rtp使用ntp时间戳作为时间戳,用于实现rtsp音视频同步

src/Common/Stamp.cpp

@@ -35,7 +35,8 @@ int64_t DeltaStamp::deltaStamp(int64_t stamp) {
 
     //时间戳增量为负，说明时间戳回环了或回退了
     _last_stamp = stamp;
-    return 0;
+    //如果时间戳回退不多，那么返回负值
+    return -ret < MAX_CTS ? ret : 0;
 }
 
 void Stamp::setPlayBack(bool playback) {
@@ -215,4 +216,32 @@ bool DtsGenerator::getDts_l(uint32_t pts, uint32_t &dts){
     return false;
 }
 
+void NtpStamp::setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms) {
+    _rtp_stamp_ms = uint64_t(rtp_stamp) * 1000 / sample_rate;
+    _ntp_stamp_ms = ntp_stamp_ms;
+}
+
+uint64_t NtpStamp::getNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate) {
+    uint64_t rtp_stamp_ms = uint64_t(rtp_stamp) * 1000 / sample_rate;
+    if (!_rtp_stamp_ms && !_ntp_stamp_ms) {
+        return rtp_stamp_ms;
+    }
+    if (rtp_stamp_ms > _rtp_stamp_ms) {
+        //时间戳正常增长
+        _last_ret = _ntp_stamp_ms + (rtp_stamp_ms - _rtp_stamp_ms);
+        return _last_ret;
+    }
+    if (_rtp_stamp_ms - rtp_stamp_ms < 10 * 1000) {
+        //小于10秒的时间戳回退，说明收到rtp乱序了
+        return _ntp_stamp_ms - (_rtp_stamp_ms - rtp_stamp_ms);
+    }
+    uint64_t max_rtp_ms = uint64_t(UINT32_MAX) * 1000 / sample_rate;
+    if (rtp_stamp_ms < 60 * 1000 && _rtp_stamp_ms > max_rtp_ms - 60 * 1000) {
+        //确定是时间戳溢出
+        return _ntp_stamp_ms + rtp_stamp_ms + (max_rtp_ms - _rtp_stamp_ms);
+    }
+    //不明原因的时间戳回退，直接返回上次值
+    return _last_ret;
+}
+
 }//namespace mediakit
\ No newline at end of file

src/Common/Stamp.h

@@ -114,6 +114,20 @@ private:
     set<uint32_t> _pts_sorter;
 };
 
+class NtpStamp {
+public:
+    NtpStamp() = default;
+    ~NtpStamp() = default;
+
+    void setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms);
+    uint64_t getNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate);
+
+private:
+    uint64_t _rtp_stamp_ms = 0;
+    uint64_t _ntp_stamp_ms = 0;
+    uint64_t _last_ret = 0;
+};
+
 }//namespace mediakit
 
 #endif //ZLMEDIAKIT_STAMP_H

src/Rtcp/RtcpContext.cpp

@@ -22,7 +22,7 @@ RtcpContext::RtcpContext(bool is_receiver) {
     _is_receiver = is_receiver;
 }
 
-void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint32_t sample_rate, size_t bytes) {
+void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) {
     if (_is_receiver) {
         //接收者才做复杂的统计运算
         auto sys_stamp = getCurrentMillisecond();
@@ -65,6 +65,7 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint32_t sample_rate, size
     ++_packets;
     _bytes += bytes;
     _last_rtp_stamp = stamp;
+    _last_ntp_stamp_ms = ntp_stamp_ms;
 }
 
 void RtcpContext::onRtcp(RtcpHeader *rtcp) {
@@ -154,7 +155,7 @@ Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
         throw std::runtime_error("rtp接收者尝试发送sr包");
     }
     auto rtcp = RtcpSR::create(0);
-    rtcp->setNtpStamp(getCurrentMillisecond(true));
+    rtcp->setNtpStamp(_last_ntp_stamp_ms);
     rtcp->rtpts = htonl(_last_rtp_stamp);
     rtcp->ssrc = htonl(rtcp_ssrc);
     rtcp->packet_count = htonl((uint32_t) _packets);

src/Rtcp/RtcpContext.h

@@ -30,10 +30,11 @@ public:
      * 输出或输入rtp时调用
      * @param seq rtp的seq
      * @param stamp rtp的时间戳，单位采样数(非毫秒)
+     * @param ntp_stamp_ms ntp时间戳
      * @param rtp rtp时间戳采样率，视频一般为90000，音频一般为采样率
      * @param bytes rtp数据长度
      */
-    void onRtp(uint16_t seq, uint32_t stamp, uint32_t sample_rate, size_t bytes);
+    void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
 
