diff --git a/3rdpart/ZLToolKit b/3rdpart/ZLToolKit index a4771b35..12c2ec7b 160000 --- a/3rdpart/ZLToolKit +++ b/3rdpart/ZLToolKit @@ -1 +1 @@ -Subproject commit a4771b353d9ebe1bc162364237a31f4184a7c435 +Subproject commit 12c2ec7b07dce1b6a0543d9979f5c5231f69e828 diff --git a/postman/ZLMediaKit.postman_collection.json b/postman/ZLMediaKit.postman_collection.json index 39377f1a..784bb60c 100644 --- a/postman/ZLMediaKit.postman_collection.json +++ b/postman/ZLMediaKit.postman_collection.json @@ -1758,10 +1758,16 @@ "value": "obs", "description": "流id,例如 obs" }, + { + "key": "ssrc_multi_send", + "value": "0", + "description": "是否支持同ssrc推流到多个上级服务器,该参数非必选参数 默认false", + "disabled": true + }, { "key": "ssrc", "value": "1", - "description": "rtp推流的ssrc,ssrc不同时,可以推流到多个上级服务器" + "description": "rtp推流的ssrc" }, { "key": "dst_url", diff --git a/server/WebApi.cpp b/server/WebApi.cpp index 1034ae4c..63b4d5e3 100755 --- a/server/WebApi.cpp +++ b/server/WebApi.cpp @@ -1182,6 +1182,7 @@ void installWebApi() { //回复json val["port"] = port; }); + api_regist("/index/api/openRtpServerMultiplex", [](API_ARGS_MAP) { CHECK_SECRET(); CHECK_ARGS("port", "stream_id"); @@ -1263,6 +1264,7 @@ void installWebApi() { args.passive = false; args.dst_url = allArgs["dst_url"]; args.dst_port = allArgs["dst_port"]; + args.ssrc_multi_send = allArgs["ssrc_multi_send"].empty() ? false : allArgs["ssrc_multi_send"].as(); args.ssrc = allArgs["ssrc"]; args.is_udp = allArgs["is_udp"]; args.src_port = allArgs["src_port"]; diff --git a/src/Common/MediaSource.h b/src/Common/MediaSource.h index ffced627..73306385 100644 --- a/src/Common/MediaSource.h +++ b/src/Common/MediaSource.h @@ -104,6 +104,8 @@ public: bool passive = false; // rtp payload type uint8_t pt = 96; + //是否支持同ssrc多服务器发送 + bool ssrc_multi_send = false; // 指定rtp ssrc std::string ssrc; // 指定本地发送端口 diff --git a/src/Common/MultiMediaSourceMuxer.cpp b/src/Common/MultiMediaSourceMuxer.cpp index bbb9b357..99898b95 100644 --- a/src/Common/MultiMediaSourceMuxer.cpp +++ b/src/Common/MultiMediaSourceMuxer.cpp @@ -45,6 +45,7 @@ static string getTrackInfoStr(const TrackSource *track_src){ _StrPrinter codec_info; auto tracks = track_src->getTracks(true); for (auto &track : tracks) { + track->update(); auto codec_type = track->getTrackType(); codec_info << track->getCodecName(); switch (codec_type) { @@ -290,12 +291,14 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE auto ring = _ring; auto ssrc = args.ssrc; + auto ssrc_multi_send = args.ssrc_multi_send; auto tracks = getTracks(false); auto poller = getOwnerPoller(sender); auto rtp_sender = std::make_shared(poller); + weak_ptr weak_self = shared_from_this(); - rtp_sender->startSend(args, [ssrc, weak_self, rtp_sender, cb, tracks, ring, poller](uint16_t local_port, const SockException &ex) mutable { + rtp_sender->startSend(args, [ssrc,ssrc_multi_send, weak_self, rtp_sender, cb, tracks, ring, poller](uint16_t local_port, const SockException &ex) mutable { cb(local_port, ex); auto strong_self = weak_self.lock(); if (!strong_self || ex) { @@ -324,7 +327,10 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE // 可能归属线程发生变更 strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() { - strong_self->_rtp_sender[ssrc] = std::move(reader); + if(!ssrc_multi_send) { + strong_self->_rtp_sender.erase(ssrc); + } + strong_self->_rtp_sender.emplace(ssrc,reader); }); }); #else diff --git a/src/Common/MultiMediaSourceMuxer.h b/src/Common/MultiMediaSourceMuxer.h index 4db34d42..45ab7623 100644 --- a/src/Common/MultiMediaSourceMuxer.h +++ b/src/Common/MultiMediaSourceMuxer.h @@ -168,7 +168,7 @@ private: toolkit::Ticker _last_check; Stamp _stamp[2]; std::weak_ptr _track_listener; - std::unordered_map _rtp_sender; + std::unordered_multimap _rtp_sender; FMP4MediaSourceMuxer::Ptr _fmp4; RtmpMediaSourceMuxer::Ptr _rtmp; RtspMediaSourceMuxer::Ptr _rtsp; diff --git a/src/Extension/AAC.cpp b/src/Extension/AAC.cpp index 1047a8b3..27f18359 100644 --- a/src/Extension/AAC.cpp +++ b/src/Extension/AAC.cpp @@ -21,53 +21,56 @@ namespace mediakit{ #ifndef ENABLE_MP4 unsigned const samplingFrequencyTable[16] = { 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0, 0 }; -class AdtsHeader{ +class AdtsHeader { public: - unsigned int syncword = 0; //12 bslbf 同步字The bit string ‘1111 1111 1111’,说明一个ADTS帧的开始 - unsigned int id; //1 bslbf MPEG 标示符, 设置为1 - unsigned int layer; //2 uimsbf Indicates which layer is used. Set to ‘00’ - unsigned int protection_absent; //1 bslbf 表示是否误码校验 - unsigned int profile; //2 uimsbf 表示使用哪个级别的AAC,如01 Low Complexity(LC)--- AACLC - unsigned int sf_index; //4 uimsbf 表示使用的采样率下标 - unsigned int private_bit; //1 bslbf - unsigned int channel_configuration; //3 uimsbf 表示声道数 - unsigned int original; //1 bslbf - unsigned int home; //1 bslbf - //下面的为改变的参数即每一帧都不同 - unsigned int copyright_identification_bit; //1 bslbf - unsigned int copyright_identification_start; //1 bslbf + unsigned int syncword = 0; // 12 bslbf 同步字The bit string ‘1111 1111 1111’,说明一个ADTS帧的开始 + unsigned int id; // 1 bslbf MPEG 标示符, 设置为1 + unsigned int layer; // 2 uimsbf Indicates which layer is used. Set to ‘00’ + unsigned int protection_absent; // 1 bslbf 表示是否误码校验 + unsigned int profile; // 2 uimsbf 表示使用哪个级别的AAC,如01 Low Complexity(LC)--- AACLC + unsigned int sf_index; // 4 uimsbf 表示使用的采样率下标 + unsigned int private_bit; // 1 bslbf + unsigned int channel_configuration; // 3 uimsbf 表示声道数 + unsigned int original; // 1 bslbf + unsigned int home; // 1 bslbf + // 下面的为改变的参数即每一帧都不同 + unsigned int copyright_identification_bit; // 1 bslbf + unsigned int copyright_identification_start; // 1 bslbf unsigned int aac_frame_length; // 13 bslbf 一个ADTS帧的长度包括ADTS头和raw data block - unsigned int adts_buffer_fullness; //11 bslbf 0x7FF 说明是码率可变的码流 - //no_raw_data_blocks_in_frame 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧. - //所以说number_of_raw_data_blocks_in_frame == 0 - //表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据) - unsigned int no_raw_data_blocks_in_frame; //2 uimsfb + unsigned int adts_buffer_fullness; // 11 bslbf 0x7FF 说明是码率可变的码流 + // no_raw_data_blocks_in_frame 表示ADTS帧中有number_of_raw_data_blocks_in_frame + 1个AAC原始帧. + // 所以说number_of_raw_data_blocks_in_frame == 0 + // 表示说ADTS帧中有一个AAC数据块并不是说没有。