Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one.

content/renderer/media/webrtc/webrtc_media_stream_adapter_unittest.cc

 // Copyright 2014 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 "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 
 #include <stddef.h>
 
 #include <memory>
 
 #include "base/message_loop/message_loop.h"
 #include "content/child/child_process.h"
 #include "content/renderer/media/media_stream.h"
+#include "content/renderer/media/media_stream_audio_source.h"
 #include "content/renderer/media/media_stream_video_source.h"
 #include "content/renderer/media/media_stream_video_track.h"
-#include "content/renderer/media/mock_audio_device_factory.h"
-#include "content/renderer/media/mock_constraint_factory.h"
 #include "content/renderer/media/mock_media_stream_video_source.h"
 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
-#include "content/renderer/media/webrtc/processed_local_audio_source.h"
-#include "testing/gmock/include/gmock/gmock.h"
+#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
+#include "content/renderer/media/webrtc_local_audio_track.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/platform/WebVector.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 
-using ::testing::_;
-
 namespace content {
 
 class WebRtcMediaStreamAdapterTest : public ::testing::Test {
  public:
   void SetUp() override {
     child_process_.reset(new ChildProcess());
     dependency_factory_.reset(new MockPeerConnectionDependencyFactory());
   }
 
   void TearDown() override {
     adapter_.reset();
     blink::WebHeap::collectAllGarbageForTesting();
   }
 
   blink::WebMediaStream CreateBlinkMediaStream(bool audio, bool video) {
     blink::WebVector<blink::WebMediaStreamTrack> audio_track_vector(
         audio ? static_cast<size_t>(1) : 0);
     if (audio) {
       blink::WebMediaStreamSource audio_source;
       audio_source.initialize("audio",
                               blink::WebMediaStreamSource::TypeAudio,
                               "audio",
                               false /* remote */);
-      ProcessedLocalAudioSource* const source =
-          new ProcessedLocalAudioSource(
-              -1 /* consumer_render_frame_id is N/A for non-browser tests */,
-              StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "Mock audio device",
-                               "mock_audio_device_id",
-                               media::AudioParameters::kAudioCDSampleRate,
-                               media::CHANNEL_LAYOUT_STEREO,
-                               media::AudioParameters::kAudioCDSampleRate / 50),
-              dependency_factory_.get());
-      source->SetAllowInvalidRenderFrameIdForTesting(true);
-      source->SetSourceConstraints(
-          MockConstraintFactory().CreateWebMediaConstraints());
-      audio_source.setExtraData(source);  // Takes ownership.
+      audio_source.setExtraData(new MediaStreamAudioSource());
+
       audio_track_vector[0].initialize(audio_source);
-      EXPECT_CALL(*mock_audio_device_factory_.mock_capturer_source(),
-                  Initialize(_, _, -1));
-      EXPECT_CALL(*mock_audio_device_factory_.mock_capturer_source(),
-                  SetAutomaticGainControl(true));
-      EXPECT_CALL(*mock_audio_device_factory_.mock_capturer_source(), Start());
-      EXPECT_CALL(*mock_audio_device_factory_.mock_capturer_source(), Stop());
-      CHECK(source->ConnectToTrack(audio_track_vector[0]));
+      scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
+          WebRtcLocalAudioTrackAdapter::Create(
+              audio_track_vector[0].id().utf8(), nullptr));
+      std::unique_ptr<WebRtcLocalAudioTrack> native_track(
+          new WebRtcLocalAudioTrack(adapter.get()));
+      audio_track_vector[0].setExtraData(native_track.release());
     }
 
     blink::WebVector<blink::WebMediaStreamTrack> video_track_vector(
         video ? static_cast<size_t>(1) : 0);
     MediaStreamSource::SourceStoppedCallback dummy_callback;
     if (video) {
       blink::WebMediaStreamSource video_source;
       video_source.initialize("video",
                               blink::WebMediaStreamSource::TypeVideo,
                               "video",
@@ skipping 27 matching lines @@
     EXPECT_EQ(expected_number_of_video_tracks,
               adapter_->webrtc_media_stream()->GetVideoTracks().size());
     EXPECT_EQ(blink_stream.id().utf8(),
               adapter_->webrtc_media_stream()->label());
   }
 
   webrtc::MediaStreamInterface* webrtc_stream() {
     return adapter_->webrtc_media_stream();
   }
 
- private:
+ protected:
   base::MessageLoop message_loop_;
   std::unique_ptr<ChildProcess> child_process_;
   std::unique_ptr<MockPeerConnectionDependencyFactory> dependency_factory_;
   std::unique_ptr<WebRtcMediaStreamAdapter> adapter_;
-  MockAudioDeviceFactory mock_audio_device_factory_;
 };
 
 TEST_F(WebRtcMediaStreamAdapterTest, CreateWebRtcMediaStream) {
   blink::WebMediaStream blink_stream = CreateBlinkMediaStream(true, true);
   CreateWebRtcMediaStream(blink_stream, 1, 1);
 }
 
 // Test that we don't crash if a MediaStream is created in Blink with an unknown
 // audio sources. This can happen if a MediaStream is created with
 // remote audio track.
@@ skipping 37 matching lines @@
   EXPECT_TRUE(webrtc_stream()->GetVideoTracks().empty());
 
   native_stream->AddTrack(audio_tracks[0]);
   EXPECT_EQ(1u, webrtc_stream()->GetAudioTracks().size());
 
   native_stream->AddTrack(video_tracks[0]);
   EXPECT_EQ(1u, webrtc_stream()->GetVideoTracks().size());
 }
 
 }  // namespace content