MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc/webrtc_local_audio_track_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 <stddef.h>
 
-#include "content/renderer/media/mock_media_constraint_factory.h"
+#include "content/renderer/media/media_stream_audio_level_calculator.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_local_audio_track.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
 
 using ::testing::_;
 using ::testing::AnyNumber;
 
@@ skipping 13 matching lines @@
 };
 
 }  // namespace
 
 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
  public:
   WebRtcLocalAudioTrackAdapterTest()
       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
-    MockMediaConstraintFactory constraint_factory;
-    capturer_ = WebRtcAudioCapturer::CreateCapturer(
-        -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
-        constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
-    track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
+    track_.reset(new WebRtcLocalAudioTrack(adapter_.get()));
   }
 
  protected:
   void SetUp() override {
     track_->OnSetFormat(params_);
     EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
   }
 
   media::AudioParameters params_;
   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
-  scoped_refptr<WebRtcAudioCapturer> capturer_;
   scoped_ptr<WebRtcLocalAudioTrack> track_;
 };
 
 // Adds and Removes a WebRtcAudioSink to a local audio track.
 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   // Add a sink to the webrtc track.
   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
   webrtc_track->AddSink(sink.get());
 
   // Send a packet via |track_| and the data should reach the sink of the
   // |adapter_|.
   const scoped_ptr<media::AudioBus> audio_bus =
       media::AudioBus::Create(params_);
   // While this test is not checking the signal data being passed around, the
   // implementation in WebRtcLocalAudioTrack reads the data for its signal level
   // computation.  Initialize all samples to zero to make the memory sanitizer
   // happy.
   audio_bus->Zero();
 
   base::TimeTicks estimated_capture_time = base::TimeTicks::Now();
   EXPECT_CALL(*sink,
               OnData(_, 16, params_.sample_rate(), params_.channels(),
                      params_.frames_per_buffer()));
-  track_->Capture(*audio_bus, estimated_capture_time, false);
+  track_->Capture(*audio_bus, estimated_capture_time);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
   estimated_capture_time +=
       params_.frames_per_buffer() * base::TimeDelta::FromSeconds(1) /
           params_.sample_rate();
-  track_->Capture(*audio_bus, estimated_capture_time, false);
+  track_->Capture(*audio_bus, estimated_capture_time);
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
   webrtc::AudioTrackInterface* webrtc_track =
       static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
-  int signal_level = 0;
+  int signal_level = -1;
+  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
+  MediaStreamAudioLevelCalculator calculator;
+  adapter_->SetLevel(calculator.level());
+  signal_level = -1;
   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
+  EXPECT_EQ(0, signal_level);
 }
 
 }  // namespace content