Only turn on the audio processing by default for MEDIA_DEVICE_AUDIO_CAPTURE

content/renderer/media/webrtc_local_audio_track_unittest.cc

 // Copyright 2013 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/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
@@ skipping 146 matching lines @@
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackTest : public ::testing::Test {
  protected:
   virtual void SetUp() OVERRIDE {
     params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                   media::CHANNEL_LAYOUT_STEREO, 2, 0, 48000, 16, 480);
     blink::WebMediaConstraints constraints;
-    capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo(),
+    StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
+                            std::string(), std::string());
+    capturer_ = WebRtcAudioCapturer::CreateCapturer(-1, device,
                                                     constraints, NULL);
     capturer_source_ = new MockCapturerSource(capturer_.get());
     EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
         .WillOnce(Return());
     capturer_->SetCapturerSourceForTesting(capturer_source_, params_);
   }
 
   media::AudioParameters params_;
   scoped_refptr<MockCapturerSource> capturer_source_;
   scoped_refptr<WebRtcAudioCapturer> capturer_;
@@ skipping 286 matching lines @@
   EXPECT_CALL(
       *sink_1.get(),
       CaptureData(
           kNumberOfNetworkChannelsForTrack1, 48000, 2, _, 0, 0, true, false))
       .Times(AnyNumber()).WillRepeatedly(Return());
   EXPECT_CALL(*sink_1.get(), OnSetFormat(_)).Times(AnyNumber());
   track_1->AddSink(sink_1.get());
 
   // Create a new capturer with new source with different audio format.
   blink::WebMediaConstraints constraints;
+  StreamDeviceInfo device(MEDIA_DEVICE_AUDIO_CAPTURE,
+                          std::string(), std::string());
   scoped_refptr<WebRtcAudioCapturer> new_capturer(
-      WebRtcAudioCapturer::CreateCapturer(-1, StreamDeviceInfo(),
-                                          constraints, NULL));
+      WebRtcAudioCapturer::CreateCapturer(-1, device, constraints, NULL));
   scoped_refptr<MockCapturerSource> new_source(
       new MockCapturerSource(new_capturer.get()));
   EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
   media::AudioParameters new_param(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
       media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
   new_capturer->SetCapturerSourceForTesting(new_source, new_param);
 
   // Setup the second audio track, connect it to the new capturer and start it.
   EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
@@ skipping 88 matching lines @@
       .Times(AtLeast(1)).WillRepeatedly(SignalEvent(&event));
   track->AddSink(sink.get());
   EXPECT_TRUE(event.TimedWait(TestTimeouts::tiny_timeout()));
 
   // Stopping the new source will stop the second track.
   EXPECT_CALL(*source, OnStop()).Times(1);
   capturer->Stop();
 }
 
 }  // namespace content