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fix(RtpSender): 修复 SSRC 解析溢出问题 (#4646)
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@ -51,9 +51,9 @@ void RtpSender::startSend(const MediaSourceEvent &sender, const MediaSourceEvent
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// Do not recreate the object when reconnecting
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// Do not recreate the object when reconnecting
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auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
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auto lam = [this](std::shared_ptr<List<Buffer::Ptr>> list) { onFlushRtpList(std::move(list)); };
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switch (args.data_type) {
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switch (args.data_type) {
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case MediaSourceEvent::SendRtpArgs::kRtpPS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, true); break;
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case MediaSourceEvent::SendRtpArgs::kRtpPS: _interface = std::make_shared<RtpCachePS>(lam, stoll(args.ssrc), args.pt, true); break;
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case MediaSourceEvent::SendRtpArgs::kRtpTS: _interface = std::make_shared<RtpCachePS>(lam, atoi(args.ssrc.data()), args.pt, false); break;
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case MediaSourceEvent::SendRtpArgs::kRtpTS: _interface = std::make_shared<RtpCachePS>(lam, stoll(args.ssrc), args.pt, false); break;
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case MediaSourceEvent::SendRtpArgs::kRtpES: _interface = std::make_shared<RtpCacheRaw>(lam, atoi(args.ssrc.data()), args.pt, args.only_audio); break;
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case MediaSourceEvent::SendRtpArgs::kRtpES: _interface = std::make_shared<RtpCacheRaw>(lam, stoll(args.ssrc), args.pt, args.only_audio); break;
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default: CHECK(0, "invalid rtp type: " + to_string(args.data_type)); break;
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default: CHECK(0, "invalid rtp type: " + to_string(args.data_type)); break;
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}
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}
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}
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}
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@ -386,7 +386,7 @@ void RtpSender::onSendRtpUdp(const toolkit::Buffer::Ptr &buf, bool check) {
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_rtcp_send_ticker.resetTime();
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_rtcp_send_ticker.resetTime();
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// rtcp ssrc为rtp ssrc + 1 [AUTO-TRANSLATED:318fada3]
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// rtcp ssrc为rtp ssrc + 1 [AUTO-TRANSLATED:318fada3]
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// rtcp ssrc is rtp ssrc + 1
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// rtcp ssrc is rtp ssrc + 1
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auto sr = _rtcp_context->createRtcpSR(atoi(_args.ssrc.data()) + 1);
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auto sr = _rtcp_context->createRtcpSR(stoll(_args.ssrc) + 1);
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// send sender report rtcp
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// send sender report rtcp
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_socket_rtcp->send(sr);
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_socket_rtcp->send(sr);
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}
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}
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