修复rtcp rr/sr时间戳转换相关问题,计算rtt

src/Rtcp/Rtcp.cpp

@@ -263,11 +263,25 @@ string RtcpSR::getNtpStamp() const{
     return LogChannel::printTime(tv);
 }
 
+uint64_t RtcpSR::getNtpUnixStampMS() const {
+    struct timeval tv;
+    tv.tv_sec = ntpmsw - 0x83AA7E80;
+    tv.tv_usec = (decltype(tv.tv_usec)) (ntplsw / ((double) (((uint64_t) 1) << 32) * 1.0e-6));
+    return 1000 * tv.tv_sec + tv.tv_usec / 1000;
+}
+
 void RtcpSR::setNtpStamp(struct timeval tv) {
     ntpmsw = htonl(tv.tv_sec + 0x83AA7E80); /* 0x83AA7E80 is the number of seconds from 1900 to 1970 */
     ntplsw = htonl((uint32_t) ((double) tv.tv_usec * (double) (((uint64_t) 1) << 32) * 1.0e-6));
 }
 
+void RtcpSR::setNtpStamp(uint64_t unix_stamp_ms) {
+    struct timeval tv;
+    tv.tv_sec = unix_stamp_ms / 1000;
+    tv.tv_usec = (unix_stamp_ms % 1000) * 1000;
+    setNtpStamp(tv);
+}
+
 string RtcpSR::dumpString() const{
     _StrPrinter printer;
     printer << RtcpHeader::dumpHeader();

src/Rtcp/Rtcp.h

@@ -347,12 +347,14 @@ public:
      * @param tv 时间
      */
     void setNtpStamp(struct timeval tv);
+    void setNtpStamp(uint64_t unix_stamp_ms);
 
     /**
      * 返回ntp时间的字符串
      * 使用net2Host转换成主机字节序后才可使用此函数
      */
     string getNtpStamp() const;
+    uint64_t getNtpUnixStampMS() const;
 
     /**
      * 获取ReportItem对象指针列表

src/Rtcp/RtcpContext.cpp

@@ -22,14 +22,14 @@ RtcpContext::RtcpContext(bool is_receiver) {
     _is_receiver = is_receiver;
 }
 
-void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, size_t bytes) {
+void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, 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();
         if (_last_rtp_sys_stamp) {
             //计算时间戳抖动值
-            double diff = double(
-                    int64_t(sys_stamp) - int64_t(_last_rtp_sys_stamp) - int64_t(stamp) + int64_t(_last_rtp_stamp));
+            double diff = double((int64_t(sys_stamp) - int64_t(_last_rtp_sys_stamp)) * (sample_rate / double(1000.0))
+                                 - (int64_t(stamp) - int64_t(_last_rtp_stamp)));
             if (diff < 0) {
                 diff = -diff;
             }
@@ -68,23 +68,49 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, size_t bytes) {
 }
 
