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

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..0a30d4ec0e3c7c9f2983cefd5d43336bbeae4bde
--- /dev/null
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
@@ -0,0 +1,102 @@
+// 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