src/content/renderer/media/webrtc_audio_renderer_unittest.cc

   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 <vector>
   6
   7 #include "content/renderer/media/audio_device_factory.h"
   8 #include "content/renderer/media/audio_message_filter.h"
   9 #include "content/renderer/media/media_stream_audio_renderer.h"
  10 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
  11 #include "content/renderer/media/webrtc_audio_device_impl.h"
  12 #include "content/renderer/media/webrtc_audio_renderer.h"
  13 #include "media/audio/audio_output_device.h"
  14 #include "media/audio/audio_output_ipc.h"
  15 #include "media/base/audio_bus.h"
  16 #include "media/base/mock_audio_renderer_sink.h"
  17 #include "testing/gmock/include/gmock/gmock.h"
  18 #include "testing/gtest/include/gtest/gtest.h"
  19 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  20
  21 using testing::Return;
  22
  23 namespace content {
  24
  25 namespace {
  26
  27 const int kHardwareSampleRate = 44100;
  28 const int kHardwareBufferSize = 512;
  29
  30 class MockAudioOutputIPC : public media::AudioOutputIPC {
  31  public:
  32   MockAudioOutputIPC() {}
  33   virtual ~MockAudioOutputIPC() {}
  34
  35   MOCK_METHOD3(CreateStream, void(media::AudioOutputIPCDelegate* delegate,
  36                                   const media::AudioParameters& params,
  37                                   int session_id));
  38   MOCK_METHOD0(PlayStream, void());
  39   MOCK_METHOD0(PauseStream, void());
  40   MOCK_METHOD0(CloseStream, void());
  41   MOCK_METHOD1(SetVolume, void(double volume));
  42 };
  43
  44 class FakeAudioOutputDevice
  45     : NON_EXPORTED_BASE(public media::AudioOutputDevice) {
  46  public:
  47   FakeAudioOutputDevice(
  48       scoped_ptr<media::AudioOutputIPC> ipc,
  49       const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner)
  50       : AudioOutputDevice(ipc.Pass(),
  51                           io_task_runner) {}
  52   MOCK_METHOD0(Start, void());
  53   MOCK_METHOD0(Stop, void());
  54   MOCK_METHOD0(Pause, void());
  55   MOCK_METHOD0(Play, void());
  56   MOCK_METHOD1(SetVolume, bool(double volume));
  57
  58  protected:
  59   virtual ~FakeAudioOutputDevice() {}
  60 };
  61
  62 class MockAudioDeviceFactory : public AudioDeviceFactory {
  63  public:
  64   MockAudioDeviceFactory() {}
  65   virtual ~MockAudioDeviceFactory() {}
  66   MOCK_METHOD1(CreateOutputDevice, media::AudioOutputDevice*(int));
  67   MOCK_METHOD1(CreateInputDevice, media::AudioInputDevice*(int));
  68 };
  69
  70 class MockAudioRendererSource : public WebRtcAudioRendererSource {
  71  public:
  72   MockAudioRendererSource() {}
  73   virtual ~MockAudioRendererSource() {}
  74   MOCK_METHOD4(RenderData, void(media::AudioBus* audio_bus,
  75                                 int sample_rate,
  76                                 int audio_delay_milliseconds,
  77                                 base::TimeDelta* current_time));
  78   MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
  79 };
  80
  81 }  // namespace
  82
  83 class WebRtcAudioRendererTest : public testing::Test {
  84  protected:
  85   WebRtcAudioRendererTest()
  86       : message_loop_(new base::MessageLoopForIO),
  87         mock_ipc_(new MockAudioOutputIPC()),
  88         mock_output_device_(new FakeAudioOutputDevice(
  89             scoped_ptr<media::AudioOutputIPC>(mock_ipc_),
  90             message_loop_->message_loop_proxy())),
  91         factory_(new MockAudioDeviceFactory()),
  92         source_(new MockAudioRendererSource()),
  93         stream_(new rtc::RefCountedObject<MockMediaStream>("label")),
  94         renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100,
  95                                           kHardwareBufferSize)) {
  96     EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
  97         .WillOnce(Return(mock_output_device_.get()));
  98     EXPECT_CALL(*mock_output_device_.get(), Start());
  99     EXPECT_TRUE(renderer_->Initialize(source_.get()));
 100     renderer_proxy_ = renderer_->CreateSharedAudioRendererProxy(stream_);
 101   }
 102
 103   // Used to construct |mock_output_device_|.
