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rewrite the ts remux of hls. 2.0.117

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winlin 2015-02-15 16:37:28 +08:00
parent 1ed3e283ab
commit 3c813847bd
5 changed files with 817 additions and 316 deletions
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54
trunk/src/app/srs_app_hls.cpp
View file

@ -313,7 +313,7 @@ int SrsHlsMuxer::update_acodec(SrsCodecAudio ac)
return current->muxer->update_acodec(ac);
}
int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
int SrsHlsMuxer::flush_audio(SrsTsCache* cache)
{
int ret = ERROR_SUCCESS;
@ -323,24 +323,24 @@ int SrsHlsMuxer::flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab)
return ret;
}
if (ab->length() <= 0) {
if (!cache->audio || cache->audio->payload->length() <= 0) {
return ret;
}
// update the duration of segment.
current->update_duration(af->pts);
current->update_duration(cache->audio->pts);
if ((ret = current->muxer->write_audio(af, ab)) != ERROR_SUCCESS) {
if ((ret = current->muxer->write_audio(cache->audio)) != ERROR_SUCCESS) {
return ret;
}
// write success, clear and free the buffer
ab->erase(ab->length());
// write success, clear and free the msg
srs_freep(cache->audio);
return ret;
}
int SrsHlsMuxer::flush_video(SrsMpegtsFrame* /*af*/, SrsSimpleBuffer* /*ab*/, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb)
int SrsHlsMuxer::flush_video(SrsTsCache* cache)
{
int ret = ERROR_SUCCESS;
@ -350,17 +350,21 @@ int SrsHlsMuxer::flush_video(SrsMpegtsFrame* /*af*/, SrsSimpleBuffer* /*ab*/, Sr
return ret;
}
srs_assert(current);
// update the duration of segment.
current->update_duration(vf->dts);
if ((ret = current->muxer->write_video(vf, vb)) != ERROR_SUCCESS) {
if (!cache->video || cache->video->payload->length() <= 0) {
return ret;
}
// write success, clear and free the buffer
vb->erase(vb->length());
srs_assert(current);
// update the duration of segment.
current->update_duration(cache->video->dts);
if ((ret = current->muxer->write_video(cache->video)) != ERROR_SUCCESS) {
return ret;
}
// write success, clear and free the msg
srs_freep(cache->video);
return ret;
}
@ -649,7 +653,7 @@ int SrsHlsCache::on_unpublish(SrsHlsMuxer* muxer)
{
int ret = ERROR_SUCCESS;
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
return ret;
}
@ -682,8 +686,8 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
}
// flush if buffer exceed max size.
if (cache->ab->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
if (cache->audio->payload->length() > SRS_AUTO_HLS_AUDIO_CACHE_SIZE) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
}
@ -692,8 +696,8 @@ 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 - cache->audio_buffer_start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
if (pts - cache->audio->start_pts > audio_delay * 90) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
return ret;
}
}
@ -707,7 +711,7 @@ int SrsHlsCache::write_audio(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// we use absolutely overflow of segment to make jwplayer/ffplay happy
// @see https://github.com/winlinvip/simple-rtmp-server/issues/151#issuecomment-71155184
if (muxer->is_segment_absolutely_overflow()) {
if ((ret = reap_segment("audio", muxer, cache->af->pts)) != ERROR_SUCCESS) {
if ((ret = reap_segment("audio", muxer, cache->audio->pts)) != ERROR_SUCCESS) {
return ret;
}
}
@ -728,14 +732,14 @@ int SrsHlsCache::write_video(SrsAvcAacCodec* codec, SrsHlsMuxer* muxer, int64_t
// 1. base on gop.
// 2. some gops duration overflow.
if (sample->frame_type == SrsCodecVideoAVCFrameKeyFrame && muxer->is_segment_overflow()) {
if ((ret = reap_segment("video", muxer, cache->vf->dts)) != ERROR_SUCCESS) {
if ((ret = reap_segment("video", muxer, cache->video->dts)) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
// flush video when got one
if ((ret = muxer->flush_video(cache->af, cache->ab, cache->vf, cache->vb)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_video(cache)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush video failed. ret=%d", ret);
return ret;
}
@ -761,7 +765,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
// TODO: fresh segment begin with audio or video?
// segment open, flush video first.
if ((ret = muxer->flush_video(cache->af, cache->ab, cache->vf, cache->vb)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_video(cache)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush video failed. ret=%d", ret);
return ret;
}
@ -769,7 +773,7 @@ int SrsHlsCache::reap_segment(string log_desc, SrsHlsMuxer* muxer, int64_t segme
// segment open, flush the audio.
// @see: ngx_rtmp_hls_open_fragment
/* start fragment with audio to make iPhone happy */
if ((ret = muxer->flush_audio(cache->af, cache->ab)) != ERROR_SUCCESS) {
if ((ret = muxer->flush_audio(cache)) != ERROR_SUCCESS) {
srs_error("m3u8 muxer flush audio failed. ret=%d", ret);
return ret;
}