(一个AAC原始帧包含一段时间内1024个采样及相关数据) + unsigned int no_raw_data_blocks_in_frame; // 2 uimsfb }; static void dumpAdtsHeader(const AdtsHeader &hed, uint8_t *out) { - out[0] = (hed.syncword >> 4 & 0xFF); //8bit - out[1] = (hed.syncword << 4 & 0xF0); //4 bit - out[1] |= (hed.id << 3 & 0x08); //1 bit - out[1] |= (hed.layer << 1 & 0x06); //2bit - out[1] |= (hed.protection_absent & 0x01); //1 bit + out[0] = (hed.syncword >> 4 & 0xFF); // 8bit + out[1] = (hed.syncword << 4 & 0xF0); // 4 bit + out[1] |= (hed.id << 3 & 0x08); // 1 bit + out[1] |= (hed.layer << 1 & 0x06); // 2bit + out[1] |= (hed.protection_absent & 0x01); // 1 bit out[2] = (hed.profile << 6 & 0xC0); // 2 bit - out[2] |= (hed.sf_index << 2 & 0x3C); //4bit - out[2] |= (hed.private_bit << 1 & 0x02); //1 bit - out[2] |= (hed.channel_configuration >> 2 & 0x03); //1 bit - out[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit - out[3] |= (hed.original << 5 & 0x20); //1 bit - out[3] |= (hed.home << 4 & 0x10); //1 bit - out[3] |= (hed.copyright_identification_bit << 3 & 0x08); //1 bit - out[3] |= (hed.copyright_identification_start << 2 & 0x04); //1 bit - out[3] |= (hed.aac_frame_length >> 11 & 0x03); //2 bit - out[4] = (hed.aac_frame_length >> 3 & 0xFF); //8 bit - out[5] = (hed.aac_frame_length << 5 & 0xE0); //3 bit - out[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); //5 bit - out[6] = (hed.adts_buffer_fullness << 2 & 0xFC); //6 bit - out[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); //2 bit + out[2] |= (hed.sf_index << 2 & 0x3C); // 4bit + out[2] |= (hed.private_bit << 1 & 0x02); // 1 bit + out[2] |= (hed.channel_configuration >> 2 & 0x03); // 1 bit + out[3] = (hed.channel_configuration << 6 & 0xC0); // 2 bit + out[3] |= (hed.original << 5 & 0x20); // 1 bit + out[3] |= (hed.home << 4 & 0x10); // 1 bit + out[3] |= (hed.copyright_identification_bit << 3 & 0x08); // 1 bit + out[3] |= (hed.copyright_identification_start << 2 & 0x04); // 1 bit + out[3] |= (hed.aac_frame_length >> 11 & 0x03); // 2 bit + out[4] = (hed.aac_frame_length >> 3 & 0xFF); // 8 bit + out[5] = (hed.aac_frame_length << 5 & 0xE0); // 3 bit + out[5] |= (hed.adts_buffer_fullness >> 6 & 0x1F); // 5 bit + out[6] = (hed.adts_buffer_fullness << 2 & 0xFC); // 6 bit + out[6] |= (hed.no_raw_data_blocks_in_frame & 0x03); // 2 bit } -static void parseAacConfig(const string &config, AdtsHeader &adts) { +static bool parseAacConfig(const string &config, AdtsHeader &adts) { + if (config.size() < 2) { + return false; + } uint8_t cfg1 = config[0]; uint8_t cfg2 = config[1]; @@ -94,6 +97,7 @@ static void parseAacConfig(const string &config, AdtsHeader &adts) { adts.aac_frame_length = 