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https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-10-31 00:37:39 +08:00
优化rtp解析代码
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d191347839
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@ -21,12 +21,6 @@ static string printAddress(const struct sockaddr *addr){
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}
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RtpProcess::RtpProcess(const string &stream_id) {
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_track = std::make_shared<SdpTrack>();
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_track->_interleaved = 0;
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_track->_samplerate = 90000;
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_track->_type = TrackVideo;
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_track->_ssrc = 0;
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_media_info._schema = RTP_APP_NAME;
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_media_info._vhost = DEFAULT_VHOST;
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_media_info._app = RTP_APP_NAME;
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@ -104,7 +98,7 @@ bool RtpProcess::inputRtp(const Socket::Ptr &sock, const char *data, int data_le
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}
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_total_bytes += data_len;
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bool ret = handleOneRtp(0,_track,(unsigned char *)data,data_len);
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bool ret = handleOneRtp(0, TrackVideo, 90000, (unsigned char *) data, data_len);
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if(dts_out){
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*dts_out = _dts;
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}
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@ -78,7 +78,6 @@ private:
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std::shared_ptr<FILE> _save_file_rtp;
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std::shared_ptr<FILE> _save_file_ps;
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std::shared_ptr<FILE> _save_file_video;
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SdpTrack::Ptr _track;
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struct sockaddr *_addr = nullptr;
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uint16_t _sequence = 0;
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MultiMediaSourceMuxer::Ptr _muxer;
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@ -27,7 +27,7 @@ namespace mediakit {
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RtpReceiver::RtpReceiver() {}
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RtpReceiver::~RtpReceiver() {}
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bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
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bool RtpReceiver::handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
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if(rtp_raw_len < 12){
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WarnL << "rtp包太小:" << rtp_raw_len;
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return false;
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@ -46,8 +46,8 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
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auto rtp_ptr = _rtp_pool.obtain();
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auto &rtp = *rtp_ptr;
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rtp.type = track->_type;
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rtp.interleaved = 2 * track->_type;
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rtp.type = type;
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rtp.interleaved = 2 * type;
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rtp.mark = rtp_raw_ptr[1] >> 7;
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rtp.PT = rtp_raw_ptr[1] & 0x7F;
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@ -59,29 +59,29 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
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memcpy(&rtp.timeStamp, rtp_raw_ptr + 4, 4);
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rtp.timeStamp = ntohl(rtp.timeStamp);
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if(!track->_samplerate){
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if(!samplerate){
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//无法把时间戳转换成毫秒
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return false;
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}
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//时间戳转换成毫秒
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rtp.timeStamp = rtp.timeStamp * 1000LL / track->_samplerate;
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rtp.timeStamp = rtp.timeStamp * 1000LL / samplerate;
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//ssrc,内存对齐
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memcpy(&rtp.ssrc, rtp_raw_ptr + 8, 4);
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rtp.ssrc = ntohl(rtp.ssrc);
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if (track->_ssrc != rtp.ssrc) {
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if (track->_ssrc == 0) {
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if (_ssrc[track_index] != rtp.ssrc) {
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if (_ssrc[track_index] == 0) {
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//保存SSRC至track对象
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track->_ssrc = rtp.ssrc;
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_ssrc[track_index] = rtp.ssrc;
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}else{
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//ssrc错误
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WarnL << "ssrc错误:" << rtp.ssrc << " != " << track->_ssrc;
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WarnL << "ssrc错误:" << rtp.ssrc << " != " << _ssrc[track_index];
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if (_ssrc_err_count[track_index]++ > 10) {
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//ssrc切换后清除老数据
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WarnL << "ssrc更换:" << track->_ssrc << " -> " << rtp.ssrc;
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WarnL << "ssrc更换:" << _ssrc[track_index] << " -> " << rtp.ssrc;
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_rtp_sort_cache_map[track_index].clear();
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track->_ssrc = rtp.ssrc;
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_ssrc[track_index] = rtp.ssrc;
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}
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return false;
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}
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@ -168,11 +168,12 @@ void RtpReceiver::sortRtp(const RtpPacket::Ptr &rtp,int track_index){
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}
