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| author | Markus Handell <handellm@webrtc.org> | 2021-06-22 10:46:48 +0200 |
|---|---|---|
| committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | 2021-06-22 14:51:04 +0000 |
| commit | eb61b7f620c1f84cb9d34c7752cea65fe5d1d721 (patch) | |
| tree | 4240c7418e1c2a9fde352664bd6927b20fac07f9 /audio/voip/audio_channel.cc | |
| parent | 6e65f6a4288674ff33614f557c33c9dc871fa2e2 (diff) | |
| download | webrtc-eb61b7f620c1f84cb9d34c7752cea65fe5d1d721.tar.gz | |
ModuleRtcRtcpImpl2: remove Module inheritance.upstream-master
This change achieves an Idle Wakeup savings of 200 Hz.
ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.
Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
Diffstat (limited to 'audio/voip/audio_channel.cc')
| -rw-r--r-- | audio/voip/audio_channel.cc | 13 |
1 files changed, 3 insertions, 10 deletions
diff --git a/audio/voip/audio_channel.cc b/audio/voip/audio_channel.cc index d11e6d79f9..b4a50eec12 100644 --- a/audio/voip/audio_channel.cc +++ b/audio/voip/audio_channel.cc @@ -32,12 +32,10 @@ AudioChannel::AudioChannel( Transport* transport, uint32_t local_ssrc, TaskQueueFactory* task_queue_factory, - ProcessThread* process_thread, AudioMixer* audio_mixer, rtc::scoped_refptr<AudioDecoderFactory> decoder_factory) - : audio_mixer_(audio_mixer), process_thread_(process_thread) { + : audio_mixer_(audio_mixer) { RTC_DCHECK(task_queue_factory); - RTC_DCHECK(process_thread); RTC_DCHECK(audio_mixer); Clock* clock = Clock::GetRealTimeClock(); @@ -56,9 +54,6 @@ AudioChannel::AudioChannel( rtp_rtcp_->SetSendingMediaStatus(false); rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound); - // ProcessThread periodically services RTP stack for RTCP. - process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE); - ingress_ = std::make_unique<AudioIngress>(rtp_rtcp_.get(), clock, receive_statistics_.get(), std::move(decoder_factory)); @@ -80,12 +75,10 @@ AudioChannel::~AudioChannel() { audio_mixer_->RemoveSource(ingress_.get()); - // AudioEgress could hold current global TaskQueueBase that we need to clear - // before ProcessThread::DeRegisterModule. + // TODO(bugs.webrtc.org/11581): unclear if we still need to clear |egress_| + // here. egress_.reset(); ingress_.reset(); - - process_thread_->DeRegisterModule(rtp_rtcp_.get()); } bool AudioChannel::StartSend() { |