
@ -141,6 +143,8 @@ public:
* @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
*/
virtual bool is_segment_absolutely_overflow();
public:
virtual int update_acodec(SrsCodecAudio acodec);
virtual int flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
virtual int flush_video(SrsMpegtsFrame* af, SrsSimpleBuffer* ab, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
/**
@ -174,8 +178,6 @@ private:
class SrsHlsCache
{
private:
// the audio cache buffer start pts, to flush audio if full.
int64_t audio_buffer_start_pts;
SrsTsCache* cache;
public:
SrsHlsCache();

27
trunk/src/app/srs_app_source.cpp
View file

@ -1349,12 +1349,17 @@ int SrsSource::on_audio(SrsCommonMessage* __audio)
}
}
// cache the sequence header if h264
// donot cache the sequence header to gop_cache, return here.
if (SrsFlvCodec::audio_is_sequence_header(msg.payload, msg.size)) {
// cache the sequence header of aac, or first packet of mp3.
// for example, the mp3 is used for hls to write the "right" audio codec.
bool is_aac_sequence_header = SrsFlvCodec::audio_is_sequence_header(msg.payload, msg.size);
if (is_aac_sequence_header || !cache_sh_audio) {
srs_freep(cache_sh_audio);
cache_sh_audio = msg.copy();
}
// cache the sequence header if aac
// donot cache the sequence header to gop_cache, return here.
if (is_aac_sequence_header) {
// parse detail audio codec
SrsAvcAacCodec codec;
SrsCodecSample sample;
@ -1768,18 +1773,20 @@ int SrsSource::create_consumer(SrsConsumer*& consumer, bool ds, bool dm, bool dg
srs_info("dispatch metadata success");
// copy sequence header
// copy audio sequence first, for hls to fast parse the "right" audio codec.
// @see https://github.com/winlinvip/simple-rtmp-server/issues/301
if (ds && cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
srs_error("dispatch audio sequence header failed. ret=%d", ret);
return ret;
}
srs_info("dispatch audio sequence header success");
if (ds && cache_sh_video && (ret = consumer->enqueue(cache_sh_video, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
srs_error("dispatch video sequence header failed. ret=%d", ret);
return ret;
}
srs_info("dispatch video sequence header success");
if (cache_sh_audio && (ret = consumer->enqueue(cache_sh_audio, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
srs_error("dispatch audio sequence header failed. ret=%d", ret);
return ret;
}
srs_info("dispatch audio sequence header success");
// copy gop cache to client.
if (dg && (ret = gop_cache->dump(consumer, atc, tba, tbv, ag)) != ERROR_SUCCESS) {
return ret;

2
trunk/src/core/srs_core.hpp
View file

@ -31,7 +31,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
// current release version
#define VERSION_MAJOR 2
#define VERSION_MINOR 0
#define VERSION_REVISION 105
#define VERSION_REVISION 106
// server info.
#define RTMP_SIG_SRS_KEY "SRS"

