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for #301, http ts stream support h.264+mp3. 2.0.106
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winlin
parent
aaade0f04f
commit
2c42350489
10 changed files with 305 additions and 75 deletions
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@ -53,6 +53,7 @@ using namespace std;
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// ts aac stream id.
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#define TS_AUDIO_AAC 0xc0
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#define TS_AUDIO_MP3 0x04
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// ts avc stream id.
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#define TS_VIDEO_AVC 0xe0
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@ -118,11 +119,18 @@ u_int8_t mpegts_header[] = {
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// must generate header with/without video, @see:
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// https://github.com/winlinvip/simple-rtmp-server/issues/40
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0x1b, 0xe1, 0x00, 0xf0, 0x00, /* h264, pid=0x100=256 */
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};
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u_int8_t mpegts_header_aac[] = {
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0x0f, 0xe1, 0x01, 0xf0, 0x00, /* aac, pid=0x101=257 */
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/*0x03, 0xe1, 0x01, 0xf0, 0x00,*/ /* mp3 */
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/* CRC */
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0x2f, 0x44, 0xb9, 0x9b, /* crc for aac */
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/*0x4e, 0x59, 0x3d, 0x1e,*/ /* crc for mp3 */
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};
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u_int8_t mpegts_header_mp3[] = {
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0x03, 0xe1, 0x01, 0xf0, 0x00, /* mp3 */
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/* CRC */
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0x4e, 0x59, 0x3d, 0x1e, /* crc for mp3 */
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};
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u_int8_t mpegts_header_padding[] = {
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/* stuffing 157 bytes */
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0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
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0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff, 0xff,
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@ -147,7 +155,7 @@ u_int8_t mpegts_header[] = {
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class SrsMpegtsWriter
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{
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public:
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static int write_header(SrsFileWriter* writer)
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static int write_header(SrsFileWriter* writer, SrsCodecAudio acodec)
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{
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int ret = ERROR_SUCCESS;
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@ -157,6 +165,26 @@ public:
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return ret;
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}
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if (acodec == SrsCodecAudioAAC) {
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if ((ret = writer->write(mpegts_header_aac, sizeof(mpegts_header_aac), NULL)) != ERROR_SUCCESS) {
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ret = ERROR_HLS_WRITE_FAILED;
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srs_error("write ts file aac header failed. ret=%d", ret);
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return ret;
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}
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} else {
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if ((ret = writer->write(mpegts_header_mp3, sizeof(mpegts_header_mp3), NULL)) != ERROR_SUCCESS) {
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ret = ERROR_HLS_WRITE_FAILED;
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srs_error("write ts file mp3 header failed. ret=%d", ret);
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return ret;
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}
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}
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if ((ret = writer->write(mpegts_header_padding, sizeof(mpegts_header_padding), NULL)) != ERROR_SUCCESS) {
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ret = ERROR_HLS_WRITE_FAILED;
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srs_error("write ts file padding header failed. ret=%d", ret);
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return ret;
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}
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return ret;
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}
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static int write_frame(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* buffer)
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@ -375,6 +403,11 @@ SrsMpegtsFrame::SrsMpegtsFrame()
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SrsTSMuxer::SrsTSMuxer(SrsFileWriter* w)
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{
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writer = w;
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// reserved is not written.
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previous = SrsCodecAudioReserved1;
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// current default to aac.
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current = SrsCodecAudioAAC;
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}
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SrsTSMuxer::~SrsTSMuxer()
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@ -393,12 +426,19 @@ int SrsTSMuxer::open(string _path)
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if ((ret = writer->open(path)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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// write mpegts header
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if ((ret = SrsMpegtsWriter::write_header(writer)) != ERROR_SUCCESS) {
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int SrsTSMuxer::update_acodec(SrsCodecAudio acodec)
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{
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int ret = ERROR_SUCCESS;
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if (current == acodec) {
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return ret;
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}
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current = acodec;
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return ret;
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}
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@ -406,6 +446,14 @@ int SrsTSMuxer::write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
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{
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int ret = ERROR_SUCCESS;
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// when acodec changed, write header.
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if (current != previous) {
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previous = current;
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if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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if ((ret = SrsMpegtsWriter::write_frame(writer, af, ab)) != ERROR_SUCCESS) {
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return ret;
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}
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@ -417,6 +465,14 @@ int SrsTSMuxer::write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb)
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{
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int ret = ERROR_SUCCESS;
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// when acodec changed, write header.
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if (current != previous) {
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previous = current;
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if ((ret = SrsMpegtsWriter::write_header(writer, previous)) != ERROR_SUCCESS) {
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return ret;
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}
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}
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if ((ret = SrsMpegtsWriter::write_frame(writer, vf, vb)) != ERROR_SUCCESS) {
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return ret;
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}
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@ -501,6 +557,8 @@ SrsTsCache::SrsTsCache()
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af = new SrsMpegtsFrame();
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vf = new SrsMpegtsFrame();
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audio_buffer_start_pts = 0;
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}
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SrsTsCache::~SrsTsCache()
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@ -520,23 +578,53 @@ SrsTsCache::~SrsTsCache()
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int SrsTsCache::cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample)
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{
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int ret = ERROR_SUCCESS;
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// start buffer, set the af
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// @remark, always use the orignal pts.
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if (ab->length() == 0) {
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pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
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af->dts = af->pts = pts;
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af->pid = TS_AUDIO_PID;
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af->sid = TS_AUDIO_AAC;
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} else {
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aac_jitter->on_buffer_continue();
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audio_buffer_start_pts = pts;
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}
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// write audio to cache.
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if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
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// must be aac or mp3
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SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
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srs_assert(acodec == SrsCodecAudioAAC || acodec == SrsCodecAudioMP3);
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// cache the aac audio.
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if (codec->audio_codec_id == SrsCodecAudioAAC) {
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// for aac audio, recalc the timestamp by aac jitter.
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if (ab->length() == 0) {
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pts = aac_jitter->on_buffer_start(pts, sample->sound_rate, codec->aac_sample_rate);
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af->dts = af->pts = pts;
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af->pid = TS_AUDIO_PID;
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af->sid = TS_AUDIO_AAC;
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} else {
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aac_jitter->on_buffer_continue();
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}
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// write aac audio to cache.
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if ((ret = do_cache_audio(codec, sample)) != ERROR_SUCCESS) {
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return ret;
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}
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return ret;
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}
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// cache the mp3 audio.
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if (codec->audio_codec_id == SrsCodecAudioMP3) {
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// for mp3 audio, recalc the timestamp by mp3 jitter.
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// TODO: FIXME: implements it.
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af->dts = af->pts = pts;
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af->pid = TS_AUDIO_PID;
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af->sid = SrsCodecAudioMP3;
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// for mp3, directly write to cache.
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// TODO: FIXME: implements it.
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for (int i = 0; i < sample->nb_sample_units; i++) {
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SrsCodecSampleUnit* sample_unit = &sample->sample_units[i];
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ab->append(sample_unit->bytes, sample_unit->size);
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}
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}
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return ret;
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}
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@ -784,16 +872,30 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
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sample->clear();
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if ((ret = codec->audio_aac_demux(data, size, sample)) != ERROR_SUCCESS) {
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srs_error("http: ts codec demux audio failed. ret=%d", ret);
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if (ret != ERROR_HLS_TRY_MP3) {
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srs_error("http: ts aac demux audio failed. ret=%d", ret);
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return ret;
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}
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if ((ret = codec->audio_mp3_demux(data, size, sample)) != ERROR_SUCCESS) {
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srs_error("http: ts mp3 demux audio failed. ret=%d", ret);
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return ret;
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}
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}
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SrsCodecAudio acodec = (SrsCodecAudio)codec->audio_codec_id;
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// ts support audio codec: aac/mp3
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if (acodec != SrsCodecAudioAAC && acodec != SrsCodecAudioMP3) {
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return ret;
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}
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// when codec changed, write new header.
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if ((ret = muxer->update_acodec(acodec)) != ERROR_SUCCESS) {
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srs_error("http: ts audio write header failed. ret=%d", ret);
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return ret;
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}
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if (codec->audio_codec_id != SrsCodecAudioAAC) {
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return ret;
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}
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// ignore sequence header
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if (sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
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// for aac: ignore sequence header
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if (acodec == SrsCodecAudioAAC && sample->aac_packet_type == SrsCodecAudioTypeSequenceHeader) {
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return ret;
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}
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@ -809,12 +911,15 @@ int SrsTsEncoder::write_audio(int64_t timestamp, char* data, int size)
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// flush if buffer exceed max size.
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if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
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if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
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return ret;
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}
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// write success, clear and free the buffer
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cache->ab->erase(cache->ab->length());
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return flush_video();
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}
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// TODO: config it.
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// in ms, audio delay to flush the audios.
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int64_t audio_delay = SRS_CONF_DEFAULT_AAC_DELAY;
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// flush if audio delay exceed
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if (dts - cache->audio_buffer_start_pts > audio_delay * 90) {
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return flush_audio();
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}
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return ret;
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@ -852,6 +957,27 @@ int SrsTsEncoder::write_video(int64_t timestamp, char* data, int size)
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if ((ret = cache->cache_video(codec, dts, sample)) != ERROR_SUCCESS) {
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return ret;
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}
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return flush_video();
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}
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int SrsTsEncoder::flush_audio()
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{
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int ret = ERROR_SUCCESS;
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if ((ret = muxer->write_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
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return ret;
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}
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// write success, clear and free the buffer
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cache->ab->erase(cache->ab->length());
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return ret;
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}
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int SrsTsEncoder::flush_video()
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{
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int ret = ERROR_SUCCESS;
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if ((ret = muxer->write_video(cache->vf, cache->vb)) != ERROR_SUCCESS) {
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return ret;
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