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Remove Call::sync_stream_mapping_

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This std::map was used to look up audio streams from ssrcs when
creating/destroying and/or modifying streams. Those operations aren't
frequent enough to justify having a separate lookup map. Removing
the variable, simplifies the thread ownership work a bit.

Bug: webrtc:11993
Change-Id: I94f9f2f56c138051a8b9c5f6a6d7cae3a4e78b48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249091
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35806}
This commit is contained in:
Tommi 2022-01-26 14:06:42 +01:00 committed by WebRTC LUCI CQ
parent 5f0eb93d2a
commit d3b3a3b6bb
1 changed files with 29 additions and 43 deletions
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72
call/call.cc
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@ -346,6 +346,9 @@ class Call final : public webrtc::Call,
DeliveryStatus DeliverRtp(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) RTC_RUN_ON(worker_thread_);
AudioReceiveStream* FindAudioStreamForSyncGroup(const std::string& sync_group)
RTC_RUN_ON(worker_thread_);
void ConfigureSync(const std::string& sync_group) RTC_RUN_ON(worker_thread_);
void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
@ -390,14 +393,11 @@ class Call final : public webrtc::Call,
// Audio, Video, and FlexFEC receive streams are owned by the client that
// creates them.
// TODO(bugs.webrtc.org/11993): Move audio_receive_streams_,
// video_receive_streams_ and sync_stream_mapping_ over to the network thread.
// video_receive_streams_ over to the network thread.
std::set<AudioReceiveStream*> audio_receive_streams_
RTC_GUARDED_BY(worker_thread_);
std::set<VideoReceiveStream2*> video_receive_streams_
RTC_GUARDED_BY(worker_thread_);
std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
RTC_GUARDED_BY(worker_thread_);
// TODO(nisse): Should eventually be injected at creation,
// with a single object in the bundled case.
RtpStreamReceiverController audio_receiver_controller_
@ -1009,11 +1009,11 @@ void Call::DestroyAudioReceiveStream(
audio_receive_streams_.erase(audio_receive_stream);
const auto it = sync_stream_mapping_.find(config.sync_group);
if (it != sync_stream_mapping_.end() && it->second == audio_receive_stream) {
sync_stream_mapping_.erase(it);
ConfigureSync(config.sync_group);
}
// After calling erase(), call ConfigureSync. This will clear associated
// video streams or associate them with a different audio stream if one exists
// for this sync_group.
ConfigureSync(audio_receive_stream->config().sync_group);
UnregisterReceiveStream(ssrc);
UpdateAggregateNetworkState();
@ -1459,51 +1459,37 @@ void Call::OnAllocationLimitsChanged(BitrateAllocationLimits limits) {
}
// RTC_RUN_ON(worker_thread_)
void Call::ConfigureSync(const std::string& sync_group) {
// TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
// Set sync only if there was no previous one.
if (sync_group.empty())
return;
AudioReceiveStream* sync_audio_stream = nullptr;
// Find existing audio stream.
const auto it = sync_stream_mapping_.find(sync_group);
if (it != sync_stream_mapping_.end()) {
sync_audio_stream = it->second;
} else {
// No configured audio stream, see if we can find one.
AudioReceiveStream* Call::FindAudioStreamForSyncGroup(
const std::string& sync_group) {
RTC_DCHECK_RUN_ON(&receive_11993_checker_);
if (!sync_group.empty()) {
for (AudioReceiveStream* stream : audio_receive_streams_) {
if (stream->config().sync_group == sync_group) {
if (sync_audio_stream != nullptr) {
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio stream "
"within the same sync group. This is not "
"supported in the current implementation.";
break;
}
sync_audio_stream = stream;
}
if (stream->config().sync_group == sync_group)
return stream;
}
}
if (sync_audio_stream)
sync_stream_mapping_[sync_group] = sync_audio_stream;
return nullptr;
}
// TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
// RTC_RUN_ON(worker_thread_)
void Call::ConfigureSync(const std::string& sync_group) {
// `audio_stream` may be nullptr when clearing the audio stream for a group.
AudioReceiveStream* audio_stream = FindAudioStreamForSyncGroup(sync_group);
size_t num_synced_streams = 0;
for (VideoReceiveStream2* video_stream : video_receive_streams_) {
if (video_stream->sync_group() != sync_group)
continue;
++num_synced_streams;
if (num_synced_streams > 1) {
// TODO(pbos): Support synchronizing more than one A/V pair.
// https://code.google.com/p/webrtc/issues/detail?id=4762
RTC_LOG(LS_WARNING)
<< "Attempting to sync more than one audio/video pair "
"within the same sync group. This is not supported in "
"the current implementation.";
}
// TODO(bugs.webrtc.org/4762): Support synchronizing more than one A/V pair.
// Attempting to sync more than one audio/video pair within the same sync
// group is not supported in the current implementation.
// Only sync the first A/V pair within this sync group.
if (num_synced_streams == 1) {
// sync_audio_stream may be null and that's ok.
video_stream->SetSync(sync_audio_stream);
video_stream->SetSync(audio_stream);
} else {
video_stream->SetSync(nullptr);
}