89
trunk/src/kernel/srs_kernel_avc.cpp
View file

@ -60,6 +60,7 @@ void SrsCodecSample::clear()
frame_type = SrsCodecVideoAVCFrameReserved;
avc_packet_type = SrsCodecVideoAVCTypeReserved;
acodec = SrsCodecAudioReserved1;
sound_rate = SrsCodecAudioSampleRateReserved;
sound_size = SrsCodecAudioSampleSizeReserved;
sound_type = SrsCodecAudioSoundTypeReserved;
@ -91,10 +92,13 @@ SrsAvcAacCodec::SrsAvcAacCodec()
duration = 0;
NAL_unit_length = 0;
frame_rate = 0;
video_data_rate = 0;
video_codec_id = 0;
audio_data_rate = 0;
audio_codec_id = 0;
avc_profile = 0;
avc_level = 0;
aac_profile = 0;
@ -104,6 +108,7 @@ SrsAvcAacCodec::SrsAvcAacCodec()
avc_extra_data = NULL;
aac_extra_size = 0;
aac_extra_data = NULL;
sequenceParameterSetLength = 0;
sequenceParameterSetNALUnit = NULL;
pictureParameterSetLength = 0;
@ -129,7 +134,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
sample->is_video = false;
if (!data || size <= 0) {
srs_trace("no audio present, hls ignore it.");
srs_trace("no audio present, ignore it.");
return ret;
}
@ -140,7 +145,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
// audio decode
if (!stream->require(1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode audio sound_format failed. ret=%d", ret);
srs_error("audio codec decode sound_format failed. ret=%d", ret);
return ret;
}
@ -153,20 +158,27 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
sound_format = (sound_format >> 4) & 0x0f;
audio_codec_id = sound_format;
sample->acodec = (SrsCodecAudio)audio_codec_id;
sample->sound_type = (SrsCodecAudioSoundType)sound_type;
sample->sound_rate = (SrsCodecAudioSampleRate)sound_rate;
sample->sound_size = (SrsCodecAudioSampleSize)sound_size;
// we support h.264+mp3 for hls.
if (audio_codec_id == SrsCodecAudioMP3) {
return ERROR_HLS_TRY_MP3;
}
// only support aac
if (audio_codec_id != SrsCodecAudioAAC) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls only support audio aac codec. actual=%d, ret=%d", audio_codec_id, ret);
srs_error("audio codec only support mp3/aac codec. actual=%d, ret=%d", audio_codec_id, ret);
return ret;
}
if (!stream->require(1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode audio aac_packet_type failed. ret=%d", ret);
srs_error("audio codec decode aac_packet_type failed. ret=%d", ret);
return ret;
}
@ -189,7 +201,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
// channelConfiguration, aac_channels, 4bits
if (!stream->require(2)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode audio aac sequence header failed. ret=%d", ret);
srs_error("audio codec decode aac sequence header failed. ret=%d", ret);
return ret;
}
aac_profile = stream->read_1bytes();
@ -201,7 +213,7 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
if (aac_profile == 0 || aac_profile == 0x1f) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode audio aac sequence header failed, "
srs_error("audio codec decode aac sequence header failed, "
"adts object=%d invalid. ret=%d", aac_profile, ret);
return ret;
}
@ -221,14 +233,14 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
// ensure the sequence header demuxed
if (aac_extra_size <= 0 || !aac_extra_data) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode audio aac failed, sequence header not found. ret=%d", ret);
srs_error("audio codec decode aac failed, sequence header not found. ret=%d", ret);
return ret;
}
// Raw AAC frame data in UI8 []
// 6.3 Raw Data, aac-iso-13818-7.pdf, page 28
if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), stream->size() - stream->pos())) != ERROR_SUCCESS) {
srs_error("hls add audio sample failed. ret=%d", ret);
srs_error("audio codec add sample failed. ret=%d", ret);
return ret;
}
} else {
@ -264,6 +276,31 @@ int SrsAvcAacCodec::audio_aac_demux(char* data, int size, SrsCodecSample* sample
return ret;
}
int SrsAvcAacCodec::audio_mp3_demux(char* data, int size, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// we always decode aac then mp3.
srs_assert(sample->acodec == SrsCodecAudioMP3);
// @see: E.4.2 Audio Tags, video_file_format_spec_v10_1.pdf, page 76
if (!data || size <= 1) {
srs_trace("no mp3 audio present, ignore it.");
return ret;
}
// mp3 payload.
if ((ret = sample->add_sample_unit(data + 1, size - 1)) != ERROR_SUCCESS) {
srs_error("audio codec add mp3 sample failed. ret=%d", ret);
return ret;
}
srs_info("audio decoded, type=%d, codec=%d, asize=%d, rate=%d, format=%d, size=%d",
sample->sound_type, audio_codec_id, sample->sound_size, sample->sound_rate, sample->acodec, size);
return ret;
}
int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
@ -271,7 +308,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
sample->is_video = true;
if (!data || size <= 0) {
srs_trace("no video present, hls ignore it.");
srs_trace("no video present, ignore it.");
return ret;
}
@ -282,7 +319,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
// video decode
if (!stream->require(1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video frame_type failed. ret=%d", ret);
