Adds Analog Gain Control (AGC) to the WebRTC client.

content/renderer/media/webrtc_audio_device_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager.h"
@@ skipping 419 matching lines @@
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
   int err = base->Init(audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
   EXPECT_EQ(0, base->StartPlayout(ch));
 
   ScopedWebRTCPtr<webrtc::VoEFile> file(engine.get());
+  ASSERT_TRUE(file.valid());
   int duration = 0;
   EXPECT_EQ(0, file->GetFileDuration(file_path.c_str(), duration,
                                      webrtc::kFileFormatPcm16kHzFile));
   EXPECT_NE(0, duration);
 
   EXPECT_EQ(0, file->StartPlayingFileLocally(ch, file_path.c_str(), false,
                                              webrtc::kFileFormatPcm16kHzFile));
 
   // Play 2 seconds worth of audio and then quit.
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
                                 base::TimeDelta::FromSeconds(2));
   message_loop_.Run();
 
 
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 // Uses WebRtcAudioDeviceImpl to play out recorded audio in loopback.
 // An external transport implementation is utilized to feed back RTP packets
 // which are recorded, encoded, packetized into RTP packets and finally
 // "transmitted". The RTP packets are then fed back into the VoiceEngine
 // where they are decoded and played out on the default audio output device.
 // Disabled when running headless since the bots don't have the required config.
 // TODO(henrika): improve quality by using a wideband codec, enabling noise-
-// suppressions and perhaps also the digital AGC.
-TEST_F(WebRTCAudioDeviceTest, FullDuplexAudio) {
+// suppressions etc.
+TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {
   if (IsRunningHeadless())
     return;
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   EXPECT_CALL(media_observer(),
-    OnSetAudioStreamStatus(_, 1, StrEq("created")));
+      OnSetAudioStreamStatus(_, 1, StrEq("created")));
   EXPECT_CALL(media_observer(),
-    OnSetAudioStreamPlaying(_, 1, true));
+      OnSetAudioStreamPlaying(_, 1, true));
   EXPECT_CALL(media_observer(),
-    OnSetAudioStreamStatus(_, 1, StrEq("closed")));
+      OnSetAudioStreamStatus(_, 1, StrEq("closed")));
   EXPECT_CALL(media_observer(),
-    OnDeleteAudioStream(_, 1)).Times(AnyNumber());
+      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> audio_device(
       new WebRtcAudioDeviceImpl());
   audio_device->SetSessionId(1);
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
   int err = base->Init(audio_device);
   ASSERT_EQ(0, err);
 
+  ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
+  ASSERT_TRUE(audio_processing.valid());
+  bool enabled = false;
+  webrtc::AgcModes agc_mode =  webrtc::kAgcDefault;
+  EXPECT_EQ(0, audio_processing->GetAgcStatus(enabled, agc_mode));
+  EXPECT_TRUE(enabled);
+  EXPECT_EQ(agc_mode, webrtc::kAgcAdaptiveAnalog);
+
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
 
   ScopedWebRTCPtr<webrtc::VoENetwork> network(engine.get());
+  ASSERT_TRUE(network.valid());
   scoped_ptr<WebRTCTransportImpl> transport(
       new WebRTCTransportImpl(network.get()));
   EXPECT_EQ(0, network->RegisterExternalTransport(ch, *transport.get()));
   EXPECT_EQ(0, base->StartPlayout(ch));
   EXPECT_EQ(0, base->StartSend(ch));
 
   LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
                                 TestTimeouts::action_timeout());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }