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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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This commit is contained in:
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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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;