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https://github.com/ZLMediaKit/ZLMediaKit.git
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refactor: 梳理 rtx 处理逻辑
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@ -385,10 +385,10 @@ void WebRtcTransportImp::onStartWebRTC() {
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_ssrc_to_track[track->offer_ssrc_rtx] = track;
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//rtp pt --> MediaTrack
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_pt_to_track.emplace(track->plan_rtp->pt, std::make_pair(false, track));
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_pt_to_track.emplace(track->plan_rtp->pt, new WrappedRtpTrack(track, _twcc_ctx, *this));
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if (track->plan_rtx) {
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//rtx pt --> MediaTrack
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_pt_to_track.emplace(track->plan_rtx->pt, std::make_pair(true, track));
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_pt_to_track.emplace(track->plan_rtx->pt, new WrappedRtxTrack(track));
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}
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if (m_offer->type != TrackApplication) {
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//记录rtp ext类型与id的关系,方便接收或发送rtp时修改rtp ext id
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@ -691,28 +691,25 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
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WarnL << "unknown rtp pt:" << (int)rtp->pt;
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return;
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}
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bool is_rtx = it->second.first;
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it->second->inputRtp(buf, len, stamp_ms, rtp);
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}
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void WrappedRtpTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
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auto ssrc = ntohl(rtp->ssrc);
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auto &track = it->second.second;
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//修改ext id至统一
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string rid;
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auto twcc_ext = track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
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if (twcc_ext && !is_rtx) {
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if (twcc_ext) {
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_twcc_ctx.onRtp(ssrc, twcc_ext.getTransportCCSeq(), stamp_ms);
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}
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auto &ref = track->rtp_channel[rid];
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if (!ref) {
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if (is_rtx) {
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//再接收到对应的rtp前,丢弃rtx包
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WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ssrc << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
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return;
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}
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createRtpChannel(rid, ssrc, *track);
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_transport.createRtpChannel(rid, ssrc, *track);
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}
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if (!is_rtx) {
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//这是普通的rtp数据
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#if 0
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auto seq = ntohs(rtp->seq);
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@ -723,11 +720,22 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
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#endif
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//解析并排序rtp
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ref->inputRtp(track->media->type, track->plan_rtp->sample_rate, (uint8_t *) buf, len, false);
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}
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void WrappedRtxTrack::inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) {
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//修改ext id至统一
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string rid;
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track->rtp_ext_ctx->changeRtpExtId(rtp, true, &rid, RtpExtType::transport_cc);
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auto &ref = track->rtp_channel[rid];
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if (!ref) {
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//再接收到对应的rtp前,丢弃rtx包
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WarnL << "unknown rtx rtp, rid:" << rid << ", ssrc:" << ntohl(rtp->ssrc) << ", codec:" << track->plan_rtp->codec << ", seq:" << ntohs(rtp->seq);
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return;
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}
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//这里是rtx重传包
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//https://datatracker.ietf.org/doc/html/rfc4588#section-4
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// https://datatracker.ietf.org/doc/html/rfc4588#section-4
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auto payload = rtp->getPayloadData();
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auto size = rtp->getPayloadSize(len);
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if (size < 2) {
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@ -736,7 +744,7 @@ void WebRtcTransportImp::onRtp(const char *buf, size_t len, uint64_t stamp_ms) {
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//前两个字节是原始的rtp的seq
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auto origin_seq = payload[0] << 8 | payload[1];
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//rtx 转换为 rtp
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// rtx 转换为 rtp
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rtp->pt = track->plan_rtp->pt;
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rtp->seq = htons(origin_seq);
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rtp->ssrc = htonl(ref->getSSRC());
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@ -189,6 +189,33 @@ public:
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std::shared_ptr<RtpChannel> getRtpChannel(uint32_t ssrc) const;
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};
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struct WrappedMediaTrack {
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MediaTrack::Ptr track;
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explicit WrappedMediaTrack(MediaTrack::Ptr ptr): track(ptr) {}
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virtual ~WrappedMediaTrack() {}
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virtual void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) = 0;
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};
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struct WrappedRtxTrack: public WrappedMediaTrack {
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explicit WrappedRtxTrack(MediaTrack::Ptr ptr)
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: WrappedMediaTrack(std::move(ptr)) {}
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void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
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};
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class WebRtcTransportImp;
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struct WrappedRtpTrack : public WrappedMediaTrack {
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explicit WrappedRtpTrack(MediaTrack::Ptr ptr, TwccContext& twcc, WebRtcTransportImp& t)
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: WrappedMediaTrack(std::move(ptr))
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, _twcc_ctx(twcc)
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, _transport(t) {}
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TwccContext& _twcc_ctx;
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WebRtcTransportImp& _transport;
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void inputRtp(const char *buf, size_t len, uint64_t stamp_ms, RtpHeader *rtp) override;
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};
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class WebRtcTransportImp : public WebRtcTransport {
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public:
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using Ptr = std::shared_ptr<WebRtcTransportImp>;
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@ -202,6 +229,8 @@ public:
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bool canRecvRtp() const;
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void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
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void createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track);
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protected:
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WebRtcTransportImp(const EventPoller::Ptr &poller);
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void onStartWebRTC() override;
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@ -224,7 +253,7 @@ private:
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void onSortedRtp(MediaTrack &track, const string &rid, RtpPacket::Ptr rtp);
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void onSendNack(MediaTrack &track, const FCI_NACK &nack, uint32_t ssrc);
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void onSendTwcc(uint32_t ssrc, const string &twcc_fci);
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void createRtpChannel(const string &rid, uint32_t ssrc, MediaTrack &track);
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void registerSelf();
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void unregisterSelf();
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void unrefSelf();
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@ -251,7 +280,7 @@ private:
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//根据rtcp的ssrc获取相关信息,收发rtp和rtx的ssrc都会记录
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unordered_map<uint32_t/*ssrc*/, MediaTrack::Ptr> _ssrc_to_track;
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//根据接收rtp的pt获取相关信息
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unordered_map<uint8_t/*pt*/, std::pair<bool/*is rtx*/,MediaTrack::Ptr> > _pt_to_track;
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unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
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};
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class WebRtcTransportManager {
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