srs_error("video codec decode frame_type failed. ret=%d", ret);
return ret;
}
@ -296,21 +333,21 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
// ignore info frame without error,
// @see https://github.com/winlinvip/simple-rtmp-server/issues/288#issuecomment-69863909
if (sample->frame_type == SrsCodecVideoAVCFrameVideoInfoFrame) {
srs_warn("hls igone the info frame, ret=%d", ret);
srs_warn("video codec igone the info frame, ret=%d", ret);
return ret;
}
// only support h.264/avc
if (codec_id != SrsCodecVideoAVC) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls only support video h.264/avc codec. actual=%d, ret=%d", codec_id, ret);
srs_error("video codec only support video h.264/avc codec. actual=%d, ret=%d", codec_id, ret);
return ret;
}
video_codec_id = codec_id;
if (!stream->require(4)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc_packet_type failed. ret=%d", ret);
srs_error("video codec decode avc_packet_type failed. ret=%d", ret);
return ret;
}
int8_t avc_packet_type = stream->read_1bytes();
@ -328,7 +365,7 @@ int SrsAvcAacCodec::video_avc_demux(char* data, int size, SrsCodecSample* sample
// ensure the sequence header demuxed
if (avc_extra_size <= 0 || !avc_extra_data) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc failed, sequence header not found. ret=%d", ret);
srs_error("avc decode failed, sequence header not found. ret=%d", ret);
return ret;
}
@ -371,7 +408,7 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
if (!stream->require(6)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header failed. ret=%d", ret);
srs_error("avc decode sequenc header failed. ret=%d", ret);
return ret;
}
//int8_t configurationVersion = stream->read_1bytes();
@ -402,25 +439,25 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
// 1 sps
if (!stream->require(1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header sps failed. ret=%d", ret);
srs_error("avc decode sequenc header sps failed. ret=%d", ret);
return ret;
}
int8_t numOfSequenceParameterSets = stream->read_1bytes();
numOfSequenceParameterSets &= 0x1f;
if (numOfSequenceParameterSets != 1) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header sps failed. ret=%d", ret);
srs_error("avc decode sequenc header sps failed. ret=%d", ret);
return ret;
}
if (!stream->require(2)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header sps size failed. ret=%d", ret);
srs_error("avc decode sequenc header sps size failed. ret=%d", ret);
return ret;
}
sequenceParameterSetLength = stream->read_2bytes();
if (!stream->require(sequenceParameterSetLength)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header sps data failed. ret=%d", ret);
srs_error("avc decode sequenc header sps data failed. ret=%d", ret);
return ret;
}
if (sequenceParameterSetLength > 0) {
@ -432,25 +469,25 @@ int SrsAvcAacCodec::avc_demux_sps_pps(SrsStream* stream)
// 1 pps
if (!stream->require(1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header pps failed. ret=%d", ret);
srs_error("avc decode sequenc header pps failed. ret=%d", ret);
return ret;
}
int8_t numOfPictureParameterSets = stream->read_1bytes();
numOfPictureParameterSets &= 0x1f;
if (numOfPictureParameterSets != 1) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header pps failed. ret=%d", ret);
srs_error("avc decode sequenc header pps failed. ret=%d", ret);
return ret;
}
if (!stream->require(2)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header pps size failed. ret=%d", ret);
srs_error("avc decode sequenc header pps size failed. ret=%d", ret);
return ret;
}
pictureParameterSetLength = stream->read_2bytes();
if (!stream->require(pictureParameterSetLength)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc sequenc header pps data failed. ret=%d", ret);
srs_error("avc decode sequenc header pps data failed. ret=%d", ret);
return ret;
}
if (pictureParameterSetLength > 0) {
@ -534,7 +571,7 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsStream* stream, SrsCodecSample* sam
// unsigned int((NAL_unit_length+1)*8) NALUnitLength;
if (!stream->require(NAL_unit_length + 1)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc NALU size failed. ret=%d", ret);
srs_error("avc decode NALU size failed. ret=%d", ret);
return ret;
}
int32_t NALUnitLength = 0;
@ -557,12 +594,12 @@ int SrsAvcAacCodec::avc_demux_ibmf_format(SrsStream* stream, SrsCodecSample* sam
// NALUnit
if (!stream->require(NALUnitLength)) {
ret = ERROR_HLS_DECODE_ERROR;
srs_error("hls decode video avc NALU data failed. ret=%d", ret);
srs_error("avc decode NALU data failed. ret=%d", ret);
return ret;
}
// 7.3.1 NAL unit syntax, H.264-AVC-ISO_IEC_14496-10.pdf, page 44.
if ((ret = sample->add_sample_unit(stream->data() + stream->pos(), NALUnitLength)) != ERROR_SUCCESS) {
srs_error("hls add video sample failed. ret=%d", ret);
srs_error("avc add video sample failed. ret=%d", ret);
return ret;
}
stream->skip(NALUnitLength);