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

@ -42,7 +42,6 @@ CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
class SrsSharedPtrMessage;
class SrsCodecSample;
class SrsMpegtsFrame;
class SrsAmf0Object;
class SrsRtmpJitter;
class SrsTSMuxer;
@ -55,6 +54,7 @@ class SrsSimpleBuffer;
class SrsTsAacJitter;
class SrsTsCache;
class SrsHlsSegment;
class SrsTsCache;
/**
* the handler for hls event.
@ -224,8 +224,8 @@ public:
virtual bool is_segment_absolutely_overflow();
public:
virtual int update_acodec(SrsCodecAudio ac);
virtual int flush_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
virtual int flush_video(SrsMpegtsFrame* af, SrsSimpleBuffer* ab, SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
virtual int flush_audio(SrsTsCache* cache);
virtual int flush_video(SrsTsCache* cache);
/**
* close segment(ts).
* @param log_desc the description for log.

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 116
#define VERSION_REVISION 117
// server info.
#define RTMP_SIG_SRS_KEY "SRS"

933
trunk/src/kernel/srs_kernel_ts.cpp
View file

File diff suppressed because it is too large Load diff

138
trunk/src/kernel/srs_kernel_ts.hpp
View file

@ -185,6 +185,8 @@ struct SrsTsChannel
SrsTsPidApply apply;
SrsTsStream stream;
SrsTsMessage* msg;
// for encoder.
u_int8_t continuity_counter;
SrsTsChannel();
virtual ~SrsTsChannel();
@ -209,13 +211,17 @@ enum SrsTsPESStreamId
// ISO/IEC 13818-3 or ISO/IEC 11172-3 or ISO/IEC 13818-7 or ISO/IEC
// 14496-3 audio stream number x xxxx
// ((sid >> 5) & 0x07) == SrsTsPESStreamIdAudio
SrsTsPESStreamIdAudio = 0x06, // 0b110
// @remark, use SrsTsPESStreamIdAudioCommon as actually audio, SrsTsPESStreamIdAudio to check whether audio.
SrsTsPESStreamIdAudioChecker = 0x06, // 0b110
SrsTsPESStreamIdAudioCommon = 0xc0,
// 1110 xxxx
// ITU-T Rec. H.262 | ISO/IEC 13818-2 or ISO/IEC 11172-2 or ISO/IEC
// 14496-2 video stream number xxxx
// ((stream_id >> 4) & 0x0f) == SrsTsPESStreamIdVideo
SrsTsPESStreamIdVideo = 0x0e, // 0b1110
// @remark, use SrsTsPESStreamIdVideoCommon as actually video, SrsTsPESStreamIdVideo to check whether video.
SrsTsPESStreamIdVideoChecker = 0x0e, // 0b1110
SrsTsPESStreamIdVideoCommon = 0xe0,
// ECM_stream
SrsTsPESStreamIdEcmStream = 0xf0, // 0b11110000
@ -253,8 +259,20 @@ enum SrsTsPESStreamId
class SrsTsMessage
{
public:
// decoder only,
// the ts messgae does not use them,
// for user to get the channel and packet.
SrsTsChannel* channel;
SrsTsPacket* packet;
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 start_pts;
// whether this message with pcr info,
// generally, the video IDR(I frame, the keyframe of h.264) carray the pcr info.
bool write_pcr;
// whether got discontinuity ts, for example, sequence header changed.
bool discontinuity;
public:
// the timestamp in 90khz
int64_t dts;
@ -269,8 +287,9 @@ public:
// the payload bytes.
SrsSimpleBuffer* payload;
public:
SrsTsMessage(SrsTsChannel* c, SrsTsPacket* p);
SrsTsMessage(SrsTsChannel* c = NULL, SrsTsPacket* p = NULL);
virtual ~SrsTsMessage();
// decoder
public:
/**
* dumps all bytes in stream to ts message.
@ -361,15 +380,14 @@ public:
public:
/**
* write the PES packet, the video/audio stream.
