Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index c5dfd77836e3005c215600d2814b7a294f957368..c0a709e3c6296765df1cf6c0ab8ec6146010a636 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -116,18 +116,6 @@ void AudioSendStream::Stop() { |
} |
} |
-void AudioSendStream::SignalNetworkState(NetworkState state) { |
- RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
-} |
- |
-bool 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 |
- // thread. Then this check can be enabled. |
- // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
- return channel_proxy_->ReceivedRTCPPacket(packet, length); |
-} |
- |
bool AudioSendStream::SendTelephoneEvent(int payload_type, int event, |
int duration_ms) { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
@@ -218,6 +206,18 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
return stats; |
} |
+void AudioSendStream::SignalNetworkState(NetworkState state) { |
+ RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
+} |
+ |
+bool 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 |
+ // thread. Then this check can be enabled. |
+ // RTC_DCHECK(!thread_checker_.CalledOnValidThread()); |
+ return channel_proxy_->ReceivedRTCPPacket(packet, length); |
+} |
+ |
const webrtc::AudioSendStream::Config& AudioSendStream::config() const { |
RTC_DCHECK(thread_checker_.CalledOnValidThread()); |
return config_; |