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

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/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;
-
-namespace content {
-
-namespace {
-
-class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface {
- public:
-  MockWebRtcAudioSink() {}
-  ~MockWebRtcAudioSink() {}
-  MOCK_METHOD5(OnData, void(const void* audio_data,
-                            int bits_per_sample,
-                            int sample_rate,
-                            size_t number_of_channels,
-                            size_t number_of_frames));
-};
-
-}  // 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)) {
-    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_;
-  std::unique_ptr<WebRtcLocalAudioTrack> track_;
-};
-
-// Adds and Removes a WebRtcAudioSink to a local audio track.
-TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
-  // Add a sink to the webrtc track.
-  std::unique_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 std::unique_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);
-
-  // 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);
-}
-
-TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
-  webrtc::AudioTrackInterface* webrtc_track =
-      static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
-  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