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Remove network_enabled_crit_ in call.cc.

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After #10321 (5a289393928c18af580c6339ba77600fb67006e2) I don't see that
we still need this lock.

R=pbos@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1409193003 .

Cr-Commit-Position: refs/heads/master@{#10410}
This commit is contained in:
mflodman 2015-10-26 16:34:46 +01:00
parent 09b38f3ca0
commit 717432f130
1 changed files with 6 additions and 20 deletions
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26
webrtc/call/call.cc
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@ -101,11 +101,7 @@ class Call : public webrtc::Call, public PacketReceiver {
Call::Config config_;
rtc::ThreadChecker configuration_thread_checker_;
// Needs to be held while write-locking |receive_crit_| or |send_crit_|. This
// ensures that we have a consistent network state signalled to all senders
// and receivers.
rtc::CriticalSection network_enabled_crit_;
bool network_enabled_ GUARDED_BY(network_enabled_crit_);
bool network_enabled_;
rtc::scoped_ptr<RWLockWrapper> receive_crit_;
// Audio and Video receive streams are owned by the client that creates them.
@ -204,15 +200,13 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
AudioSendStream* send_stream = new AudioSendStream(config);
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
{
rtc::CritScope lock(&network_enabled_crit_);
WriteLockScoped write_lock(*send_crit_);
RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
audio_send_ssrcs_.end());
audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
}
return send_stream;
}
@ -288,9 +282,9 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
congestion_controller_.get(), config, encoder_config,
suspended_video_send_ssrcs_);
// This needs to be taken before send_crit_ as both locks need to be held
// while changing network state.
rtc::CritScope lock(&network_enabled_crit_);
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
WriteLockScoped write_lock(*send_crit_);
for (uint32_t ssrc : config.rtp.ssrcs) {
RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
@ -301,8 +295,6 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
if (event_log_)
event_log_->LogVideoSendStreamConfig(config);
if (!network_enabled_)
send_stream->SignalNetworkState(kNetworkDown);
return send_stream;
}
@ -348,9 +340,6 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
num_cpu_cores_, congestion_controller_.get(), config,
config_.voice_engine, module_process_thread_.get(), call_stats_.get());
// This needs to be taken before receive_crit_ as both locks need to be held
// while changing network state.
rtc::CritScope lock(&network_enabled_crit_);
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
video_receive_ssrcs_.end());
@ -454,9 +443,6 @@ void Call::SetBitrateConfig(
void Call::SignalNetworkState(NetworkState state) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
// Take crit for entire function, it needs to be held while updating streams
// to guarantee a consistent state across streams.
rtc::CritScope lock(&network_enabled_crit_);
network_enabled_ = state == kNetworkUp;
congestion_controller_->SignalNetworkState(state);
{