| 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_;
|
|
|