mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-26 04:31:37 +08:00
注释,说明
注释,说明
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e3139134e8
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036b36f012
@ -70,7 +70,7 @@ void MediaSink::inputFrame(const Frame::Ptr &frame) {
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it->second->inputFrame(frame);
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if(!_allTrackReady && !_trackReadyCallback.empty() && it->second->ready()){
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//Track由未就绪状态装换成就绪状态,我们就触发onTrackReady回调
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//Track由未就绪状态转换成就绪状态,我们就触发onTrackReady回调
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auto it_callback = _trackReadyCallback.find(codec_id);
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if(it_callback != _trackReadyCallback.end()){
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it_callback->second();
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@ -123,6 +123,7 @@ MediaSource::Ptr MediaSource::find(
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lock_guard<recursive_mutex> lock(g_mtxMediaSrc);
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MediaSource::Ptr ret;
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//查找某一媒体源,找到后返回
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searchMedia(schema, vhost, app, id,
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[&](SchemaVhostAppStreamMap::iterator &it0 ,
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VhostAppStreamMap::iterator &it1,
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@ -138,7 +139,7 @@ MediaSource::Ptr MediaSource::find(
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return true;
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});
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if(!ret && bMake){
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//查找某一媒体源,找到后返回
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//未查找媒体源,则创建一个
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ret = MediaReader::onMakeMediaSource(schema, vhost,app,id);
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}
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return ret;
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@ -36,6 +36,7 @@ using namespace toolkit;
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namespace mediakit{
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/**
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* h264 Rtmp解码类
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* 将 h264 over rtmp 解复用出 h264-Frame
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*/
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class H264RtmpDecoder : public RtmpCodec ,public ResourcePoolHelper<H264Frame> {
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public:
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@ -90,7 +90,24 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
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* Type==7:SPS frame
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* Type==8:PPS frame
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*/
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/*
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RTF3984 5.2节 Common Structure of the RTP Payload Format
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Table 1. Summary of NAL unit types and their payload structures
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Type Packet Type name Section
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---------------------------------------------------------
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0 undefined -
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1-23 NAL unit Single NAL unit packet per H.264 5.6
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24 STAP-A Single-time aggregation packet 5.7.1
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25 STAP-B Single-time aggregation packet 5.7.1
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26 MTAP16 Multi-time aggregation packet 5.7.2
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27 MTAP24 Multi-time aggregation packet 5.7.2
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28 FU-A Fragmentation unit 5.8
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29 FU-B Fragmentation unit 5.8
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30-31 undefined -
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*/
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const uint8_t *frame = (uint8_t *) rtppack->data() + rtppack->offset;
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int length = rtppack->size() - rtppack->offset;
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NALU nal;
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@ -145,7 +162,7 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
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FU fu;
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MakeFU(frame[1], fu);
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if (fu.S) {
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//该帧的第一个rtp包
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//该帧的第一个rtp包 FU-A start
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char tmp = (nal.forbidden_zero_bit << 7 | nal.nal_ref_idc << 5 | fu.type);
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_h264frame->buffer.assign("\x0\x0\x0\x1", 4);
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_h264frame->buffer.push_back(tmp);
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@ -164,14 +181,14 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
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}
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if (!fu.E) {
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//该帧的中间rtp包
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//该帧的中间rtp包 FU-A mid
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_h264frame->buffer.append((char *)frame + 2, length - 2);
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//该函数return时,保存下当前sequence,以便下次对比seq是否连续
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_lastSeq = rtppack->sequence;
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return false;
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}
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//该帧最后一个rtp包
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//该帧最后一个rtp包 FU-A end
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_h264frame->buffer.append((char *)frame + 2, length - 2);
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_h264frame->timeStamp = rtppack->timeStamp;
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auto key = _h264frame->keyFrame();
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@ -36,6 +36,8 @@ namespace mediakit{
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/**
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* h264 rtp解码类
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* 将 h264 over rtsp-rtp 解复用出 h264-Frame
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* rfc3984
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*/
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class H264RtpDecoder : public RtpCodec , public ResourcePoolHelper<H264Frame> {
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public:
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@ -37,6 +37,8 @@ namespace mediakit{
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/**
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* h265 rtp解码类
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* 将 h265 over rtsp-rtp 解复用出 h265-Frame
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* 《草案(H265-over-RTP)draft-ietf-payload-rtp-h265-07.pdf》
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*/
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class H265RtpDecoder : public RtpCodec , public ResourcePoolHelper<H265Frame> {
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public:
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@ -490,6 +490,7 @@ inline void HttpSession::Handle_Req_GET(int64_t &content_len) {
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//再看看是否为http-flv直播请求
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if(checkLiveFlvStream()){
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//若是,return!
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return;
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}
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@ -44,7 +44,7 @@ using namespace toolkit;
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using namespace mediakit::Client;
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namespace mediakit {
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//实现了rtmp播放器协议部分的功能,及数据接收功能
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class RtmpPlayer:public PlayerBase, public TcpClient, public RtmpProtocol{
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public:
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typedef std::shared_ptr<RtmpPlayer> Ptr;
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@ -63,11 +63,11 @@ protected:
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void onMediaData_l(const RtmpPacket::Ptr &chunkData);
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void onPlayResult_l(const SockException &ex);
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//for Tcpclient
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//form Tcpclient
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void onRecv(const Buffer::Ptr &pBuf) override;
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void onConnect(const SockException &err) override;
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void onErr(const SockException &ex) override;
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//fro RtmpProtocol
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//from RtmpProtocol
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void onRtmpChunk(RtmpPacket &chunkData) override;
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void onStreamDry(uint32_t ui32StreamId) override;
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void onSendRawData(const Buffer::Ptr &buffer) override{
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@ -39,7 +39,7 @@ RtspMuxer::RtspMuxer(const TitleSdp::Ptr &title){
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}
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void RtspMuxer::onTrackReady(const Track::Ptr &track) {
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//根据track生产sdp
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//根据track生成sdp
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Sdp::Ptr sdp = track->getSdp();
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if (!sdp) {
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return;
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@ -46,7 +46,7 @@ using namespace toolkit;
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namespace mediakit {
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//实现了rtsp播放器协议部分的功能
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//实现了rtsp播放器协议部分的功能,及数据接收功能
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class RtspPlayer: public PlayerBase,public TcpClient, public RtspSplitter, public RtpReceiver {
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public:
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typedef std::shared_ptr<RtspPlayer> Ptr;
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@ -361,10 +361,11 @@ void RtspSession::onAuthSuccess() {
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strongSelf->shutdown(SockException(Err_shutdown,err));
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return;
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}
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//找到了响应的rtsp流
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//找到了相应的rtsp流
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strongSelf->_aTrackInfo = SdpParser(rtsp_src->getSdp()).getAvailableTrack();
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if (strongSelf->_aTrackInfo.empty()) {
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//该流无效
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DebugL << "无trackInfo,该流无效";
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strongSelf->send_StreamNotFound();
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strongSelf->shutdown(SockException(Err_shutdown,"can not find any availabe track in sdp"));
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return;
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