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Delete audio methods SignalNetworkState

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These methods were defined, and called, but not doing anything.

Bug: None
Change-Id: I9955843a6bd86e4a583b0213ddb6b3b42e2ab815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150792
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29020}
This commit is contained in:
Niels Möller 2019-08-29 13:56:00 +02:00 committed by Commit Bot
parent 4894fdeba2
commit f13df86414
5 changed files with 0 additions and 21 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

4
audio/audio_receive_stream.cc
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@ -325,10 +325,6 @@ void AudioReceiveStream::AssociateSendStream(AudioSendStream* send_stream) {
associated_send_stream_ = send_stream;
}
void AudioReceiveStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK_RUN_ON(&worker_thread_checker_);
}
void AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network

1
audio/audio_receive_stream.h
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@ -91,7 +91,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
void SetMinimumPlayoutDelay(int delay_ms) override;
void AssociateSendStream(AudioSendStream* send_stream);
void SignalNetworkState(NetworkState state);
void DeliverRtcp(const uint8_t* packet, size_t length);
const webrtc::AudioReceiveStream::Config& config() const;
const AudioSendStream* GetAssociatedSendStreamForTesting() const;

4
audio/audio_send_stream.cc
View File

@ -472,10 +472,6 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats(
return stats;
}
void AudioSendStream::SignalNetworkState(NetworkState state) {
RTC_DCHECK(worker_thread_checker_.IsCurrent());
}
void AudioSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
// TODO(solenberg): Tests call this function on a network thread, libjingle
// calls on the worker thread. We should move towards always using a network

1
audio/audio_send_stream.h
View File

@ -79,7 +79,6 @@ class AudioSendStream final : public webrtc::AudioSendStream,
webrtc::AudioSendStream::Stats GetStats(
bool has_remote_tracks) const override;
void SignalNetworkState(NetworkState state);
void DeliverRtcp(const uint8_t* packet, size_t length);
// Implements BitrateAllocatorObserver.

11
call/call.cc
View File

@ -648,7 +648,6 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
}
}
}
send_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return send_stream;
}
@ -706,7 +705,6 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
receive_stream->AssociateSendStream(it->second);
}
}
receive_stream->SignalNetworkState(audio_network_state_);
UpdateAggregateNetworkState();
return receive_stream;
}
@ -1010,17 +1008,8 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
}
UpdateAggregateNetworkState();
{
ReadLockScoped read_lock(*send_crit_);
for (auto& kv : audio_send_ssrcs_) {
kv.second->SignalNetworkState(audio_network_state_);
}
}
{
ReadLockScoped read_lock(*receive_crit_);
for (AudioReceiveStream* audio_receive_stream : audio_receive_streams_) {
audio_receive_stream->SignalNetworkState(audio_network_state_);
}
for (VideoReceiveStream* video_receive_stream : video_receive_streams_) {
video_receive_stream->SignalNetworkState(video_network_state_);
}