     /**
      * 输入sr rtcp包
@@ -110,6 +111,7 @@ private:
     uint16_t _last_rtp_seq = 0;
     //上次的rtp时间戳,毫秒
     uint32_t _last_rtp_stamp = 0;
+    uint64_t _last_ntp_stamp_ms = 0;
     //上次的rtp的系统时间戳(毫秒)用于统计抖动
     uint64_t _last_rtp_sys_stamp = 0;
     //上次统计的丢包总数

src/Rtp/RtpServer.cpp

@@ -35,7 +35,7 @@ public:
     void onRecvRtp(const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len){
         //统计rtp接受情况，用于发送rr包
         auto header = (RtpHeader *) buf->data();
-        onRtp(ntohs(header->seq), ntohl(header->stamp), _sample_rate, buf->size());
+        onRtp(ntohs(header->seq), ntohl(header->stamp), 0/*不发送sr,所以可以设置为0*/ , _sample_rate, buf->size());
         sendRtcp(ntohl(header->ssrc), addr, addr_len);
     }
 

src/Rtsp/RtpReceiver.cpp

@@ -91,12 +91,19 @@ bool RtpTrack::inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t le
     //拷贝rtp
     memcpy(&data[4], ptr, len);
 
+    //设置ntp时间戳
+    rtp->ntp_stamp = _ntp_stamp.getNtpStamp(ntohl(rtp->getHeader()->stamp), sample_rate);
+
     onBeforeRtpSorted(rtp);
     auto seq = rtp->getSeq();
     sortPacket(seq, std::move(rtp));
     return true;
 }
 
+void RtpTrack::setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms){
+    _ntp_stamp.setNtpStamp(rtp_stamp, sample_rate, ntp_stamp_ms);
+}
+
 ////////////////////////////////////////////////////////////////////////////////////
 
 void RtpTrackImp::setOnSorted(OnSorted cb) {

src/Rtsp/RtpReceiver.h

@@ -16,6 +16,7 @@
 #include <memory>
 #include "RtpCodec.h"
 #include "RtspMediaSource.h"
+#include "Common/Stamp.h"
 using namespace std;
 using namespace toolkit;
 
@@ -175,6 +176,7 @@ public:
     void clear();
     uint32_t getSSRC() const;
     bool inputRtp(TrackType type, int sample_rate, uint8_t *ptr, size_t len);
+    void setNtpStamp(uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms);
 
 protected:
     virtual void onRtpSorted(RtpPacket::Ptr rtp) {}
@@ -183,6 +185,7 @@ protected:
 private:
     uint32_t _ssrc = 0;
     Ticker _ssrc_alive;
+    NtpStamp _ntp_stamp;
 };
 
 class RtpTrackImp : public RtpTrack{
@@ -236,6 +239,17 @@ public:
         return _track[index].inputRtp(type, sample_rate, ptr, len);
     }
 
+    /**
+     * 设置ntp时间戳，在收到rtcp sender report时设置
+     * @param index track下标索引
+     * @param rtp_stamp rtp时间戳
+     * @param sample_rate 时间戳采样率
+     * @param ntp_stamp_ms ntp时间戳
+     */
+    void setNtpStamp(int index, uint32_t rtp_stamp, uint32_t sample_rate, uint64_t ntp_stamp_ms){
+        _track[index].setNtpStamp(rtp_stamp, sample_rate, ntp_stamp_ms);
+    }
+
     void clear() {
         for (auto &track : _track) {
             track.clear();

src/Rtsp/Rtsp.cpp

@@ -525,32 +525,40 @@ string RtpHeader::dumpString(size_t rtp_size) const{
 
 ///////////////////////////////////////////////////////////////////////
 
-RtpHeader* RtpPacket::getHeader(){
+RtpHeader *RtpPacket::getHeader() {
     //需除去rtcp over tcp 4个字节长度
-    return (RtpHeader*)(data() + RtpPacket::kRtpTcpHeaderSize);
+    return (RtpHeader *) (data() + RtpPacket::kRtpTcpHeaderSize);
 }
 
