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Delete unnesseccary Call::RegisterReceiveStream and Call::DeregisterReceiveStream methods.

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Bug: webrtc:7135
Change-Id: I12e417b9bc5ed8bfae64e4591c37f882ead04092
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291481
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40959}
This commit is contained in:
Per K 2023-10-18 10:02:53 +02:00 committed by WebRTC LUCI CQ
parent 8ef094f66a
commit 8c1f122452
1 changed files with 0 additions and 53 deletions
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53
call/call.cc
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@ -388,11 +388,6 @@ class Call final : public webrtc::Call,
RTC_NO_UNIQUE_ADDRESS SequenceChecker receive_11993_checker_;
// TODO(bugs.webrtc.org/11993): Move receive_rtp_config_ over to the
// network thread.
std::map<uint32_t, ReceiveStreamInterface*> receive_rtp_config_
RTC_GUARDED_BY(&receive_11993_checker_);
// Audio and Video send streams are owned by the client that creates them.
// TODO(bugs.webrtc.org/11993): `audio_send_ssrcs_` and `video_send_ssrcs_`
// should be accessed on the network thread.
@ -846,11 +841,6 @@ webrtc::AudioReceiveStreamInterface* Call::CreateAudioReceiveStream(
// to live on the network thread.
receive_stream->RegisterWithTransport(&audio_receiver_controller_);
// TODO(bugs.webrtc.org/11993): Update the below on the network thread.
// We could possibly set up the audio_receiver_controller_ association up
// as part of the async setup.
RegisterReceiveStream(config.rtp.remote_ssrc, receive_stream);
ConfigureSync(config.sync_group);
auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
@ -885,8 +875,6 @@ void Call::DestroyAudioReceiveStream(
// for this sync_group.
ConfigureSync(audio_receive_stream->sync_group());
UnregisterReceiveStream(ssrc);
UpdateAggregateNetworkState();
// TODO(bugs.webrtc.org/11993): Consider if deleting `audio_receive_stream`
// on the network thread would be better or if we'd need to tear down the
@ -1023,15 +1011,6 @@ webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
// TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
// thread.
receive_stream->RegisterWithTransport(&video_receiver_controller_);
if (receive_stream->rtx_ssrc()) {
// We record identical config for the rtx stream as for the main
// stream. Since the transport_send_cc negotiation is per payload
// type, we may get an incorrect value for the rtx stream, but
// that is unlikely to matter in practice.
RegisterReceiveStream(receive_stream->rtx_ssrc(), receive_stream);
}
RegisterReceiveStream(receive_stream->remote_ssrc(), receive_stream);
video_receive_streams_.insert(receive_stream);
ConfigureSync(receive_stream->sync_group());
@ -1050,14 +1029,6 @@ void Call::DestroyVideoReceiveStream(
static_cast<VideoReceiveStream2*>(receive_stream);
// TODO(bugs.webrtc.org/11993): Unregister on the network thread.
receive_stream_impl->UnregisterFromTransport();
// Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
// separate SSRC there can be either one or two.
UnregisterReceiveStream(receive_stream_impl->remote_ssrc());
if (receive_stream_impl->rtx_ssrc()) {
UnregisterReceiveStream(receive_stream_impl->rtx_ssrc());
}
video_receive_streams_.erase(receive_stream_impl);
ConfigureSync(receive_stream_impl->sync_group());
@ -1085,8 +1056,6 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
// TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
// thread.
receive_stream->RegisterWithTransport(&video_receiver_controller_);
RegisterReceiveStream(receive_stream->remote_ssrc(), receive_stream);
// TODO(brandtr): Store config in RtcEventLog here.
return receive_stream;
@ -1102,8 +1071,6 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
receive_stream_impl->UnregisterFromTransport();
auto ssrc = receive_stream_impl->remote_ssrc();
UnregisterReceiveStream(ssrc);
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
// destroyed.
receive_side_cc_.RemoveStream(ssrc);
@ -1467,26 +1434,6 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
receive_side_cc_.OnReceivedPacket(packet, media_type);
}
bool Call::RegisterReceiveStream(uint32_t ssrc,
ReceiveStreamInterface* stream) {
RTC_DCHECK_RUN_ON(&receive_11993_checker_);
RTC_DCHECK(stream);
auto inserted = receive_rtp_config_.emplace(ssrc, stream);
if (!inserted.second) {
RTC_DLOG(LS_WARNING) << "ssrc already registered: " << ssrc;
}
return inserted.second;
}
bool Call::UnregisterReceiveStream(uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&receive_11993_checker_);
size_t erased = receive_rtp_config_.erase(ssrc);
if (!erased) {
RTC_DLOG(LS_WARNING) << "ssrc wasn't registered: " << ssrc;
}
return erased != 0u;
}
} // namespace internal
} // namespace webrtc