Associate audio streams with their source/destination RenderView.

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/renderer/media/webrtc_audio_renderer.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/include/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/include/voe_base.h"
 #include "third_party/webrtc/voice_engine/include/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/include/voe_file.h"
 #include "third_party/webrtc/voice_engine/include/voe_network.h"
 
 using testing::_;
 using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace content {
 
 namespace {
 
+const int kRenderViewId = 1;
+
 class AudioUtil : public AudioUtilInterface {
  public:
   AudioUtil() {}
 
   virtual int GetAudioHardwareSampleRate() OVERRIDE {
     return media::GetAudioHardwareSampleRate();
   }
   virtual int GetAudioInputHardwareSampleRate(
       const std::string& device_id) OVERRIDE {
     return media::GetAudioInputHardwareSampleRate(device_id);
@@ skipping 218 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<WebRtcAudioRenderer> renderer = new WebRtcAudioRenderer();
+  scoped_refptr<WebRtcAudioRenderer> renderer =
+      new WebRtcAudioRenderer(kRenderViewId);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
@@ skipping 128 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<WebRtcAudioRenderer> renderer = new WebRtcAudioRenderer();
-
+  scoped_refptr<WebRtcAudioRenderer> renderer =
+      new WebRtcAudioRenderer(kRenderViewId);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
@@ skipping 50 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<WebRtcAudioRenderer> renderer = new WebRtcAudioRenderer();
-
+  scoped_refptr<WebRtcAudioRenderer> renderer =
+      new WebRtcAudioRenderer(kRenderViewId);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   webrtc_audio_device->SetSessionId(1);
   EXPECT_TRUE(webrtc_audio_device->SetRenderer(renderer));
 
   WebRTCAutoDelete<webrtc::VoiceEngine> engine(webrtc::VoiceEngine::Create());
   ASSERT_TRUE(engine.valid());
 
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
@@ skipping 28 matching lines @@
 
   renderer->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
 
 }  // namespace content