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rewrite the ts remux of hls. 2.0.117
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winlin
parent
1ed3e283ab
commit
3c813847bd
5 changed files with 817 additions and 316 deletions
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@ -185,6 +185,8 @@ struct SrsTsChannel
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SrsTsPidApply apply;
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SrsTsStream stream;
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SrsTsMessage* msg;
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// for encoder.
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u_int8_t continuity_counter;
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SrsTsChannel();
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virtual ~SrsTsChannel();
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@ -209,13 +211,17 @@ enum SrsTsPESStreamId
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// ISO/IEC 13818-3 or ISO/IEC 11172-3 or ISO/IEC 13818-7 or ISO/IEC
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// 14496-3 audio stream number x xxxx
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// ((sid >> 5) & 0x07) == SrsTsPESStreamIdAudio
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SrsTsPESStreamIdAudio = 0x06, // 0b110
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// @remark, use SrsTsPESStreamIdAudioCommon as actually audio, SrsTsPESStreamIdAudio to check whether audio.
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SrsTsPESStreamIdAudioChecker = 0x06, // 0b110
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SrsTsPESStreamIdAudioCommon = 0xc0,
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// 1110 xxxx
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// ITU-T Rec. H.262 | ISO/IEC 13818-2 or ISO/IEC 11172-2 or ISO/IEC
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// 14496-2 video stream number xxxx
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// ((stream_id >> 4) & 0x0f) == SrsTsPESStreamIdVideo
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SrsTsPESStreamIdVideo = 0x0e, // 0b1110
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// @remark, use SrsTsPESStreamIdVideoCommon as actually video, SrsTsPESStreamIdVideo to check whether video.
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SrsTsPESStreamIdVideoChecker = 0x0e, // 0b1110
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SrsTsPESStreamIdVideoCommon = 0xe0,
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// ECM_stream
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SrsTsPESStreamIdEcmStream = 0xf0, // 0b11110000
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@ -253,8 +259,20 @@ enum SrsTsPESStreamId
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class SrsTsMessage
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{
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public:
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// decoder only,
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// the ts messgae does not use them,
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// for user to get the channel and packet.
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SrsTsChannel* channel;
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SrsTsPacket* packet;
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public:
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// the audio cache buffer start pts, to flush audio if full.
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// @remark the pts is not the adjust one, it's the orignal pts.
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int64_t start_pts;
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// whether this message with pcr info,
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// generally, the video IDR(I frame, the keyframe of h.264) carray the pcr info.
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bool write_pcr;
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// whether got discontinuity ts, for example, sequence header changed.
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bool discontinuity;
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public:
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// the timestamp in 90khz
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int64_t dts;
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@ -269,8 +287,9 @@ public:
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// the payload bytes.
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SrsSimpleBuffer* payload;
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public:
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SrsTsMessage(SrsTsChannel* c, SrsTsPacket* p);
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SrsTsMessage(SrsTsChannel* c = NULL, SrsTsPacket* p = NULL);
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virtual ~SrsTsMessage();
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// decoder
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public:
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/**
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* dumps all bytes in stream to ts message.
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@ -361,15 +380,14 @@ public:
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public:
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/**
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* write the PES packet, the video/audio stream.
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* @param frame the video/audio frame info.
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* @param payload the video/audio payload bytes.
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* @param msg the video/audio msg to write to ts.
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* @param vc the video codec, write the PAT/PMT table when changed.
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* @param ac the audio codec, write the PAT/PMT table when changed.
