MediaStream audio sourcing: Bypass audio processing for non-WebRTC cases.

content/renderer/media/webrtc_local_audio_source_provider_unittest.cc

Index: content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
diff --git a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
index 28acddd2d4fe3fc60b928df5aff93926c60e4a32..8f87e21497a607b55fe158e8427357b9f14cd389 100644
--- a/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
+++ b/content/renderer/media/webrtc_local_audio_source_provider_unittest.cc
@@ -5,16 +5,13 @@
 #include <stddef.h>
 
 #include "base/logging.h"
-#include "base/strings/utf_string_conversions.h"
-#include "content/renderer/media/mock_media_constraint_factory.h"
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
-#include "content/renderer/media/webrtc_audio_capturer.h"
+#include "content/renderer/media/media_stream_audio_track.h"
 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
-#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "media/audio/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
+#include "third_party/WebKit/public/platform/WebString.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 
 namespace content {
@@ -29,19 +26,12 @@ class WebRtcLocalAudioSourceProviderTest : public testing::Test {
         media::CHANNEL_LAYOUT_STEREO, 44100, 16,
         WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize);
     sink_bus_ = media::AudioBus::Create(sink_params_);
-    MockMediaConstraintFactory constraint_factory;
-    scoped_refptr<WebRtcAudioCapturer> capturer(
-        WebRtcAudioCapturer::CreateCapturer(
-            -1, StreamDeviceInfo(),
-            constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
-    scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
-        WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
-    scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
+    scoped_ptr<MediaStreamAudioTrack> native_track(
+        new MediaStreamAudioTrack(true /* is_local_track */));
     blink::WebMediaStreamSource audio_source;
-    audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
+    audio_source.initialize(blink::WebString::fromUTF8("dummy_source_id"),
                             blink::WebMediaStreamSource::TypeAudio,
-                            base::UTF8ToUTF16("dummy_source_name"),
+                            blink::WebString::fromUTF8("dummy_source_name"),
                             false /* remote */, true /* readonly */);
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
@@ -126,17 +116,13 @@ TEST_F(WebRtcLocalAudioSourceProviderTest,
   source_provider_.reset();
 
   // Stop the audio track.
-  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
-      MediaStreamTrack::GetTrack(blink_track_));
-  native_track->Stop();
+  MediaStreamAudioTrack::Get(blink_track_)->Stop();
 }
 
 TEST_F(WebRtcLocalAudioSourceProviderTest,
        StopTrackBeforeDeletingSourceProvider) {
   // Stop the audio track.
-  WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
-      MediaStreamTrack::GetTrack(blink_track_));
-  native_track->Stop();
+  MediaStreamAudioTrack::Get(blink_track_)->Stop();
 
   // Delete the source provider.
   source_provider_.reset();