commit 217fb66e16627be23b8769edded8a23f6406ddab
author solenberg <solenberg@webrtc.org> Fri Jun 17 08:30:54 2016 -0700
committer Commit bot <commit-bot@chromium.org> Fri Jun 17 15:30:58 2016 +0000
tree 0a4f66c80f9e8ce0ea6fa6b62294c4e72050cda3
parent 387000114d4436fe2128715582e8a2f755cc004b

Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().

Removes the need to use VoEVolume::SetChannelOutputVolumeScaling().

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2062193002
Cr-Commit-Position: refs/heads/master@{#13194}

webrtc/audio/audio_receive_stream.cc

diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 15e4f61..c933f78 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -210,6 +210,11 @@
   channel_proxy_->SetSink(std::move(sink));
 }
 
+void AudioReceiveStream::SetGain(float gain) {
+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
+  channel_proxy_->SetChannelOutputVolumeScaling(gain);
+}
+
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   return config_;

webrtc/audio/audio_receive_stream.h

diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index d99956c..284f98e 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -41,6 +41,7 @@
   void Stop() override;
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
+  void SetGain(float gain) override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);

webrtc/audio/audio_receive_stream_unittest.cc

diff --git a/webrtc/audio/audio_receive_stream_unittest.cc b/webrtc/audio/audio_receive_stream_unittest.cc
index 2e789f9..8e26dd9 100644
--- a/webrtc/audio/audio_receive_stream_unittest.cc
+++ b/webrtc/audio/audio_receive_stream_unittest.cc
@@ -30,6 +30,7 @@
 namespace {
 
 using testing::_;
+using testing::FloatEq;
 using testing::Return;
 using testing::ReturnRef;
 
@@ -92,12 +93,12 @@
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
                   .Times(1);
-          EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
-                                           kTransportSequenceNumberId))
-              .Times(1);
           EXPECT_CALL(*channel_proxy_,
-                      RegisterReceiverCongestionControlObjects(&packet_router_))
-              .Times(1);
+              EnableReceiveTransportSequenceNumber(kTransportSequenceNumberId))
+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_,
+              RegisterReceiverCongestionControlObjects(&packet_router_))
+                  .Times(1);
           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
@@ -303,7 +304,6 @@
   EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
 }
 
-
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
@@ -345,5 +345,14 @@
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
+
+TEST(AudioReceiveStreamTest, SetGain) {
+  ConfigHelper helper;
+  internal::AudioReceiveStream recv_stream(
+      helper.congestion_controller(), helper.config(), helper.audio_state());
+  EXPECT_CALL(*helper.channel_proxy(),
+      SetChannelOutputVolumeScaling(FloatEq(0.765f)));
+  recv_stream.SetGain(0.765f);
+}
 }  // namespace test
 }  // namespace webrtc