 void RtcpContext::onRtcp(RtcpHeader *rtcp) {
-    if ((RtcpType) rtcp->pt != RtcpType::RTCP_SR) {
-        return;
-    }
-    if (!_is_receiver) {
-        WarnL << "rtp发送者收到sr包";
-        return;
+    switch ((RtcpType) rtcp->pt) {
+        case RtcpType::RTCP_SR: {
+            auto rtcp_sr = (RtcpSR *) rtcp;
+            /**
+             last SR timestamp (LSR): 32 bits
+              The middle 32 bits out of 64 in the NTP timestamp (as explained in
+              Section 4) received as part of the most recent RTCP sender report
+              (SR) packet from source SSRC_n.  If no SR has been received yet,
+              the field is set to zero.
+             */
+            _last_sr_lsr = ((rtcp_sr->ntpmsw & 0xFFFF) << 16) | ((rtcp_sr->ntplsw >> 16) & 0xFFFF);
+            _last_sr_ntp_sys = getCurrentMillisecond();
+            break;
+        }
+        case RtcpType::RTCP_RR: {
+            auto rtcp_rr = (RtcpRR *) rtcp;
+            for (auto item :  rtcp_rr->getItemList()) {
+                if (!item->last_sr_stamp) {
+                    continue;
+                }
+                //rtp接收端收到sr包后，回复rr包的延时，已转换为毫秒
+                auto delay_ms = (uint64_t) item->delay_since_last_sr * 1000 / 65536;
+                //这个rr包对应sr包的ntpmsw和ntplsw
+                auto ntpmsw = item->last_sr_stamp >> 16;
+                auto ntplsw = (item->last_sr_stamp & 0xFFFF) << 16;
+                RtcpSR sr;
+                //获取当前时间戳
+                sr.setNtpStamp(getCurrentMillisecond(true));
+
+                //当前时间戳与上次发送的sr包直接的ntp时间差
+                int64_t ntpmsw_inc = (int64_t)(ntohl(sr.ntpmsw) & 0xFFFF) - (int64_t)ntpmsw;
+                int64_t ntplsw_inc = (int64_t)(ntohl(sr.ntplsw)) - (int64_t)ntplsw;
+
+                //转换为毫秒
+                auto ms_inc = ntpmsw_inc * 1000 + (ntplsw_inc / ((double) (((uint64_t) 1) << 32) * 1.0e-3));
+                auto rtt = (int) ((ms_inc - delay_ms) / 2);
+                _rtt[item->ssrc] = rtt;
+                //InfoL << "ssrc:" << item->ssrc << ",rtt:" << rtt;
+            }
+            break;
+        }
+        default: break;
     }
-    auto rtcp_sr = (RtcpSR *) (rtcp);
-    /**
-     last SR timestamp (LSR): 32 bits
-      The middle 32 bits out of 64 in the NTP timestamp (as explained in
-      Section 4) received as part of the most recent RTCP sender report
-      (SR) packet from source SSRC_n.  If no SR has been received yet,
-      the field is set to zero.
-     */
-    _last_sr_lsr = ((rtcp_sr->ntpmsw & 0xFFFF) << 16) | ((rtcp_sr->ntplsw >> 16) & 0xFFFF);
-    _last_sr_ntp_sys = getCurrentMillisecond();
 }
 
 size_t RtcpContext::getExpectedPackets() const {
@@ -120,14 +146,9 @@ 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->ssrc = htonl(rtcp_ssrc);
-
-    struct timeval tv;
-    gettimeofday(&tv, NULL);
-    rtcp->setNtpStamp(tv);
-
-    //转换成rtp时间戳
+    rtcp->setNtpStamp(getCurrentMillisecond(true));
     rtcp->rtpts = htonl(_last_rtp_stamp);
+    rtcp->ssrc = htonl(rtcp_ssrc);
     rtcp->packet_count = htonl((uint32_t) _packets);
     rtcp->octet_count = htonl((uint32_t) _bytes);
     return RtcpHeader::toBuffer(std::move(rtcp));

src/Rtcp/RtcpContext.h

@@ -30,9 +30,10 @@ 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 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, size_t bytes);
+    void onRtp(uint16_t seq, uint32_t stamp, uint32_t sample_rate, size_t bytes);
 
     /**
      * 输入sr rtcp包
@@ -112,6 +113,7 @@ private:
     uint32_t _last_sr_lsr = 0;
     //上次收到sr时的系统时间戳,单位毫秒
     uint64_t _last_sr_ntp_sys = 0;
+    unordered_map<uint32_t/*ssrc*/, uint32_t/*rtt*/> _rtt;
 };
 
 }//namespace mediakit

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), buf->size());
+        onRtp(ntohs(header->seq), ntohl(header->stamp), _sample_rate, buf->size());
         sendRtcp(ntohl(header->ssrc), addr, addr_len);
     }
 

src/Rtsp/RtspPlayer.cpp

@@ -591,7 +591,7 @@ 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->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->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->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     //send rtcp every 5 second
     if (ticker.elapsedTime() > 5 * 1000) {

src/Rtsp/RtspSession.cpp

@@ -1126,7 +1126,7 @@ 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->size() - RtpPacket::kRtpTcpHeaderSize);
+    rtcp_ctx->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
 
     auto &ticker = _rtcp_send_tickers[track_index];
     //send rtcp every 5 second

webrtc/WebRtcTransport.cpp

@@ -586,7 +586,7 @@ public:
             //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), len);
+            _rtcp_context.onRtp(seq, ntohl(rtp->stamp), sample_rate, len);
         }
         return RtpTrack::inputRtp(type, sample_rate, ptr, len);
     }
@@ -825,7 +825,7 @@ 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->size() - RtpPacket::kRtpTcpHeaderSize);
+        track->rtcp_context_send->onRtp(rtp->getSeq(), ntohl(rtp->getHeader()->stamp), rtp->sample_rate, rtp->size() - RtpPacket::kRtpTcpHeaderSize);
         track->nack_list.push_back(rtp);
 #if 0
         //此处模拟发送丢包