7; adts.adts_buffer_fullness = 2047; adts.no_raw_data_blocks_in_frame = 0; + return true; } #endif// ENABLE_MP4 @@ -168,10 +172,12 @@ int dumpAacConfig(const string &config, size_t length, uint8_t *out, size_t out_ #endif } -bool parseAacConfig(const string &config, int &samplerate, int &channels){ +bool parseAacConfig(const string &config, int &samplerate, int &channels) { #ifndef ENABLE_MP4 AdtsHeader header; - parseAacConfig(config, header); + if (!parseAacConfig(config, header)) { + return false; + } samplerate = samplingFrequencyTable[header.sf_index]; channels = header.channel_configuration; return true; @@ -326,11 +332,14 @@ bool AACTrack::inputFrame_l(const Frame::Ptr &frame) { return false; } +bool AACTrack::update() { + return parseAacConfig(_cfg, _sampleRate, _channel); +} + void AACTrack::onReady() { - if (_cfg.size() < 2) { - return; + if (!parseAacConfig(_cfg, _sampleRate, _channel)) { + _cfg.clear(); } - parseAacConfig(_cfg, _sampleRate, _channel); } Track::Ptr AACTrack::clone() { @@ -342,6 +351,7 @@ Sdp::Ptr AACTrack::getSdp() { WarnL << getCodecName() << " Track未准备好"; return nullptr; } + update(); return std::make_shared(getConfig(), getAudioSampleRate(), getAudioChannel(), getBitRate() / 1024); } diff --git a/src/Extension/AAC.h b/src/Extension/AAC.h index b95fc6f2..32cf934b 100644 --- a/src/Extension/AAC.h +++ b/src/Extension/AAC.h @@ -52,6 +52,7 @@ public: int getAudioSampleRate() const override; int getAudioSampleBit() const override; bool inputFrame(const Frame::Ptr &frame) override; + bool update() override; private: void onReady(); diff --git a/src/Extension/H264.cpp b/src/Extension/H264.cpp index eeccd600..eda6b188 100644 --- a/src/Extension/H264.cpp +++ b/src/Extension/H264.cpp @@ -168,6 +168,10 @@ bool H264Track::inputFrame(const Frame::Ptr &frame) { return ret; } +bool H264Track::update() { + return getAVCInfo(_sps, _width, _height, _fps); +} + void H264Track::onReady() { if (!getAVCInfo(_sps, _width, _height, _fps)) { _sps.clear(); diff --git a/src/Extension/H264.h b/src/Extension/H264.h index 30a8f747..0afe593d 100644 --- a/src/Extension/H264.h +++ b/src/Extension/H264.h @@ -128,6 +128,7 @@ public: int getVideoWidth() const override; float getVideoFps() const override; bool inputFrame(const Frame::Ptr &frame) override; + bool update() override; private: void onReady(); diff --git a/src/Extension/H264Rtp.cpp b/src/Extension/H264Rtp.cpp index 28c775c6..442205a6 100644 --- a/src/Extension/H264Rtp.cpp +++ b/src/Extension/H264Rtp.cpp @@ -44,13 +44,15 @@ H264Frame::Ptr H264RtpDecoder::obtainFrame() { bool H264RtpDecoder::inputRtp(const RtpPacket::Ptr &rtp, bool key_pos) { auto seq = rtp->getSeq(); - auto ret = decodeRtp(rtp); - if (!_gop_dropped && seq != (uint16_t) (_last_seq + 1) && _last_seq) { + auto last_is_gop = _is_gop; + _is_gop = decodeRtp(rtp); + if (!