* @param frame the video/audio frame info.
* @param payload the video/audio payload bytes.
* @param msg the video/audio msg to write to ts.
* @param vc the video codec, write the PAT/PMT table when changed.
* @param ac the audio codec, write the PAT/PMT table when changed.
*/
virtual int encode(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* payload, SrsCodecVideo vc, SrsCodecAudio ac);
virtual int encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo vc, SrsCodecAudio ac);
private:
virtual int encode_pat_pmt(SrsFileWriter* writer, SrsCodecVideo vcodec, SrsCodecAudio acodec);
virtual int encode_pes(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* payload);
virtual int encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as);
virtual int encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid);
};
/**
@ -478,9 +496,22 @@ public:
public:
virtual int size();
virtual int encode(SrsStream* stream);
virtual void padding(int nb_stuffings);
public:
static SrsTsPacket* create_pat(SrsTsContext* context, int16_t pmt_number, int16_t pmt_pid);
static SrsTsPacket* create_pmt(SrsTsContext* context, int16_t pmt_number, int16_t pmt_pid, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as);
static SrsTsPacket* create_pat(SrsTsContext* context,
int16_t pmt_number, int16_t pmt_pid
);
static SrsTsPacket* create_pmt(SrsTsContext* context,
int16_t pmt_number, int16_t pmt_pid, int16_t vpid, SrsTsStream vs,
int16_t apid, SrsTsStream as
);
static SrsTsPacket* create_pes_first(SrsTsContext* context,
int16_t pid, SrsTsPESStreamId sid, u_int8_t continuity_counter, bool discontinuity,
int64_t pcr, int64_t dts, int64_t pts, int size
);
static SrsTsPacket* create_pes_continue(SrsTsContext* context,
int16_t pid, SrsTsPESStreamId sid, u_int8_t continuity_counter
);
};
/**
@ -627,7 +658,10 @@ public:
* the last bit of the program_clock_reference_base at the input of the system target decoder.
*/
int64_t program_clock_reference_base; //33bits
//6bits reserved.
/**
* 6bits reserved, must be '1'
*/
int8_t const1_value0; // 6bits
int16_t program_clock_reference_extension; //9bits
// if OPCR_flag, 6B
@ -646,7 +680,10 @@ public:
* in the original single program Transport Stream.
*/
int64_t original_program_clock_reference_base; //33bits
//6bits reserved.
/**
* 6bits reserved, must be '1'
*/
int8_t const1_value2; // 6bits
int16_t original_program_clock_reference_extension; //9bits
// if splicing_point_flag, 1B
@ -711,7 +748,10 @@ public:
* constraints indicated by the splice_type value.
*/
int8_t seamless_splice_flag; //1bit
//5bits reserved
/**
* reserved 5bits, must be '1'
*/
int8_t const1_value1; //5bits
// if ltw_flag, 2B
/**
* (legal time window_valid_flag) - This is a 1-bit field which when set to '1' indicates that the value of the
@ -856,6 +896,7 @@ public:
* elementary stream type as defined in Table 2-18. In Transport Streams, the elementary stream type is specified in the
* Program Specific Information as specified in 2.4.4.
*/
// @see SrsTsPESStreamId, value can be SrsTsPESStreamIdAudioCommon or SrsTsPESStreamIdVideoCommon.
u_int8_t stream_id; //8bits
// 2B
/**
@ -866,7 +907,10 @@ public:
u_int16_t PES_packet_length; //16bits
// 1B