-string RtpPacket::dumpString() const{
+const RtpHeader *RtpPacket::getHeader() const {
+    return (RtpHeader *) (data() + RtpPacket::kRtpTcpHeaderSize);
+}
+
+string RtpPacket::dumpString() const {
     return ((RtpPacket *) this)->getHeader()->dumpString(size() - RtpPacket::kRtpTcpHeaderSize);
 }
 
-uint16_t RtpPacket::getSeq(){
+uint16_t RtpPacket::getSeq() const {
     return ntohs(getHeader()->seq);
 }
 
-uint32_t RtpPacket::getStampMS(){
-    return ntohl(getHeader()->stamp) * uint64_t(1000) / sample_rate;
+uint32_t RtpPacket::getStamp() const {
+    return ntohl(getHeader()->stamp);
+}
+
+uint32_t RtpPacket::getStampMS() const {
+    return ntp_stamp & 0xFFFFFFFF;
 }
 
-uint32_t RtpPacket::getSSRC(){
+uint32_t RtpPacket::getSSRC() const {
     return ntohl(getHeader()->ssrc);
 }
 
-uint8_t* RtpPacket::getPayload(){
+uint8_t *RtpPacket::getPayload() {
     return getHeader()->getPayloadData();
 }
 
-size_t RtpPacket::getPayloadSize(){
+size_t RtpPacket::getPayloadSize() const {
     //需除去rtcp over tcp 4个字节长度
     return getHeader()->getPayloadSize(size() - kRtpTcpHeaderSize);
 }

src/Rtsp/Rtsp.h

@@ -154,24 +154,29 @@ public:
 
     //获取rtp头
     RtpHeader* getHeader();
+    const RtpHeader* getHeader() const;
+
     //打印调试信息
     string dumpString() const;
 
     //主机字节序的seq
-    uint16_t getSeq();
+    uint16_t getSeq() const;
+    uint32_t getStamp() const;
     //主机字节序的时间戳，已经转换为毫秒
-    uint32_t getStampMS();
+    uint32_t getStampMS() const;
     //主机字节序的ssrc
-    uint32_t getSSRC();
+    uint32_t getSSRC() const;
     //有效负载，跳过csrc、ext
     uint8_t* getPayload();
     //有效负载长度，不包括csrc、ext、padding
-    size_t getPayloadSize();
+    size_t getPayloadSize() const;
 
     //音视频类型
     TrackType type;
     //音频为采样率，视频一般为90000
     uint32_t sample_rate;
+    //ntp时间戳
+    uint64_t ntp_stamp;
 
     static Ptr create();
 

src/Rtsp/RtspMediaSource.h

@@ -163,7 +163,7 @@ public:
         auto stamp = rtp->getStampMS();
         if (track) {
             track->_seq = rtp->getSeq();
-            track->_time_stamp = stamp;
+            track->_time_stamp = rtp->getStamp() * uint64_t(1000) / rtp->sample_rate;
             track->_ssrc = rtp->getSSRC();
         }
         if (!_ring) {

src/Rtsp/RtspMuxer.cpp

@@ -13,6 +13,36 @@
 
 namespace mediakit {
 
+class RingDelegateHelper : public RingDelegate<RtpPacket::Ptr> {
+public:
+    RingDelegateHelper(RtspMuxer *delegate) {
+        _delegate = delegate;
+    }
+
+    void onWrite(RtpPacket::Ptr in, bool is_key) override {
+        _delegate->onRtp(std::move(in), is_key);
+    }
+
+private:
+    RtspMuxer *_delegate;
+};
+
+void RtspMuxer::onRtp(RtpPacket::Ptr in, bool is_key) {
+    if (_rtp_stamp[in->type] != in->getHeader()->stamp) {
+        //rtp时间戳变化才计算ntp，节省cpu资源
+        int64_t stamp_ms = in->getStamp() * uint64_t(1000) / in->sample_rate;
+        int64_t stamp_ms_inc;
+        //求rtp时间戳增量
+        _stamp[in->type].revise(stamp_ms, stamp_ms, stamp_ms_inc, stamp_ms_inc);
+        _rtp_stamp[in->type] = in->getHeader()->stamp;
+        _ntp_stamp[in->type] = stamp_ms_inc + _ntp_stamp_start;
+    }
+
+    //rtp拦截入口，此处统一赋值ntp
+    in->ntp_stamp = _ntp_stamp[in->type];
+    _rtpRing->write(std::move(in), is_key);
+}
+
 RtspMuxer::RtspMuxer(const TitleSdp::Ptr &title){
     if(!title){
         _sdp = std::make_shared<TitleSdp>()->getSdp();
@@ -20,6 +50,9 @@ RtspMuxer::RtspMuxer(const TitleSdp::Ptr &title){
         _sdp = title->getSdp();
     }
     _rtpRing = std::make_shared<RtpRing::RingType>();
+    _rtpInterceptor = std::make_shared<RtpRing::RingType>();
+    _rtpInterceptor->setDelegate(std::make_shared<RingDelegateHelper>(this));
+    _ntp_stamp_start = getCurrentMillisecond(true);
 }
 
