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

content/renderer/media/audio_track_recorder_unittest.cc

 // Copyright 2015 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/audio_track_recorder.h"
 
 #include <stdint.h>
 
+#include "base/bind.h"
+#include "base/bind_helpers.h"
 #include "base/macros.h"
 #include "base/run_loop.h"
 #include "base/stl_util.h"
-#include "base/strings/utf_string_conversions.h"
 #include "content/renderer/media/media_stream_audio_source.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_local_audio_track.h"
 #include "media/audio/simple_sources.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
+#include "third_party/WebKit/public/platform/WebString.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 #include "third_party/opus/src/include/opus.h"
 
 using ::testing::_;
 using ::testing::DoAll;
 using ::testing::InSequence;
 using ::testing::Mock;
 using ::testing::Return;
 using ::testing::SaveArg;
 using ::testing::TestWithParam;
@@ skipping 170 matching lines @@
   media::SineWaveAudioSource first_source_;
   media::SineWaveAudioSource second_source_;
 
   // Decoder for verifying data was properly encoded.
   OpusDecoder* opus_decoder_;
   scoped_ptr<float[]> buffer_;
 
  private:
   // Prepares a blink track of a given MediaStreamType and attaches the native
   // track, which can be used to capture audio data and pass it to the producer.
-  // Adapted from media::WebRTCLocalAudioSourceProviderTest.
   void PrepareBlinkTrack() {
-    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));
     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 */);
+    audio_source.setExtraData(
+        new MediaStreamAudioSource(false /* is_remote */));
     blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
                             audio_source);
-    blink_track_.setExtraData(native_track.release());
+    ASSERT_TRUE(MediaStreamAudioSource::Get(audio_source)
+                    ->ConnectToTrack(blink_track_));
   }
 
   DISALLOW_COPY_AND_ASSIGN(AudioTrackRecorderTest);
 };
 
 TEST_P(AudioTrackRecorderTest, OnData) {
   InSequence s;
   base::RunLoop run_loop;
   base::Closure quit_closure = run_loop.QuitClosure();
 
@@ skipping 43 matching lines @@
                                   base::TimeTicks::Now());
   }
 
   run_loop.Run();
   Mock::VerifyAndClearExpectations(this);
 }
 
 INSTANTIATE_TEST_CASE_P(, AudioTrackRecorderTest, ValuesIn(kATRTestParams));
 
 }  // namespace content