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修复webrtc simulcast推流崩溃的bug
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@ -1 +1 @@
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Subproject commit 36d1122f42ab6b92d0ca8f57df0462acc632efc7
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Subproject commit a827f0bdcb4221c19ab1a243ebd1f01fa7dc50f2
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@ -436,8 +436,11 @@ void WebRtcTransportImp::onStartWebRTC() {
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if (m_offer->type != TrackApplication) {
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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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//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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track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
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track->rtp_ext_ctx = std::make_shared<RtpExtContext>(*m_offer);
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track->rtp_ext_ctx->setOnGetRtp([this, &track](uint8_t pt, uint32_t ssrc, const string &rid) {
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weak_ptr<MediaTrack> weak_track = track;
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track->rtp_ext_ctx->setOnGetRtp([this, weak_track](uint8_t pt, uint32_t ssrc, const string &rid) {
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//ssrc --> MediaTrack
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//ssrc --> MediaTrack
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auto track = weak_track.lock();
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assert(track);
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_ssrc_to_track[ssrc] = track;
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_ssrc_to_track[ssrc] = track;
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InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
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InfoL << "get rtp, pt:" << (int) pt << ", ssrc:" << ssrc << ", rid:" << rid;
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});
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});
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