优化rtp解析代码

This commit is contained in:
xiongziliang 2020-07-08 23:23:11 +08:00
parent d191347839
commit a164b28608
6 changed files with 30 additions and 30 deletions

View File

@ -21,12 +21,6 @@ static string printAddress(const struct sockaddr *addr){
}
RtpProcess::RtpProcess(const string &stream_id) {
_track = std::make_shared<SdpTrack>();
_track->_interleaved = 0;
_track->_samplerate = 90000;
_track->_type = TrackVideo;
_track->_ssrc = 0;
_media_info._schema = RTP_APP_NAME;
_media_info._vhost = DEFAULT_VHOST;
_media_info._app = RTP_APP_NAME;
@ -104,7 +98,7 @@ bool RtpProcess::inputRtp(const Socket::Ptr &sock, const char *data, int data_le
}
_total_bytes += data_len;
bool ret = handleOneRtp(0,_track,(unsigned char *)data,data_len);
bool ret = handleOneRtp(0, TrackVideo, 90000, (unsigned char *) data, data_len);
if(dts_out){
*dts_out = _dts;
}

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@ -78,7 +78,6 @@ private:
std::shared_ptr<FILE> _save_file_rtp;
std::shared_ptr<FILE> _save_file_ps;
std::shared_ptr<FILE> _save_file_video;
SdpTrack::Ptr _track;
struct sockaddr *_addr = nullptr;
uint16_t _sequence = 0;
MultiMediaSourceMuxer::Ptr _muxer;

View File

@ -27,7 +27,7 @@ namespace mediakit {
RtpReceiver::RtpReceiver() {}
RtpReceiver::~RtpReceiver() {}
bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
bool RtpReceiver::handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len) {
if(rtp_raw_len < 12){
WarnL << "rtp包太小:" << rtp_raw_len;
return false;
@ -46,8 +46,8 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
auto rtp_ptr = _rtp_pool.obtain();
auto &rtp = *rtp_ptr;
rtp.type = track->_type;
rtp.interleaved = 2 * track->_type;
rtp.type = type;
rtp.interleaved = 2 * type;
rtp.mark = rtp_raw_ptr[1] >> 7;
rtp.PT = rtp_raw_ptr[1] & 0x7F;
@ -59,29 +59,29 @@ bool RtpReceiver::handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned ch
memcpy(&rtp.timeStamp, rtp_raw_ptr + 4, 4);
rtp.timeStamp = ntohl(rtp.timeStamp);
if(!track->_samplerate){
if(!samplerate){
//无法把时间戳转换成毫秒
return false;
}
//时间戳转换成毫秒
rtp.timeStamp = rtp.timeStamp * 1000LL / track->_samplerate;
rtp.timeStamp = rtp.timeStamp * 1000LL / samplerate;
//ssrc,内存对齐
memcpy(&rtp.ssrc, rtp_raw_ptr + 8, 4);
rtp.ssrc = ntohl(rtp.ssrc);
if (track->_ssrc != rtp.ssrc) {
if (track->_ssrc == 0) {
if (_ssrc[track_index] != rtp.ssrc) {
if (_ssrc[track_index] == 0) {
//保存SSRC至track对象
track->_ssrc = rtp.ssrc;
_ssrc[track_index] = rtp.ssrc;
}else{
//ssrc错误
WarnL << "ssrc错误:" << rtp.ssrc << " != " << track->_ssrc;
WarnL << "ssrc错误:" << rtp.ssrc << " != " << _ssrc[track_index];
if (_ssrc_err_count[track_index]++ > 10) {
//ssrc切换后清除老数据
WarnL << "ssrc更换:" << track->_ssrc << " -> " << rtp.ssrc;
WarnL << "ssrc更换:" << _ssrc[track_index] << " -> " << rtp.ssrc;
_rtp_sort_cache_map[track_index].clear();
track->_ssrc = rtp.ssrc;
_ssrc[track_index] = rtp.ssrc;
}
return false;
}
@ -168,11 +168,12 @@ void RtpReceiver::sortRtp(const RtpPacket::Ptr &rtp,int track_index){
}
void RtpReceiver::clear() {
CLEAR_ARR(_last_seq)
CLEAR_ARR(_ssrc_err_count)
CLEAR_ARR(_seq_ok_count)
CLEAR_ARR(_sort_started)
CLEAR_ARR(_seq_cycle_count)
CLEAR_ARR(_last_seq);
CLEAR_ARR(_ssrc);
CLEAR_ARR(_ssrc_err_count);
CLEAR_ARR(_seq_ok_count);
CLEAR_ARR(_sort_started);
CLEAR_ARR(_seq_cycle_count);
_rtp_sort_cache_map[0].clear();
_rtp_sort_cache_map[1].clear();

