Index: webrtc/media/engine/fakewebrtccall.h |
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h |
index 5719070eeaee78049889c62518b343163838b451..dfe8fbb2bad06c0ead9235ac65a8900300ff1872 100644 |
--- a/webrtc/media/engine/fakewebrtccall.h |
+++ b/webrtc/media/engine/fakewebrtccall.h |
@@ -59,6 +59,13 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream { |
void SetMuted(bool muted) override; |
webrtc::AudioSendStream::Stats GetStats() const override; |
+ bool EnableAudioNetworkAdaptor(const std::string& config_string) override; |
+ |
+ void DisableAudioNetworkAdaptor() override; |
+ |
+ void SetReceiverFrameLengthRange(int min_frame_length_ms, |
+ int max_frame_length_ms) override; |
+ |
TelephoneEvent latest_telephone_event_; |
webrtc::AudioSendStream::Config config_; |
webrtc::AudioSendStream::Stats stats_; |