Part 1: Plumb render view ID to render host

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 215 matching lines @@
         invalid_rates[i]));
   }
 }
 
 // Basic test that instantiates and initializes an instance of
 // WebRtcAudioDeviceImpl.
 TEST_F(WebRTCAudioDeviceTest, Construct) {
   AudioUtilNoHardware audio_util(48000, 48000, CHANNEL_LAYOUT_MONO);
   SetAudioUtilCallback(&audio_util);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
-      new WebRtcAudioDeviceImpl());
+      new WebRtcAudioDeviceImpl(0));
 
   webrtc_audio_device->SetSessionId(1);
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   int err = base->Init(webrtc_audio_device);
   EXPECT_EQ(0, err);
   EXPECT_EQ(0, base->Terminate());
@@ skipping 20 matching lines @@
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
-      new WebRtcAudioDeviceImpl());
+      new WebRtcAudioDeviceImpl(0));
   webrtc_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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
@@ skipping 47 matching lines @@
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   // TODO(tommi): extend MediaObserver and MockMediaObserver with support
   // for new interfaces, like OnSetAudioStreamRecording(). When done, add
   // EXPECT_CALL() macros here.
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
-      new WebRtcAudioDeviceImpl());
+      new WebRtcAudioDeviceImpl(0));
   webrtc_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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   int ch = base->CreateChannel();
@@ skipping 55 matching lines @@
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
-      new WebRtcAudioDeviceImpl());
+      new WebRtcAudioDeviceImpl(0));
   webrtc_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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
@@ skipping 47 matching lines @@
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("created")));
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamPlaying(_, 1, true));
   EXPECT_CALL(media_observer(),
       OnSetAudioStreamStatus(_, 1, StrEq("closed")));
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
-      new WebRtcAudioDeviceImpl());
+      new WebRtcAudioDeviceImpl(0));
   webrtc_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(webrtc_audio_device);
   ASSERT_EQ(0, err);
 
   ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
@@ skipping 20 matching lines @@
                                 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());
 }