| 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_;
|
|
|