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Don't recreate the audio receive stream when updating the local_ssrc.

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Bug: webrtc:11993
Change-Id: Ic5d8a8a8b7c12fb1d906e0b3cbdf657fd9e8eafc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222042
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34299}
This commit is contained in:
Tommi
2021-06-15 23:01:57 +02:00
committed by WebRTC LUCI CQ
parent bc03259de7
commit 08be9baaa3
22 changed files with 266 additions and 34 deletions
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14
modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
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@ -69,6 +69,7 @@ ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
rtt_ms_(0) {
RTC_DCHECK(worker_queue_);
process_thread_checker_.Detach();
packet_sequence_checker_.Detach();
if (!configuration.receiver_only) {
rtp_sender_ = std::make_unique<RtpSenderContext>(configuration);
// Make sure rtcp sender use same timestamp offset as rtp sender.
@ -169,6 +170,7 @@ absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
void ModuleRtpRtcpImpl2::IncomingRtcpPacket(const uint8_t* rtcp_packet,
const size_t length) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtcp_receiver_.IncomingPacket(rtcp_packet, length);
}
@ -219,6 +221,12 @@ RtpState ModuleRtpRtcpImpl2::GetRtxState() const {
return rtp_sender_->packet_generator.GetRtxRtpState();
}
uint32_t ModuleRtpRtcpImpl2::local_media_ssrc() const {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
RTC_DCHECK_EQ(rtcp_receiver_.local_media_ssrc(), rtcp_sender_.SSRC());
return rtcp_receiver_.local_media_ssrc();
}
void ModuleRtpRtcpImpl2::SetRid(const std::string& rid) {
if (rtp_sender_) {
rtp_sender_->packet_generator.SetRid(rid);
@ -650,6 +658,12 @@ void ModuleRtpRtcpImpl2::SetRemoteSSRC(const uint32_t ssrc) {
rtcp_receiver_.SetRemoteSSRC(ssrc);
}
void ModuleRtpRtcpImpl2::SetLocalSsrc(uint32_t local_ssrc) {
RTC_DCHECK_RUN_ON(&packet_sequence_checker_);
rtcp_receiver_.set_local_media_ssrc(local_ssrc);
rtcp_sender_.SetSsrc(local_ssrc);
}
RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
// Typically called on the `rtp_transport_queue_` owned by an
// RtpTransportControllerSendInterface instance.