 void RtspMuxer::addTrack(const Track::Ptr &track) {
@@ -36,15 +69,23 @@ void RtspMuxer::addTrack(const Track::Ptr &track) {
     }
 
     //设置rtp输出环形缓存
-    encoder->setRtpRing(_rtpRing);
+    encoder->setRtpRing(_rtpInterceptor);
 
     //添加其sdp
     _sdp.append(sdp->getSdp());
+    trySyncTrack();
+}
+
+void RtspMuxer::trySyncTrack() {
+    if (_encoder[TrackAudio] && _encoder[TrackVideo]) {
+        //音频时间戳同步于视频，因为音频时间戳被修改后不影响播放
+        _stamp[TrackAudio].syncTo(_stamp[TrackVideo]);
+    }
 }
 
 void RtspMuxer::inputFrame(const Frame::Ptr &frame) {
     auto &encoder = _encoder[frame->getTrackType()];
-    if(encoder){
+    if (encoder) {
         encoder->inputFrame(frame);
     }
 }

src/Rtsp/RtspMuxer.h

@@ -13,6 +13,7 @@
 
 #include "Extension/Frame.h"
 #include "Common/MediaSink.h"
+#include "Common/Stamp.h"
 #include "RtpCodec.h"
 
 namespace mediakit{
@@ -21,7 +22,8 @@ namespace mediakit{
 */
 class RtspMuxer : public MediaSinkInterface{
 public:
-    typedef std::shared_ptr<RtspMuxer> Ptr;
+    friend class RingDelegateHelper;
+    using Ptr = std::shared_ptr<RtspMuxer>;
 
     /**
      * 构造函数
@@ -57,10 +59,21 @@ public:
      * 重置所有track
      */
     void resetTracks() override ;
+
+private:
+    void onRtp(RtpPacket::Ptr in, bool is_key);
+    void computeNtp(const Frame::Ptr &frame);
+    void trySyncTrack();
+
 private:
+    uint32_t _rtp_stamp[TrackMax]{0};
+    uint64_t _ntp_stamp[TrackMax]{0};
+    uint64_t _ntp_stamp_start;
     string _sdp;
+    Stamp _stamp[TrackMax];
     RtpCodec::Ptr _encoder[TrackMax];
     RtpRing::RingType::Ptr _rtpRing;
+    RtpRing::RingType::Ptr _rtpInterceptor;
 };
 
 

src/Rtsp/RtspPlayer.cpp

@@ -488,6 +488,11 @@ void RtspPlayer::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, uint8_t *data
     auto rtcp_arr = RtcpHeader::loadFromBytes((char *) data, len);
     for (auto &rtcp : rtcp_arr) {
         _rtcp_context[track_idx]->onRtcp(rtcp);
+        if ((RtcpType) rtcp->pt == RtcpType::RTCP_SR) {
+            auto sr = (RtcpSR *) (rtcp);
+            //设置rtp时间戳与ntp时间戳的对应关系
+            setNtpStamp(track_idx, sr->rtpts, track->_samplerate, sr->getNtpUnixStampMS());
+        }
     }
 }
 