 104   scoped_ptr<base::MessageLoopForIO> message_loop_;
 105   MockAudioOutputIPC* mock_ipc_;  // Owned by AudioOuputDevice.
 106
 107   scoped_refptr<FakeAudioOutputDevice> mock_output_device_;
 108   scoped_ptr<MockAudioDeviceFactory> factory_;
 109   scoped_ptr<MockAudioRendererSource> source_;
 110   scoped_refptr<webrtc::MediaStreamInterface> stream_;
 111   scoped_refptr<WebRtcAudioRenderer> renderer_;
 112   scoped_refptr<MediaStreamAudioRenderer> renderer_proxy_;
 113 };
 114
 115 // Verify that the renderer will be stopped if the only proxy is stopped.
 116 TEST_F(WebRtcAudioRendererTest, StopRenderer) {
 117   renderer_proxy_->Start();
 118
 119   // |renderer_| has only one proxy, stopping the proxy should stop the sink of
 120   // |renderer_|.
 121   EXPECT_CALL(*mock_output_device_.get(), Stop());
 122   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 123   renderer_proxy_->Stop();
 124 }
 125
 126 // Verify that the renderer will not be stopped unless the last proxy is
 127 // stopped.
 128 TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
 129   renderer_proxy_->Start();
 130
 131   // Create a vector of renderer proxies from the |renderer_|.
 132   std::vector<scoped_refptr<MediaStreamAudioRenderer> > renderer_proxies_;
 133   static const int kNumberOfRendererProxy = 5;
 134   for (int i = 0; i < kNumberOfRendererProxy; ++i) {
 135     scoped_refptr<MediaStreamAudioRenderer> renderer_proxy(
 136         renderer_->CreateSharedAudioRendererProxy(stream_));
 137     renderer_proxy->Start();
 138     renderer_proxies_.push_back(renderer_proxy);
 139   }
 140
 141   // Stop the |renderer_proxy_| should not stop the sink since it is used by
 142   // other proxies.
 143   EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
 144   renderer_proxy_->Stop();
 145
 146   for (int i = 0; i < kNumberOfRendererProxy; ++i) {
 147     if (i != kNumberOfRendererProxy -1) {
 148       EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
 149     } else {
 150       // When the last proxy is stopped, the sink will stop.
 151       EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 152       EXPECT_CALL(*mock_output_device_.get(), Stop());
 153     }
 154     renderer_proxies_[i]->Stop();
 155   }
 156 }
 157
 158 // Verify that the sink of the renderer is using the expected sample rate and
 159 // buffer size.
 160 TEST_F(WebRtcAudioRendererTest, VerifySinkParameters) {
 161   renderer_proxy_->Start();
 162 #if defined(OS_LINUX) || defined(OS_MACOSX)
 163   static const int kExpectedBufferSize = kHardwareSampleRate / 100;
 164 #elif defined(OS_ANDROID)
 165   static const int kExpectedBufferSize = 2 * kHardwareSampleRate / 100;
 166 #else
 167   // Windows.
 168   static const int kExpectedBufferSize = kHardwareBufferSize;
 169 #endif
 170   EXPECT_EQ(kExpectedBufferSize, renderer_->frames_per_buffer());
 171   EXPECT_EQ(kHardwareSampleRate, renderer_->sample_rate());
 172   EXPECT_EQ(2, renderer_->channels());
 173
 174   EXPECT_CALL(*mock_output_device_.get(), Stop());
 175   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 176   renderer_proxy_->Stop();
 177 }
 178
 179 }  // namespace content