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*/
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virtual int encode(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* payload, SrsCodecVideo vc, SrsCodecAudio ac);
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virtual int encode(SrsFileWriter* writer, SrsTsMessage* msg, SrsCodecVideo vc, SrsCodecAudio ac);
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private:
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virtual int encode_pat_pmt(SrsFileWriter* writer, SrsCodecVideo vcodec, SrsCodecAudio acodec);
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virtual int encode_pes(SrsFileWriter* writer, SrsMpegtsFrame* frame, SrsSimpleBuffer* payload);
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virtual int encode_pat_pmt(SrsFileWriter* writer, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as);
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virtual int encode_pes(SrsFileWriter* writer, SrsTsMessage* msg, int16_t pid, SrsTsStream sid);
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};
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/**
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@ -478,9 +496,22 @@ public:
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public:
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virtual int size();
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virtual int encode(SrsStream* stream);
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virtual void padding(int nb_stuffings);
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public:
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static SrsTsPacket* create_pat(SrsTsContext* context, int16_t pmt_number, int16_t pmt_pid);
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static SrsTsPacket* create_pmt(SrsTsContext* context, int16_t pmt_number, int16_t pmt_pid, int16_t vpid, SrsTsStream vs, int16_t apid, SrsTsStream as);
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static SrsTsPacket* create_pat(SrsTsContext* context,
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int16_t pmt_number, int16_t pmt_pid
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);
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static SrsTsPacket* create_pmt(SrsTsContext* context,
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int16_t pmt_number, int16_t pmt_pid, int16_t vpid, SrsTsStream vs,
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int16_t apid, SrsTsStream as
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);
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static SrsTsPacket* create_pes_first(SrsTsContext* context,
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int16_t pid, SrsTsPESStreamId sid, u_int8_t continuity_counter, bool discontinuity,
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int64_t pcr, int64_t dts, int64_t pts, int size
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);
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static SrsTsPacket* create_pes_continue(SrsTsContext* context,
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int16_t pid, SrsTsPESStreamId sid, u_int8_t continuity_counter
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);
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};
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/**
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@ -627,7 +658,10 @@ public:
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* the last bit of the program_clock_reference_base at the input of the system target decoder.
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*/
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int64_t program_clock_reference_base; //33bits
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//6bits reserved.
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/**
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* 6bits reserved, must be '1'
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*/
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int8_t const1_value0; // 6bits
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int16_t program_clock_reference_extension; //9bits
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// if OPCR_flag, 6B
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@ -646,7 +680,10 @@ public:
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* in the original single program Transport Stream.
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*/
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int64_t original_program_clock_reference_base; //33bits
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//6bits reserved.
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/**
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* 6bits reserved, must be '1'
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*/
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int8_t const1_value2; // 6bits
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int16_t original_program_clock_reference_extension; //9bits
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// if splicing_point_flag, 1B
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@ -711,7 +748,10 @@ public:
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* constraints indicated by the splice_type value.
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*/
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int8_t seamless_splice_flag; //1bit
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//5bits reserved
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/**
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* reserved 5bits, must be '1'
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*/
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int8_t const1_value1; //5bits
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// if ltw_flag, 2B
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/**
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* (legal time window_valid_flag) - This is a 1-bit field which when set to '1' indicates that the value of the
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@ -856,6 +896,7 @@ public:
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* elementary stream type as defined in Table 2-18. In Transport Streams, the elementary stream type is specified in the
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* Program Specific Information as specified in 2.4.4.
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*/
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// @see SrsTsPESStreamId, value can be SrsTsPESStreamIdAudioCommon or SrsTsPESStreamIdVideoCommon.
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u_int8_t stream_id; //8bits
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// 2B
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/**
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@ -866,7 +907,10 @@ public:
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u_int16_t PES_packet_length; //16bits
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// 1B
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// 2bits const '10'
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/**
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* 2bits const '10'
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*/
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int8_t const2bits; //2bits
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/**
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* The 2-bit PES_scrambling_control field indicates the scrambling mode of the PES packet
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* payload. When scrambling is performed at the PES level, the PES packet header, including the optional fields when
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@ -1068,7 +1112,10 @@ public:
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* PES header.
