1// Copyright 2014 The Chromium Authors. All rights reserved.
2// Use of this source code is governed by a BSD-style license that can be
3// found in the LICENSE file.
4
5#include "base/command_line.h"
6#include "content/public/common/content_switches.h"
7#include "content/renderer/media/mock_media_constraint_factory.h"
8#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
9#include "content/renderer/media/webrtc_local_audio_track.h"
10#include "testing/gmock/include/gmock/gmock.h"
11#include "testing/gtest/include/gtest/gtest.h"
12#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
13
14using ::testing::_;
15using ::testing::AnyNumber;
16
17namespace content {
18
19namespace {
20
21class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface {
22 public:
23  MockWebRtcAudioSink() {}
24  ~MockWebRtcAudioSink() {}
25  MOCK_METHOD5(OnData, void(const void* audio_data,
26                            int bits_per_sample,
27                            int sample_rate,
28                            int number_of_channels,
29                            int number_of_frames));
30};
31
32}  // namespace
33
34class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test {
35 public:
36  WebRtcLocalAudioTrackAdapterTest()
37      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
38                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480),
39        adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) {
40    MockMediaConstraintFactory constraint_factory;
41    capturer_ = WebRtcAudioCapturer::CreateCapturer(
42        -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""),
43        constraint_factory.CreateWebMediaConstraints(), NULL, NULL);
44    track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL));
45  }
46
47 protected:
48  virtual void SetUp() OVERRIDE {
49    track_->OnSetFormat(params_);
50    EXPECT_TRUE(track_->GetAudioAdapter()->enabled());
51  }
52
53  media::AudioParameters params_;
54  scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_;
55  scoped_refptr<WebRtcAudioCapturer> capturer_;
56  scoped_ptr<WebRtcLocalAudioTrack> track_;
57};
58
59// Adds and Removes a WebRtcAudioSink to a local audio track.
60TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
61  // Add a sink to the webrtc track.
62  scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink());
63  webrtc::AudioTrackInterface* webrtc_track =
64      static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
65  webrtc_track->AddSink(sink.get());
66
67  // Send a packet via |track_| and it data should reach the sink of the
68  // |adapter_|.
69  const int length = params_.frames_per_buffer() * params_.channels();
70  scoped_ptr<int16[]> data(new int16[length]);
71  // Initialize the data to 0 to avoid Memcheck:Uninitialized warning.
72  memset(data.get(), 0, length * sizeof(data[0]));
73
74  EXPECT_CALL(*sink,
75              OnData(_, 16, params_.sample_rate(), params_.channels(),
76                     params_.frames_per_buffer()));
77  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
78
79  // Remove the sink from the webrtc track.
80  webrtc_track->RemoveSink(sink.get());
81  sink.reset();
82
83  // Verify that no more callback gets into the sink.
84  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
85}
86
87TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {
88  webrtc::AudioTrackInterface* webrtc_track =
89      static_cast<webrtc::AudioTrackInterface*>(adapter_.get());
90  int signal_level = 0;
91  EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level));
92
93  // Disable the audio processing in the audio track.
94  CommandLine::ForCurrentProcess()->AppendSwitch(
95      switches::kDisableAudioTrackProcessing);
96  EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level));
97}
98
99}  // namespace content
100