3
trunk/src/kernel/srs_kernel_avc.hpp
View file

@ -156,6 +156,8 @@ public:
SrsCodecVideoAVCType avc_packet_type;
public:
// audio specified
SrsCodecAudio acodec;
// audio aac specified.
SrsCodecAudioSampleRate sound_rate;
SrsCodecAudioSampleSize sound_size;
SrsCodecAudioSoundType sound_type;
@ -271,6 +273,7 @@ public:
* demux the aac raw to sample units.
*/
virtual int audio_aac_demux(char* data, int size, SrsCodecSample* sample);
virtual int audio_mp3_demux(char* data, int size, SrsCodecSample* sample);
/**
* demux the video packet in h.264 codec.
* the packet mux in FLV/RTMP format defined in flv specification.

1
trunk/src/kernel/srs_kernel_error.hpp
View file

@ -202,6 +202,7 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#define ERROR_AAC_REQUIRED_ADTS 3046
#define ERROR_AAC_ADTS_HEADER 3047
#define ERROR_AAC_DATA_INVALID 3048
#define ERROR_HLS_TRY_MP3 3049
///////////////////////////////////////////////////////
// HTTP/StreamCaster protocol error.

186
trunk/src/kernel/srs_kernel_ts.cpp
View file

@ -53,6 +53,7 @@ using namespace std;
// ts aac stream id.
#define TS_AUDIO_AAC 0xc0
#define TS_AUDIO_MP3 0x04
// ts avc stream id.
#define TS_VIDEO_AVC 0xe0
@ -118,11 +119,18 @@ u_int8_t mpegts_header[] = {
// must generate header with/without video, @see:
// https://github.com/winlinvip/simple-rtmp-server/issues/40
0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
};
u_int8_t mpegts_header_aac[] = {
0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
/*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
/* CRC */
0x2f, 0x44, 0xb9, 0x9b, /* crc for aac */
/*0x4e, 0x59, 0x3d, 0x1e,*/ /* crc for mp3 */
};
u_int8_t mpegts_header_mp3[] = {
0x03, 0xe1, 0x01, 0xf0, 0x00, /* mp3 */
/* CRC */
0x4e, 0x59, 0x3d, 0x1e, /* crc for mp3 */
};
u_int8_t mpegts_header_padding[] = {
/* stuffing 157 bytes */
0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
@ -147,7 +155,7 @@ u_int8_t mpegts_header[] = {
class SrsMpegtsWriter
{
public:
static int write_header(SrsFileWriter* writer)
static int write_header(SrsFileWriter* writer, SrsCodecAudio acodec)
{
int ret = ERROR_SUCCESS;
@ -157,6 +165,26 @@ public:
return ret;
}
if (acodec == SrsCodecAudioAAC) {
if ((ret = writer->write(mpegts_header_aac, sizeof(mpegts_header_aac), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file aac header failed. ret=%d", ret);
return ret;
}
} else {
if ((ret = writer->write(mpegts_header_mp3, sizeof(mpegts_header_mp3), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file mp3 header failed. ret=%d", ret);
return ret;
}
}
if ((ret = writer->write(mpegts_header_padding, sizeof(mpegts_header_padding), NULL)) != ERROR_SUCCESS) {
ret = ERROR_HLS_WRITE_FAILED;
srs_error("write ts file padding header failed. ret=%d", ret);
return ret;
}
return ret;
}
static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* buffer)
@ -375,6 +403,11 @@ SrsMpegtsFrame::SrsMpegtsFrame()
SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
{
writer = w;
// reserved is not written.
previous = SrsCodecAudioReserved1;
// current default to aac.
current = SrsCodecAudioAAC;
}
SrsTSMuxer::~SrsTSMuxer()
@ -393,12 +426,19 @@ int SrsTSMuxer::open(string _path)
if ((ret = writer->open(path)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
// write mpegts header
if ((ret = SrsMpegtsWriter::write_header(writer)) != ERROR_SUCCESS) {
int SrsTSMuxer::update_acodec(SrsCodecAudio acodec)
{
int ret = ERROR_SUCCESS;
if (current == acodec) {
return ret;
}
current = acodec;
return ret;
}
@ -406,6 +446,14 @@ int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
{
int ret = ERROR_SUCCESS;
// when acodec changed, write header.
if (current != previous) {
previous = current;
if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
return ret;
}
}
if ((ret = SrsMpegtsWriter::write_frame(writer, af, ab)) != ERROR_SUCCESS) {
return ret;
}
@ -417,6 +465,14 @@ int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb)
