Implicit audio output device selection for getUserMedia.

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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 529 matching lines @@
   ASSERT_TRUE(external_media.valid());
   base::WaitableEvent event(false, false);
   scoped_ptr<WebRTCMediaProcessImpl> media_process(
       new WebRTCMediaProcessImpl(&event));
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
   EXPECT_EQ(0, external_media->RegisterExternalMediaProcessing(
       ch, webrtc::kPlaybackPerChannel, *media_process.get()));
 
   EXPECT_EQ(0, base->StartPlayout(ch));
-  scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+  scoped_refptr<WebRtcAudioRenderer> renderer(
+      CreateDefaultWebRtcAudioRenderer(kRenderViewId));
+
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
   renderer->Play();
 
   EXPECT_TRUE(event.TimedWait(TestTimeouts::action_timeout()));
   WaitForIOThreadCompletion();
 
   EXPECT_TRUE(webrtc_audio_device->Playing());
   EXPECT_FALSE(webrtc_audio_device->Recording());
   EXPECT_EQ(ch, media_process->channel_id());
   EXPECT_EQ(webrtc::kPlaybackPerChannel, media_process->type());
@@ skipping 153 matching lines @@
   ScopedWebRTCPtr<webrtc::VoEBase> base(engine.get());
   ASSERT_TRUE(base.valid());
 
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   int err = base->Init(webrtc_audio_device.get());
   ASSERT_EQ(0, err);
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);
   EXPECT_EQ(0, base->StartPlayout(ch));
-  scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+  scoped_refptr<WebRtcAudioRenderer> renderer(
+      CreateDefaultWebRtcAudioRenderer(kRenderViewId));
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
   renderer->Play();
 
   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);
 
@@ skipping 88 matching lines @@
   static_cast<webrtc::AudioTrackInterface*>(local_audio_track.get())->
       GetRenderer()->AddChannel(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));
-  scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+  scoped_refptr<WebRtcAudioRenderer> renderer(
+      CreateDefaultWebRtcAudioRenderer(kRenderViewId));
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
   renderer->Play();
 
   LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
                                 base::MessageLoop::QuitClosure(),
                                 base::TimeDelta::FromSeconds(2));
   message_loop_.Run();
 
   local_audio_track->Stop();
@@ skipping 89 matching lines @@
   EXPECT_CALL(media_observer(),
               OnSetAudioStreamStatus(_, 1, _)).Times(AnyNumber());
   EXPECT_CALL(media_observer(),
               OnSetAudioStreamPlaying(_, 1, true));
   EXPECT_CALL(media_observer(),
               OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
   base::WaitableEvent event(false, false);
   scoped_ptr<MockWebRtcAudioRendererSource> renderer_source(
       new MockWebRtcAudioRendererSource(&event));
-  scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+
+  scoped_refptr<WebRtcAudioRenderer> renderer(
+      CreateDefaultWebRtcAudioRenderer(kRenderViewId));
   renderer->Initialize(renderer_source.get());
 
   // Start the timer and playout.
   base::Time start_time = base::Time::Now();
   renderer->Play();
   EXPECT_TRUE(event.TimedWait(TestTimeouts::action_timeout()));
   int delay = (base::Time::Now() - start_time).InMilliseconds();
   PrintPerfResultMs("webrtc_playout_setup_c", "t", delay);
 
   renderer->Stop();
@@ skipping 25 matching lines @@
 #endif
 
 TEST_F(MAYBE_WebRTCAudioDeviceTest,
        MAYBE_WebRtcLoopbackTimeWithSignalProcessing) {
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content