Part 2: Plumb render view ID to content::MediaObserver

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"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using testing::_;
 using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace {
 
+static const int kRenderViewId = -2;
+
 ACTION_P(QuitMessageLoop, loop_or_proxy) {
   loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
 }
 
 class AudioUtil : public AudioUtilInterface {
  public:
   AudioUtil() {}
 
   virtual int GetAudioHardwareSampleRate() OVERRIDE {
     return media::GetAudioHardwareSampleRate();
@@ skipping 221 matching lines @@
     return;
   }
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+      OnSetAudioStreamStatus(_, kRenderViewId,
+                             1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamPlaying(_, 1, true)).Times(1);
+      OnSetAudioStreamPlaying(_, kRenderViewId, 1, true)).Times(1);
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
+      OnSetAudioStreamStatus(_, kRenderViewId,
+                             1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
+      OnDeleteAudioStream(_, kRenderViewId, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       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);
@@ skipping 117 matching lines @@
   std::string file_path(
       GetTestDataPath(FILE_PATH_LITERAL("speechmusic_mono_16kHz.pcm")));
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("created"))).Times(1);
+      OnSetAudioStreamStatus( _, kRenderViewId,
+                             1, StrEq("created"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamPlaying(_, 1, true)).Times(1);
+      OnSetAudioStreamPlaying(_, kRenderViewId,
+                              1, true)).Times(1);
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("closed"))).Times(1);
+      OnSetAudioStreamStatus(_, kRenderViewId,
+                             1, StrEq("closed"))).Times(1);
   EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
+      OnDeleteAudioStream(_, kRenderViewId, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       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());
@@ skipping 41 matching lines @@
     return;
   }
 
   AudioUtil audio_util;
   SetAudioUtilCallback(&audio_util);
 
   if (!HardwareSampleRatesAreValid())
     return;
 
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("created")));
+      OnSetAudioStreamStatus(_, kRenderViewId, 1, StrEq("created")));
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamPlaying(_, 1, true));
+      OnSetAudioStreamPlaying(_, kRenderViewId, 1, true));
   EXPECT_CALL(media_observer(),
-      OnSetAudioStreamStatus(_, 1, StrEq("closed")));
+      OnSetAudioStreamStatus(_, kRenderViewId, 1, StrEq("closed")));
   EXPECT_CALL(media_observer(),
-      OnDeleteAudioStream(_, 1)).Times(AnyNumber());
+      OnDeleteAudioStream(_, kRenderViewId, 1)).Times(AnyNumber());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       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);
@@ skipping 23 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());
 }