_gop_dropped && seq != (uint16_t)(_last_seq + 1) && _last_seq) { _gop_dropped = true; WarnL << "start drop h264 gop, last seq:" << _last_seq << ", rtp:\r\n" << rtp->dumpString(); } _last_seq = seq; - return ret; + // 确保有sps rtp的时候,gop从sps开始;否则从关键帧开始 + return _is_gop && !last_is_gop; } /* @@ -74,7 +76,7 @@ bool H264RtpDecoder::singleFrame(const RtpPacket::Ptr &rtp, const uint8_t *ptr, _frame->_buffer.assign("\x00\x00\x00\x01", 4); _frame->_buffer.append((char *) ptr, size); _frame->_pts = stamp; - auto key = _frame->keyFrame(); + auto key = _frame->keyFrame() || _frame->configFrame(); outputFrame(rtp, _frame); return key; } @@ -127,7 +129,7 @@ bool H264RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz if (!fu->end_bit) { //非末尾包 - return fu->start_bit ? _frame->keyFrame() : false; + return fu->start_bit ? (_frame->keyFrame() || _frame->configFrame()) : false; } //确保下一次fu必须收到第一个包 diff --git a/src/Extension/H264Rtp.h b/src/Extension/H264Rtp.h index 31200cb8..98d49cda 100644 --- a/src/Extension/H264Rtp.h +++ b/src/Extension/H264Rtp.h @@ -51,6 +51,7 @@ private: void outputFrame(const RtpPacket::Ptr &rtp, const H264Frame::Ptr &frame); private: + bool _is_gop = false; bool _gop_dropped = false; bool _fu_dropped = true; uint16_t _last_seq = 0; diff --git a/src/Extension/H265.cpp b/src/Extension/H265.cpp index 9985a9cb..926ab2a4 100644 --- a/src/Extension/H265.cpp +++ b/src/Extension/H265.cpp @@ -144,6 +144,10 @@ bool H265Track::inputFrame_l(const Frame::Ptr &frame) { return ret; } +bool H265Track::update() { + return getHEVCInfo(_vps, _sps, _width, _height, _fps); +} + void H265Track::onReady() { if (!getHEVCInfo(_vps, _sps, _width, _height, _fps)) { _vps.clear(); diff --git a/src/Extension/H265.h b/src/Extension/H265.h index 521663f4..912f1f46 100644 --- a/src/Extension/H265.h +++ b/src/Extension/H265.h @@ -150,6 +150,7 @@ public: int getVideoHeight() const override; float getVideoFps() const override; bool inputFrame(const Frame::Ptr &frame) override; + bool update() override; private: void onReady(); diff --git a/src/Extension/H265Rtp.cpp b/src/Extension/H265Rtp.cpp index 23bbb109..22866ff4 100644 --- a/src/Extension/H265Rtp.cpp +++ b/src/Extension/H265Rtp.cpp @@ -163,7 +163,7 @@ bool H265RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz if (!e_bit) { //非末尾包 - return s_bit ? _frame->keyFrame() : false; + return s_bit ? (_frame->keyFrame() || _frame->configFrame()) : false; } //确保下一次fu必须收到第一个包 @@ -175,13 +175,15 @@ bool H265RtpDecoder::mergeFu(const RtpPacket::Ptr &rtp, const uint8_t *ptr, ssiz bool H265RtpDecoder::inputRtp(const RtpPacket::Ptr &rtp, bool) { auto seq = rtp->getSeq(); - auto ret = decodeRtp(rtp); + auto last_is_gop = _is_gop; + _is_gop = decodeRtp(rtp); if (!