mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-11-22 10:40:05 +08:00
完善ssrc处理逻辑
This commit is contained in:
xia-chu
parent
d4ce5b0091
commit
a42e5f6470
@ -185,6 +185,9 @@ void WebRtcTransport::onCheckSdp(SdpType type, RtcSession &sdp){
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if (sdp.group.mids.empty()) {
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throw std::invalid_argument("只支持group BUNDLE模式");
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}
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if (type == SdpType::offer) {
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sdp.checkValidSSRC();
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}
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}
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void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
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@ -390,6 +393,16 @@ bool WebRtcTransportImp::canRecvRtp() const{
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return _push_src && (sdp.media[0].direction == RtpDirection::sendrecv || sdp.media[0].direction == RtpDirection::recvonly);
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}
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const RtcSession& WebRtcTransportImp::getSdpWithSSRC() const{
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auto &offer = getSdp(SdpType::offer);
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if (offer.haveSSRC()) {
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return offer;
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}
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auto &answer = getSdp(SdpType::answer);
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CHECK(answer.haveSSRC());
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return answer;
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}
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void WebRtcTransportImp::onStartWebRTC() {
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//获取ssrc和pt相关信息,届时收到rtp和rtcp时分别可以根据pt和ssrc找到相关的信息
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for (auto &m : getSdp(SdpType::offer).media) {
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@ -398,11 +411,12 @@ void WebRtcTransportImp::onStartWebRTC() {
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if (!hit_pan) {
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continue;
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}
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auto m_with_ssrc = getSdpWithSSRC().getMedia(m.type);
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//获取offer端rtp的ssrc和pt相关信息
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auto &ref = _rtp_info_pt[plan.pt];
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_rtp_info_ssrc[m.rtp_rtx_ssrc[0].ssrc] = &ref;
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_rtp_info_ssrc[m_with_ssrc->rtp_rtx_ssrc[0].ssrc] = &ref;
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ref.plan = &plan;
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ref.media = &m;
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ref.media = m_with_ssrc;
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ref.is_common_rtp = getCodecId(plan.codec) != CodecInvalid;
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ref.rtcp_context_recv = std::make_shared<RtcpContext>(ref.plan->sample_rate, true);
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ref.rtcp_context_send = std::make_shared<RtcpContext>(ref.plan->sample_rate, false);
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@ -441,14 +455,16 @@ void WebRtcTransportImp::onStartWebRTC() {
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RtcSession rtsp_send_sdp;
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rtsp_send_sdp.loadFrom(_play_src->getSdp(), false);
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for (auto &m : getSdp(SdpType::answer).media) {
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for (auto &m : getSdp(SdpType::answer).media) {
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if (m.type == TrackApplication) {
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continue;
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}
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auto rtsp_media = rtsp_send_sdp.getMedia(m.type);
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if (rtsp_media && getCodecId(rtsp_media->plan[0].codec) == getCodecId(m.plan[0].codec)) {
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//记录发送rtp时约定的pt，届时发送rtp时需要修改pt
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_send_rtp_pt[m.type] = m.plan[0].pt;
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auto it = _rtp_info_pt.find(m.plan[0].pt);
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CHECK(it != _rtp_info_pt.end());
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//记录发送rtp时约定的信息，届时发送rtp时需要修改pt和ssrc
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_send_rtp_info[m.type] = &it->second;
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}
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}
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}
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@ -457,10 +473,11 @@ void WebRtcTransportImp::onStartWebRTC() {
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void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
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WebRtcTransport::onCheckSdp(type, sdp);
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if (type != SdpType::answer) {
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//我们只修改answer sdp
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return;
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}
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//修改sdp的ip、端口信息
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//修改answer sdp的ip、端口信息
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GET_CONFIG(string, extern_ip, RTC::kExternIP);
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for (auto &m : sdp.media) {
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m.addr.reset();
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@ -472,7 +489,8 @@ void WebRtcTransportImp::onCheckSdp(SdpType type, RtcSession &sdp){
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sdp.origin.address = m.addr.address;
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}
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if (!canSendRtp()) {
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if (!canSendRtp() || getSdp(SdpType::offer).haveSSRC()) {
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//offer sdp未包含ssrc相关信息，那么我们才在answer sdp中回复ssrc相关信息
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return;
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}
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@ -686,20 +704,23 @@ void WebRtcTransportImp::onBeforeSortedRtp(const RtpPayloadInfo &info, const Rtp
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}
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void WebRtcTransportImp::onSendRtp(const RtpPacket::Ptr &rtp, bool flush){
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auto pt = _send_rtp_pt[rtp->type];
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if (pt == 0xFF) {
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auto info = _send_rtp_info[rtp->type];
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if (!info) {
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//忽略，对方不支持该编码类型
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return;
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}
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_bytes_usage += rtp->size() - RtpPacket::kRtpTcpHeaderSize;
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sendRtpPacket(rtp->data() + RtpPacket::kRtpTcpHeaderSize, rtp->size() - RtpPacket::kRtpTcpHeaderSize, flush, rtp->type);
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//统计rtp发送情况，好做sr汇报
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_rtp_info_pt[pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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_rtp_info_pt[info->plan->pt].rtcp_context_send->onRtp(rtp->getSeq(), rtp->getStampMS(), rtp->size() - RtpPacket::kRtpTcpHeaderSize);
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}
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void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t len, TrackType type) {
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auto header = (RtpHeader *)buf;
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header->pt = _send_rtp_pt[type];
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auto info = _send_rtp_info[type];
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//修改目标pt和ssrc
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header->pt = info->plan->pt;
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header->ssrc = htons(info->media->rtp_rtx_ssrc[0].ssrc);
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changeRtpExtId(header, _rtp_ext_type_to_id);
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}
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@ -188,6 +188,7 @@ private:
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SdpAttrCandidate::Ptr getIceCandidate() const;
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bool canSendRtp() const;
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bool canRecvRtp() const;
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const RtcSession& getSdpWithSSRC() const;
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class RtpPayloadInfo {
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public:
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@ -215,8 +216,8 @@ private:
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Ticker _alive_ticker;
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//pli rtcp计时器
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Ticker _pli_ticker;
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//记录协商的rtp的pt类型
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uint8_t _send_rtp_pt[2] = {0xFF, 0xFF};
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//记录协商的发送rtp的pt和ssrc
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RtpPayloadInfo* _send_rtp_info[2] = {nullptr, nullptr};
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//复合udp端口，接收一切rtp与rtcp
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Socket::Ptr _socket;
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//推流的rtsp源
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@ -226,9 +227,9 @@ private:
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//播放rtsp源的reader对象
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RtspMediaSource::RingType::RingReader::Ptr _reader;
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//根据rtp的pt获取相关信息
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unordered_map<uint8_t, RtpPayloadInfo> _rtp_info_pt;
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//根据推流端rtcp的ssrc获取相关信息
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unordered_map<uint32_t, RtpPayloadInfo*> _rtp_info_ssrc;
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unordered_map<uint8_t/*pt*/, RtpPayloadInfo> _rtp_info_pt;
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//根据rtcp的ssrc获取相关信息
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unordered_map<uint32_t/*ssrc*/, RtpPayloadInfo*> _rtp_info_ssrc;
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//发送rtp时需要修改rtp ext id
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map<RtpExtType, uint8_t> _rtp_ext_type_to_id;
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//接收rtp时需要修改rtp ext id
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