commit 9421853e178151e5bf6f098a739da7a3db39528a
author solenberg <solenberg@webrtc.org> Thu Jun 16 10:53:22 2016 -0700
committer Commit bot <commit-bot@chromium.org> Thu Jun 16 17:53:28 2016 +0000
tree b7ae89a29d5bba99a8e3349f0b7a164c6b9e8294
parent e565a04de3d28e6530a2599e6b90bb9fb5748185

Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().

Removes the need to use VoEVolume::SetInputMute()/GetInputMute().

BUG=webrtc:4690
NOTRY=true

Review-Url: https://codereview.webrtc.org/2066973002
Cr-Commit-Position: refs/heads/master@{#13172}

webrtc/audio/audio_send_stream.cc

diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 9aa2fc0..d08bfea 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -127,6 +127,11 @@
          channel_proxy_->SendTelephoneEventOutband(event, duration_ms);
 }
 
+void AudioSendStream::SetMuted(bool muted) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetInputMute(muted);
+}
+
 webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   webrtc::AudioSendStream::Stats stats;

webrtc/audio/audio_send_stream.h

diff --git a/webrtc/audio/audio_send_stream.h b/webrtc/audio/audio_send_stream.h
index 61dd7f2..264f9b3 100644
--- a/webrtc/audio/audio_send_stream.h
+++ b/webrtc/audio/audio_send_stream.h
@@ -39,6 +39,7 @@
   void Stop() override;
   bool SendTelephoneEvent(int payload_type, int event,
                           int duration_ms) override;
+  void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);

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 44392ea..b953554 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -110,6 +110,7 @@
 
   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 @@
       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(),