_gop_dropped && seq != (uint16_t) (_last_seq + 1) && _last_seq) { _gop_dropped = true; WarnL << "start drop h265 gop, last seq:" << _last_seq << ", rtp:\r\n" << rtp->dumpString(); } _last_seq = seq; - return ret; + // 确保有sps rtp的时候,gop从sps开始;否则从关键帧开始 + return _is_gop && !last_is_gop; } bool H265RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtp) { @@ -220,7 +222,7 @@ bool H265RtpDecoder::singleFrame(const RtpPacket::Ptr &rtp, const uint8_t *ptr, _frame->_buffer.assign("\x00\x00\x00\x01", 4); _frame->_buffer.append((char *) ptr, size); _frame->_pts = stamp; - auto key = _frame->keyFrame(); + auto key = _frame->keyFrame() || _frame->configFrame(); outputFrame(rtp, _frame); return key; } diff --git a/src/Extension/H265Rtp.h b/src/Extension/H265Rtp.h index cd9702f4..569d8091 100644 --- a/src/Extension/H265Rtp.h +++ b/src/Extension/H265Rtp.h @@ -51,6 +51,7 @@ private: void outputFrame(const RtpPacket::Ptr &rtp, const H265Frame::Ptr &frame); private: + bool _is_gop = false; bool _using_donl_field = false; bool _gop_dropped = false; bool _fu_dropped = true; diff --git a/src/Extension/Track.h b/src/Extension/Track.h index e7aa2b7c..3a1b8519 100644 --- a/src/Extension/Track.h +++ b/src/Extension/Track.h @@ -39,6 +39,11 @@ public: */ virtual Track::Ptr clone() = 0; + /** + * 更新track信息,比如触发sps/pps解析 + */ + virtual bool update() { return false; } + /** * 生成sdp * @return sdp对象 diff --git a/src/Http/HttpClient.cpp b/src/Http/HttpClient.cpp index 328872e6..dd1546db 100644 --- a/src/Http/HttpClient.cpp +++ b/src/Http/HttpClient.cpp @@ -80,7 +80,7 @@ void HttpClient::sendRequest(const string &url) { } if (!alive() || host_changed) { - startConnect(host, port, _wait_header_ms); + startConnect(host, port, _wait_header_ms / 1000.0f); } else { SockException ex; onConnect_l(ex); diff --git a/src/Player/PlayerProxy.cpp b/src/Player/PlayerProxy.cpp index 2a319ff8..73e46b71 100644 --- a/src/Player/PlayerProxy.cpp +++ b/src/Player/PlayerProxy.cpp @@ -70,6 +70,7 @@ void PlayerProxy::setTranslationInfo() _transtalion_info.stream_info.clear(); auto tracks = _muxer->getTracks(); for (auto &track : tracks) { + track->update(); _transtalion_info.stream_info.emplace_back(); auto &back = _transtalion_info.stream_info.back(); back.bitrate = track->getBitRate(); diff --git a/src/Record/MP4Muxer.cpp b/src/Record/MP4Muxer.cpp index c79cc132..5458a31d 100644 --- a/src/Record/MP4Muxer.cpp +++ b/src/Record/MP4Muxer.cpp @@ -198,6 +198,7 @@ bool MP4MuxerInterface::addTrack(const Track::Ptr &track) { return false; } + track->update(); switch (track->getCodecId()) { case CodecG711A: case CodecG711U: diff --git a/src/Rtmp/Rtmp.cpp b/src/Rtmp/Rtmp.cpp index 140afeed..33c46f11 100644 --- a/src/Rtmp/Rtmp.cpp +++ b/src/Rtmp/Rtmp.cpp @@ -57,6 +57,7 @@ AudioMeta::AudioMeta(const AudioTrack::Ptr &audio) { } uint8_t getAudioRtmpFlags(const Track::Ptr &track) { + track->update(); switch (track->getTrackType()) { case TrackAudio: { auto audioTrack = std::dynamic_pointer_cast(track); @@ -115,6 +116,7 @@ uint8_t getAudioRtmpFlags(const Track::Ptr &track) { void Metadata::addTrack(AMFValue &metadata, const Track::Ptr &track) { Metadata::Ptr new_metadata; + track->update(); switch (track->getTrackType()) { case TrackVideo: { new_metadata = std::make_shared(std::dynamic_pointer_cast(track));