{
int ret = ERROR_SUCCESS;
// when acodec changed, write header.
if (current != previous) {
previous = current;
if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
return ret;
}
}
if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
return ret;
}
@ -501,6 +557,8 @@ SrsTsCache::SrsTsCache()
af = new SrsMpegtsFrame();
vf = new SrsMpegtsFrame();
audio_buffer_start_pts = 0;
}
SrsTsCache::~SrsTsCache()
@ -520,23 +578,53 @@ SrsTsCache::~SrsTsCache()
int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample)
{
int ret = ERROR_SUCCESS;
// start buffer, set the af
// @remark, always use the orignal pts.
if (ab->length() == 0) {
pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
af->dts = af->pts = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
audio_buffer_start_pts = pts;
}
// write audio to cache.
if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
// must be aac or mp3
SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
srs_assert(acodec == SrsCodecAudioAAC || acodec == SrsCodecAudioMP3);
// cache the aac audio.
if (codec->audio_codec_id == SrsCodecAudioAAC) {
// for aac audio, recalc the timestamp by aac jitter.
if (ab->length() == 0) {
pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
af->dts = af->pts = pts;
af->pid = TS_AUDIO_PID;
af->sid = TS_AUDIO_AAC;
} else {
aac_jitter->on_buffer_continue();
}
// write aac audio to cache.
if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
// cache the mp3 audio.
if (codec->audio_codec_id == SrsCodecAudioMP3) {
// for mp3 audio, recalc the timestamp by mp3 jitter.
// TODO: FIXME: implements it.
af->dts = af->pts = pts;
af->pid = TS_AUDIO_PID;
af->sid = SrsCodecAudioMP3;
// for mp3, directly write to cache.
// TODO: FIXME: implements it.
for (int i = 0; i < sample->nb_sample_units; i++) {
SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
ab->append(sample_unit->bytes, sample_unit->size);
}
}
return ret;
}
@ -784,16 +872,30 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
sample->clear();
if ((ret = codec->audio_aac_demux(data, size, sample)) != ERROR_SUCCESS) {
srs_error("http: ts codec demux audio failed. ret=%d", ret);
if (ret != ERROR_HLS_TRY_MP3) {
srs_error("http: ts aac demux audio failed. ret=%d", ret);
return ret;
}
if ((ret = codec->audio_mp3_demux(data, size, sample)) != ERROR_SUCCESS) {
srs_error("http: ts mp3 demux audio failed. ret=%d", ret);
return ret;
}
}
SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
// ts support audio codec: aac/mp3
if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
return ret;
}
// when codec changed, write new header.
if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
srs_error("http: ts audio write header failed. ret=%d", ret);
return ret;
}
if (codec->audio_codec_id != SrsCodecAudioAAC) {
return ret;
}
// ignore sequence header
if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
// for aac: ignore sequence header
if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
return ret;
}
@ -809,12 +911,15 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
// flush if buffer exceed max size.
if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
return ret;
}
// write success, clear and free the buffer
cache->ab->erase(cache->ab->length());
return flush_video();
}
// 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 (dts - cache->audio_buffer_start_pts > audio_delay * 90) {
return flush_audio();
}
return ret;
@ -852,6 +957,27 @@ int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
return ret;
}
return flush_video();
}
int SrsTsEncoder::flush_audio()
{
int ret = ERROR_SUCCESS;
if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
return ret;
}
// write success, clear and free the buffer
cache->ab->erase(cache->ab->length());
return ret;
}
int SrsTsEncoder::flush_video()
{
int ret = ERROR_SUCCESS;
if ((ret = muxer->write_video(cache->vf, cache->vb)) != ERROR_SUCCESS) {
return ret;