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*/
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int8_t P_STD_buffer_flag; //1bit
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// reserved 3bits
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/**
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* reverved value, must be '1'
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*/
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int8_t const1_value0; //3bits
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/**
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* A 1-bit field which when set to '1' indicates the presence of the PES_extension_field_length
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* field and associated fields. When set to a value of '0' this indicates that the PES_extension_field_length field and any
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@ -1179,6 +1226,7 @@ public:
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virtual int encode(SrsStream* stream);
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private:
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virtual int decode_33bits_dts_pts(SrsStream* stream, int64_t* pv);
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virtual int encode_33bits_dts_pts(SrsStream* stream, u_int8_t fb, int64_t v);
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};
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/**
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@ -1516,52 +1564,18 @@ public:
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virtual int update_acodec(SrsCodecAudio ac);
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/**
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* write an audio frame to ts,
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* @remark write PSI first when not write yet.
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*/
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virtual int write_audio(SrsMpegtsFrame* af, SrsSimpleBuffer* ab);
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virtual int write_audio(SrsTsMessage* audio);
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/**
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* write a video frame to ts,
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* @remark write PSI first when not write yet.
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*/
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virtual int write_video(SrsMpegtsFrame* vf, SrsSimpleBuffer* vb);
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virtual int write_video(SrsTsMessage* video);
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/**
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* close the writer.
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*/
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virtual void close();
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};
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/**
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* jitter correct for audio,
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* the sample rate 44100/32000 will lost precise,
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* when mp4/ts(tbn=90000) covert to flv/rtmp(1000),
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* so the Hls on ipad or iphone will corrupt,
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* @see nginx-rtmp: est_pts
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*/
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class SrsTsAacJitter
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{
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private:
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int64_t base_pts;
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int64_t nb_samples;
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int sync_ms;
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public:
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SrsTsAacJitter();
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virtual ~SrsTsAacJitter();
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/**
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* when buffer start, calc the "correct" pts for ts,
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* @param flv_pts, the flv pts calc from flv header timestamp,
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* @param sample_rate, the sample rate in format(flv/RTMP packet header).
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* @param aac_sample_rate, the sample rate in codec(sequence header).
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* @return the calc correct pts.
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*/
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virtual int64_t on_buffer_start(int64_t flv_pts, int sample_rate, int aac_sample_rate);
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/**
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* when buffer continue, muxer donot write to file,
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* the audio buffer continue grow and donot need a pts,
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* for the ts audio PES packet only has one pts at the first time.
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*/
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virtual void on_buffer_continue();
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};
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/**
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* ts stream cache,
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* use to cache ts stream.
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@ -1575,18 +1589,9 @@ public:
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class SrsTsCache
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{
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public:
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// current frame and buffer
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SrsMpegtsFrame* af;
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SrsSimpleBuffer* ab;
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SrsMpegtsFrame* vf;
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SrsSimpleBuffer* vb;
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public:
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// the audio cache buffer start pts, to flush audio if full.
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// @remark the pts is not the adjust one, it's the orignal pts.
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int64_t audio_buffer_start_pts;
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protected:
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// time jitter for aac
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SrsTsAacJitter* aac_jitter;
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// current ts message.
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SrsTsMessage* audio;
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SrsTsMessage* video;
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public:
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SrsTsCache();
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virtual ~SrsTsCache();
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@ -1594,14 +1599,15 @@ public:
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/**
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* write audio to cache
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*/
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virtual int cache_audio(SrsAvcAacCodec* codec, int64_t pts, SrsCodecSample* sample);
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virtual int cache_audio(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
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/**
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* write video to muxer.
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*/
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virtual int cache_video(SrsAvcAacCodec* codec, int64_t dts, SrsCodecSample* sample);
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private:
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virtual int do_cache_audio(SrsAvcAacCodec* codec, SrsCodecSample* sample);
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virtual int do_cache_video(SrsAvcAacCodec* codec, SrsCodecSample* sample);
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virtual int do_cache_mp3(SrsAvcAacCodec* codec, SrsCodecSample* sample);
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virtual int do_cache_aac(SrsAvcAacCodec* codec, SrsCodecSample* sample);
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virtual int do_cache_avc(SrsAvcAacCodec* codec, SrsCodecSample* sample);
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};
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/**
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