Revert of MediaStream audio object graph untangling and clean-ups.

content/renderer/media/speech_recognition_audio_sink_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/speech_recognition_audio_sink.h"
 
 #include <stddef.h>
 #include <stdint.h>
 #include <string.h>
 #include <utility>
 
 #include "base/bind.h"
 #include "base/macros.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/audio_parameters.h"
 #include "media/base/audio_bus.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h"
 #include "third_party/WebKit/public/web/WebHeap.h"
 
 namespace {
@@ skipping 241 matching lines @@
 
   // Mock callback expected to be called when the track is stopped.
   MOCK_METHOD0(StoppedCallback, void());
 
  protected:
   // 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.
   static void PrepareBlinkTrackOfType(
       const MediaStreamType device_type,
       blink::WebMediaStreamTrack* blink_track) {
+    StreamDeviceInfo device_info(device_type, "Mock device",
+                                 "mock_device_id");
+    MockMediaConstraintFactory constraint_factory;
+    const blink::WebMediaConstraints constraints =
+        constraint_factory.CreateWebMediaConstraints();
+    scoped_refptr<WebRtcAudioCapturer> capturer(
+        WebRtcAudioCapturer::CreateCapturer(-1, device_info, constraints, NULL,
+                                            NULL));
     scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
         WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
     scoped_ptr<WebRtcLocalAudioTrack> native_track(
-        new WebRtcLocalAudioTrack(adapter.get()));
+        new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
     blink::WebMediaStreamSource blink_audio_source;
     blink_audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
                                   blink::WebMediaStreamSource::TypeAudio,
                                   base::UTF8ToUTF16("dummy_source_name"),
                                   false /* remote */, true /* readonly */);
     MediaStreamSource::SourceStoppedCallback cb;
-    blink_audio_source.setExtraData(new MediaStreamAudioSource(
-        -1, StreamDeviceInfo(device_type, "Mock device", "mock_device_id"), cb,
-        nullptr));
+    blink_audio_source.setExtraData(
+        new MediaStreamAudioSource(-1, device_info, cb, NULL));
     blink_track->initialize(blink::WebString::fromUTF8("dummy_track"),
                             blink_audio_source);
     blink_track->setExtraData(native_track.release());
   }
 
   // Emulates an audio capture device capturing data from the source.
   inline void CaptureAudio(const uint32_t buffers) {
     for (uint32_t i = 0; i < buffers; ++i) {
       const base::TimeTicks estimated_capture_time = first_frame_capture_time_ +
           (sample_frames_captured_ * base::TimeDelta::FromSeconds(1) /
                source_params_.sample_rate());
-      native_track()->Capture(*source_bus_, estimated_capture_time);
+      native_track()->Capture(*source_bus_, estimated_capture_time, false);
       sample_frames_captured_ += source_bus_->frames();
     }
   }
 
   // Used to simulate a problem with sockets.
   void SetFailureModeOnForeignSocket(bool in_failure_mode) {
     recognizer()->sending_socket()->SetFailureMode(in_failure_mode);
   }
 
   // Helper method for verifying captured audio data has been consumed.
@@ skipping 208 matching lines @@
   const uint32_t buffers_per_notification = Initialize(44100, 441, 16000, 1600);
   AssertConsumedBuffers(0U);
   CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U);
   EXPECT_CALL(*this, StoppedCallback()).Times(1);
 
   native_track()->Stop();
   CaptureAudioAndAssertConsumedBuffers(buffers_per_notification, 1U);
 }
 
 }  // namespace content