@@ -591,7 +596,7 @@ void RtspPlayer::sendRtspRequest(const string &cmd, const string &url,const StrC
 
 void RtspPlayer::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_idx){
     auto &rtcp_ctx = _rtcp_context[track_idx];
-    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_ticker[track_idx];
     if (ticker.elapsedTime() < 3 * 1000) {

src/Rtsp/RtspPusher.cpp

@@ -360,7 +360,7 @@ void RtspPusher::updateRtcpContext(const RtpPacket::Ptr &rtp){
     int track_index = getTrackIndexByTrackType(rtp->type);
     auto &ticker = _rtcp_send_ticker[track_index];
     auto &rtcp_ctx = _rtcp_context[track_index];
-    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     //send rtcp every 5 second
     if (ticker.elapsedTime() > 5 * 1000) {

src/Rtsp/RtspSession.cpp

@@ -186,6 +186,11 @@ void RtspSession::onRtcpPacket(int track_idx, SdpTrack::Ptr &track, const char *
     auto rtcp_arr = RtcpHeader::loadFromBytes((char *) data, len);
     for (auto &rtcp : rtcp_arr) {
         _rtcp_context[track_idx]->onRtcp(rtcp);
+        if ((RtcpType) rtcp->pt == RtcpType::RTCP_SR) {
+            auto sr = (RtcpSR *) (rtcp);
+            //设置rtp时间戳与ntp时间戳的对应关系
+            setNtpStamp(track_idx, sr->rtpts, track->_samplerate, sr->getNtpUnixStampMS());
+        }
     }
 }
 
@@ -1126,12 +1131,14 @@ void RtspSession::onBeforeRtpSorted(const RtpPacket::Ptr &rtp, int track_index){
 void RtspSession::updateRtcpContext(const RtpPacket::Ptr &rtp){
     int track_index = getTrackIndexByTrackType(rtp->type);
     auto &rtcp_ctx = _rtcp_context[track_index];
-    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_tickers[track_index];
     //send rtcp every 5 second
-    if (ticker.elapsedTime() > 5 * 1000) {
+    if (ticker.elapsedTime() > 5 * 1000 || (_send_sr_rtcp[track_index] && !_push_src)) {
+        //确保在发送rtp前，先发送一次sender report rtcp(用于播放器同步音视频)
         ticker.resetTime();
+        _send_sr_rtcp[track_index] = false;
 
         static auto send_rtcp = [](RtspSession *thiz, int index, Buffer::Ptr ptr) {
             if (thiz->_rtp_type == Rtsp::RTP_TCP) {

src/Rtsp/RtspSession.h

@@ -215,6 +215,7 @@ private:
     Ticker _rtcp_send_tickers[2];
     //统计rtp并发送rtcp
     vector<RtcpContext::Ptr> _rtcp_context;
+    bool _send_sr_rtcp[2] = {true, true};
 };
 
 /**

webrtc/WebRtcTransport.cpp

@@ -600,8 +600,6 @@ public:
             auto seq = ntohs(rtp->seq);
             //统计rtp接受情况，便于生成nack rtcp包
             _nack_ctx.received(seq);
-            //统计rtp收到的情况，好做rr汇报
-            _rtcp_context.onRtp(seq, ntohl(rtp->stamp), sample_rate, len);
         }
         return RtpTrack::inputRtp(type, sample_rate, ptr, len);
     }
@@ -611,6 +609,14 @@ public:
         return _rtcp_context.createRtcpRR(ssrc, getSSRC());
     }
 
+protected:
+    void onBeforeRtpSorted(const RtpPacket::Ptr &rtp) override {
+        //统计rtp收到的情况，好做rr汇报
+        _rtcp_context.onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate,
+                            rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+        RtpTrackImp::onBeforeRtpSorted(rtp);
+    }
+
 private:
     NackContext _nack_ctx;
     RtcpContext _rtcp_context{true};
@@ -639,6 +645,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) {
                     if(!rtp_chn){
                         WarnL << "未识别的sr rtcp包:" << rtcp->dumpString();
                     } else {
+                        //设置rtp时间戳与ntp时间戳的对应关系
+                        rtp_chn->setNtpStamp(sr->rtpts, track->plan_rtp->sample_rate, sr->getNtpUnixStampMS());
                         auto rr = rtp_chn->createRtcpRR(sr, track->answer_ssrc_rtp);
                         sendRtcpPacket(rr->data(), rr->size(), true);
                     }
@@ -845,7 +853,7 @@ void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool r
     }
     if (!rtx) {
         //统计rtp发送情况，好做sr汇报
-        track->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
+        track->rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStamp(), rtp->ntp_stamp, rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
         track->nack_list.push_back(rtp);
 #if 0
         //此处模拟发送丢包