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Unified Diff: webrtc/audio/audio_send_stream.cc

Issue 1924793002: Remove webrtc/stream.h and unutilized inheritance. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: re-rebase Created 4 years, 8 months ago
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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_;
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