// 2bits const '10'
/**
* 2bits const '10'
*/
int8_t const2bits; //2bits
/**
* The 2-bit PES_scrambling_control field indicates the scrambling mode of the PES packet
* payload. When scrambling is performed at the PES level, the PES packet header, including the optional fields when
@ -1068,7 +1112,10 @@ public:
* PES header.
*/
int8_t P_STD_buffer_flag; //1bit
// reserved 3bits
/**
* reverved value, must be '1'
*/
int8_t const1_value0; //3bits
/**
* A 1-bit field which when set to '1' indicates the presence of the PES_extension_field_length
* field and associated fields. When set to a value of '0' this indicates that the PES_extension_field_length field and any
@ -1179,6 +1226,7 @@ public:
virtual int encode(SrsStream* stream);
private:
virtual int decode_33bits_dts_pts(SrsStream* stream, int64_t* pv);
virtual int encode_33bits_dts_pts(SrsStream* stream, u_int8_t fb, int64_t v);
};
/**
@ -1516,52 +1564,18 @@ public:
virtual int update_acodec(SrsCodecAudio ac);
/**
* write an audio frame to ts,
* @remark write PSI first when not write yet.
*/
virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
virtual int write_audio(SrsTsMessage* audio);
/**
* write a video frame to ts,
* @remark write PSI first when not write yet.
*/
virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
virtual int write_video(SrsTsMessage* video);
/**
* close the writer.
*/
virtual void close();
};
/**
* jitter correct for audio,
* the sample rate 44100/32000 will lost precise,
* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
* so the Hls on ipad or iphone will corrupt,
* @see nginx-rtmp: est_pts
*/
class SrsTsAacJitter
{
private:
int64_t base_pts;
int64_t nb_samples;
int sync_ms;
public:
SrsTsAacJitter();
virtual ~SrsTsAacJitter();
/**
* when buffer start, calc the "correct" pts for ts,
* @param flv_pts, the flv pts calc from flv header timestamp,
* @param sample_rate, the sample rate in format(flv/RTMP packet header).
* @param aac_sample_rate, the sample rate in codec(sequence header).
* @return the calc correct pts.
*/
virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
/**
* when buffer continue, muxer donot write to file,
* the audio buffer continue grow and donot need a pts,
* for the ts audio PES packet only has one pts at the first time.
*/
virtual void on_buffer_continue();
};
/**
* ts stream cache,
* use to cache ts stream.
@ -1575,18 +1589,9 @@ public:
class SrsTsCache
{
public:
// current frame and buffer
SrsMpegtsFrame* af;
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;
// current ts message.
SrsTsMessage* audio;
SrsTsMessage* video;
public:
SrsTsCache();
virtual ~SrsTsCache();
@ -1594,14 +1599,15 @@ public:
/**
* write audio to cache
*/
virtual int cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample);
virtual int cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
/**
* write video to muxer.
*/
virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
private:
virtual int do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_mp3(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_aac(SrsAvcAacCodec* codec, SrsCodecSample* sample);
virtual int do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample);
};
/**