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(), |