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@ -32,12 +32,13 @@ protected:
/**
* rtp包
* @param track_index track下标索引
* @param track sdp track相关信息
* @param type track类型
* @param samplerate rtp时间戳基准时钟90000
* @param rtp_raw_ptr rtp数据指针
* @param rtp_raw_len rtp数据指针长度
* @return true
*/
bool handleOneRtp(int track_index,SdpTrack::Ptr &track, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
bool handleOneRtp(int track_index, TrackType type, int samplerate, unsigned char *rtp_raw_ptr, unsigned int rtp_raw_len);
/**
* rtp数据包排序后输出
@ -49,9 +50,12 @@ protected:
void setPoolSize(int size);
int getJitterSize(int track_index);
int getCycleCount(int track_index);
private:
void sortRtp(const RtpPacket::Ptr &rtp , int track_index);
private:
uint32_t _ssrc[2] = { 0, 0 };
//ssrc不匹配计数
uint32_t _ssrc_err_count[2] = { 0, 0 };
//上次seq

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@ -319,7 +319,8 @@ void RtspPlayer::handleResSETUP(const Parser &parser, unsigned int uiTrackIndex)
WarnL << "收到其他地址的rtp数据:" << SockUtil::inet_ntoa(((struct sockaddr_in *) addr)->sin_addr);
return;
}
strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex], (unsigned char *) buf->data(), buf->size());
strongSelf->handleOneRtp(uiTrackIndex, strongSelf->_sdp_track[uiTrackIndex]->_type,
strongSelf->_sdp_track[uiTrackIndex]->_samplerate, (unsigned char *) buf->data(), buf->size());
});
if(pRtcpSockRef) {
@ -464,7 +465,7 @@ void RtspPlayer::onRtpPacket(const char *data, uint64_t len) {
uint8_t interleaved = data[1];
if(interleaved %2 == 0){
trackIdx = getTrackIndexByInterleaved(interleaved);
handleOneRtp(trackIdx, _sdp_track[trackIdx], (unsigned char *)data + 4, len - 4);
handleOneRtp(trackIdx, _sdp_track[trackIdx]->_type, _sdp_track[trackIdx]->_samplerate, (unsigned char *)data + 4, len - 4);
}else{
trackIdx = getTrackIndexByInterleaved(interleaved - 1);
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) {
uint8_t interleaved = data[1];
if(interleaved %2 == 0){
trackIdx = getTrackIndexByInterleaved(interleaved);
handleOneRtp(trackIdx,_aTrackInfo[trackIdx],(unsigned char *)data + 4, len - 4);
handleOneRtp(trackIdx, _aTrackInfo[trackIdx]->_type, _aTrackInfo[trackIdx]->_samplerate, (unsigned char *) data + 4, len - 4);
}else{
trackIdx = getTrackIndexByInterleaved(interleaved - 1);
onRtcpPacket(trackIdx, _aTrackInfo[trackIdx], (unsigned char *) data + 4, len - 4);
@ -922,7 +922,8 @@ inline void RtspSession::onRcvPeerUdpData(int intervaled, const Buffer::Ptr &pBu
if(intervaled % 2 == 0){
if(_pushSrc){
//这是rtsp推流上来的rtp包
handleOneRtp(intervaled / 2,_aTrackInfo[intervaled / 2],( unsigned char *)pBuf->data(),pBuf->size());
auto &ref = _aTrackInfo[intervaled / 2];
handleOneRtp(intervaled / 2, ref->_type, ref->_samplerate, (unsigned char *) pBuf->data(), pBuf->size());
}else if(!_udpSockConnected.count(intervaled)){
//这是rtsp播放器的rtp打洞包
_udpSockConnected.emplace(intervaled);