34
trunk/src/kernel/srs_kernel_ts.hpp
View file

@ -31,6 +31,8 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
#include <string>
#include <srs_kernel_codec.hpp>
class SrsTsCache;
class SrsTSMuxer;
class SrsFileWriter;
@ -62,6 +64,9 @@ public:
*/
class SrsTSMuxer
{
private:
SrsCodecAudio previous;
SrsCodecAudio current;
private:
SrsFileWriter* writer;
std::string path;
@ -69,9 +74,31 @@ public:
SrsTSMuxer(SrsFileWriter* w);
virtual ~SrsTSMuxer();
public:
/**
* open the writer, donot write the PSI of ts.
*/
virtual int open(std::string _path);
/**
* when open ts, we donot write the header(PSI),
* for user may need to update the acodec to mp3 or others,
* so we use delay write PSI, when write audio or video.
* @remark for audio aac codec, for example, SRS1, it's ok to write PSI when open ts.
* @see https://github.com/winlinvip/simple-rtmp-server/issues/301
*/
virtual int update_acodec(SrsCodecAudio acodec);
/**
* write an audio frame to ts,
* @remark write PSI first when not write yet.
*/
virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
/**
* write a video frame to ts,
* @remark write PSI first when not write yet.
*/
virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
/**
* close the writer.
*/
virtual void close();
};
@ -125,6 +152,10 @@ public:
SrsSimpleBuffer* ab;
SrsMpegtsFrame* vf;
SrsSimpleBuffer* vb;
public:
// the audio cache buffer start pts, to flush audio if full.
// @remark the pts is not the adjust one, it's the orignal pts.
int64_t audio_buffer_start_pts;
protected:
// time jitter for aac
SrsTsAacJitter* aac_jitter;
@ -172,6 +203,9 @@ public:
*/
virtual int write_audio(int64_t timestamp, char* data, int size);
virtual int write_video(int64_t timestamp, char* data, int size);
private:
virtual int flush_audio();
virtual int flush_video();
};
#endif