| Index: webrtc/audio/audio_send_stream_unittest.cc
|
| diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
|
| index 44392ea1c0edb6238e83564cfd51fa84329cd11b..b9535543e23661891b4594350b1ccae3f9b7bd89 100644
|
| --- a/webrtc/audio/audio_send_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_send_stream_unittest.cc
|
| @@ -110,6 +110,7 @@ struct ConfigHelper {
|
|
|
| AudioSendStream::Config& config() { return stream_config_; }
|
| rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
|
| + MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
|
| CongestionController* congestion_controller() {
|
| return &congestion_controller_;
|
| }
|
| @@ -200,6 +201,14 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
|
| kTelephoneEventCode, kTelephoneEventDuration));
|
| }
|
|
|
| +TEST(AudioSendStreamTest, SetMuted) {
|
| + ConfigHelper helper;
|
| + internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
| + helper.congestion_controller());
|
| + EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
|
| + send_stream.SetMuted(true);
|
| +}
|
| +
|
| TEST(AudioSendStreamTest, GetStats) {
|
| ConfigHelper helper;
|
| internal::AudioSendStream send_stream(helper.config(), helper.audio_state(),
|
|
|