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

webrtc/audio/audio_receive_stream_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 12 matching lines @@
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
 namespace test {
 namespace {
 
 using testing::_;
+using testing::FloatEq;
 using testing::Return;
 using testing::ReturnRef;
 
 AudioDecodingCallStats MakeAudioDecodeStatsForTest() {
   AudioDecodingCallStats audio_decode_stats;
   audio_decode_stats.calls_to_silence_generator = 234;
   audio_decode_stats.calls_to_neteq = 567;
   audio_decode_stats.decoded_normal = 890;
   audio_decode_stats.decoded_plc = 123;
   audio_decode_stats.decoded_cng = 456;
@@ skipping 41 matching lines @@
         .WillOnce(Invoke([this](int channel_id) {
           EXPECT_FALSE(channel_proxy_);
           channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
           EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
           EXPECT_CALL(*channel_proxy_,
               SetReceiveAbsoluteSenderTimeStatus(true, kAbsSendTimeId))
                   .Times(1);
           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);

[kwiberg-webrtc, 2016/06/16 09:26:53]

Ummm... am I missing something, or is this hunk just a reformatting with no
other changes?

[the sun, 2016/06/16 14:34:06]

Yes, sorry about that.

           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
           EXPECT_CALL(*channel_proxy_, RegisterExternalTransport(nullptr))
               .Times(1);
           EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
               .Times(1);
           EXPECT_CALL(*channel_proxy_, GetAudioDecoderFactory())
               .WillOnce(ReturnRef(decoder_factory_));
@@ skipping 184 matching lines @@
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state());
 
   std::vector<uint8_t> rtcp_packet = CreateRtcpSenderReport();
   EXPECT_CALL(*helper.channel_proxy(),
               ReceivedRTCPPacket(&rtcp_packet[0], rtcp_packet.size()))
       .WillOnce(Return(true));
   EXPECT_TRUE(recv_stream.DeliverRtcp(&rtcp_packet[0], rtcp_packet.size()));
 }
 
-
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
   internal::AudioReceiveStream recv_stream(
       helper.congestion_controller(), helper.config(), helper.audio_state());
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
@@ skipping 21 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
             stats.decoding_calls_to_silence_generator);
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   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