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void RtpReceiver::clear() {
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CLEAR_ARR(_last_seq)
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CLEAR_ARR(_ssrc_err_count)
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CLEAR_ARR(_seq_ok_count)
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CLEAR_ARR(_sort_started)
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CLEAR_ARR(_seq_cycle_count)
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CLEAR_ARR(_last_seq);
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CLEAR_ARR(_ssrc);
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CLEAR_ARR(_ssrc_err_count);
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CLEAR_ARR(_seq_ok_count);
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CLEAR_ARR(_sort_started);
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CLEAR_ARR(_seq_cycle_count);
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_rtp_sort_cache_map[0].clear();
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_rtp_sort_cache_map[1].clear();
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@ -32,12 +32,13 @@ protected:
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/**
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* 输入数据指针生成并排序rtp包
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* @param track_index track下标索引
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* @param track sdp track相关信息
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* @param type track类型
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* @param samplerate rtp时间戳基准时钟,视频为90000,音频为采样率
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* @param rtp_raw_ptr rtp数据指针
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* @param rtp_raw_len rtp数据指针长度
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* @return 解析成功返回true
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*/
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bool handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
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bool handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
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/**
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* rtp数据包排序后输出
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@ -49,9 +50,12 @@ protected:
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void setPoolSize(int size);
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int getJitterSize(int track_index);
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int getCycleCount(int track_index);
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private:
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void sortRtp(const RtpPacket::Ptr &rtp , int track_index);
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private:
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uint32_t _ssrc[2] = { 0, 0 };
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//ssrc不匹配计数
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uint32_t _ssrc_err_count[2] = { 0, 0 };
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//上次seq
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@ -319,7 +319,8 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int uiTrackIndex)
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WarnL << "收到其他地址的rtp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
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return;
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}
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strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex], (unsigned char *) buf->data(), buf->size());
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strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex]->_type,
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strongSelf->_sdp_track[uiTrackIndex]->_samplerate, (unsigned char *) buf->data(), buf->size());
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});
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if(pRtcpSockRef) {
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@ -464,7 +465,7 @@ void RtspPlayer::onRtpPacket(const char *data, uint64_t len) {
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uint8_t interleaved = data[1];
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if(interleaved %2 == 0){
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trackIdx = getTrackIndexByInterleaved(interleaved);
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handleOneRtp(trackIdx, _sdp_track[trackIdx], (unsigned char *)data + 4, len - 4);
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handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (unsigned char *)data + 4, len - 4);
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}else{
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trackIdx = getTrackIndexByInterleaved(interleaved - 1);
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onRtcpPacket(trackIdx, _sdp_track[trackIdx], (unsigned char *) data + 4, len - 4);
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@ -189,7 +189,7 @@ void RtspSession::onRtpPacket(const char *data, uint64_t len) {
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uint8_t interleaved = data[1];
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if(interleaved %2 == 0){
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trackIdx = getTrackIndexByInterleaved(interleaved);
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handleOneRtp(trackIdx,_aTrackInfo[trackIdx],(unsigned char *)data + 4, len - 4);
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handleOneRtp(trackIdx, _aTrackInfo[trackIdx]->_type, _aTrackInfo[trackIdx]->_samplerate, (unsigned char *) data + 4, len - 4);
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}else{
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trackIdx = getTrackIndexByInterleaved(interleaved - 1);
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onRtcpPacket(trackIdx, _aTrackInfo[trackIdx], (unsigned char *) data + 4, len - 4);
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@ -922,7 +922,8 @@ inline void RtspSession::onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBu
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if(intervaled % 2 == 0){
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if(_pushSrc){
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//这是rtsp推流上来的rtp包
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handleOneRtp(intervaled / 2,_aTrackInfo[intervaled / 2],( unsigned char *)pBuf->data(),pBuf->size());
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auto &ref = _aTrackInfo[intervaled / 2];
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handleOneRtp(intervaled / 2, ref->_type, ref->_samplerate, (unsigned char *) pBuf->data(), pBuf->size());
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}else if(!_udpSockConnected.count(intervaled)){
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//这是rtsp播放器的rtp打洞包
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_udpSockConnected.emplace(intervaled);
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