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 <string>
   6 #include <vector>
   7
   8 #include "base/single_thread_task_runner.h"
   9 #include "content/public/renderer/media_stream_audio_renderer.h"
  10 #include "content/renderer/media/audio_device_factory.h"
  11 #include "content/renderer/media/audio_message_filter.h"
  12 #include "content/renderer/media/webrtc/mock_peer_connection_dependency_factory.h"
  13 #include "content/renderer/media/webrtc_audio_device_impl.h"
  14 #include "content/renderer/media/webrtc_audio_renderer.h"
  15 #include "media/audio/audio_output_device.h"
  16 #include "media/audio/audio_output_ipc.h"
  17 #include "media/base/audio_bus.h"
  18 #include "media/base/mock_audio_renderer_sink.h"
  19 #include "testing/gmock/include/gmock/gmock.h"
  20 #include "testing/gtest/include/gtest/gtest.h"
  21 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
  22
  23 using testing::Return;
  24
  25 namespace content {
  26
  27 namespace {
  28
  29 const int kHardwareSampleRate = 44100;
  30 const int kHardwareBufferSize = 512;
  31
  32 class MockAudioOutputIPC : public media::AudioOutputIPC {
  33  public:
  34   MockAudioOutputIPC() {}
  35   virtual ~MockAudioOutputIPC() {}
  36
  37   MOCK_METHOD3(CreateStream, void(media::AudioOutputIPCDelegate* delegate,
  38                                   const media::AudioParameters& params,
  39                                   int session_id));
  40   MOCK_METHOD0(PlayStream, void());
  41   MOCK_METHOD0(PauseStream, void());
  42   MOCK_METHOD0(CloseStream, void());
  43   MOCK_METHOD1(SetVolume, void(double volume));
  44   MOCK_METHOD3(SwitchOutputDevice,
  45                void(const std::string& device_id,
  46                     const GURL& security_origin,
  47                     int request_id));
  48 };
  49
  50 class FakeAudioOutputDevice
  51     : NON_EXPORTED_BASE(public media::AudioOutputDevice) {
  52  public:
  53   FakeAudioOutputDevice(
  54       scoped_ptr<media::AudioOutputIPC> ipc,
  55       const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner)
  56       : AudioOutputDevice(ipc.Pass(),
  57                           io_task_runner) {}
  58   MOCK_METHOD0(Start, void());
  59   MOCK_METHOD0(Stop, void());
  60   MOCK_METHOD0(Pause, void());
  61   MOCK_METHOD0(Play, void());
  62   MOCK_METHOD1(SetVolume, bool(double volume));
  63   MOCK_METHOD0(GetOutputDevice, media::OutputDevice*());
  64
  65  protected:
  66   virtual ~FakeAudioOutputDevice() {}
  67 };
  68
  69 class MockAudioDeviceFactory : public AudioDeviceFactory {
  70  public:
  71   MockAudioDeviceFactory() {}
  72   virtual ~MockAudioDeviceFactory() {}
  73   MOCK_METHOD1(CreateOutputDevice, media::AudioOutputDevice*(int));
  74   MOCK_METHOD1(CreateInputDevice, media::AudioInputDevice*(int));
  75 };
  76
  77 class MockAudioRendererSource : public WebRtcAudioRendererSource {
  78  public:
  79   MockAudioRendererSource() {}
  80   virtual ~MockAudioRendererSource() {}
  81   MOCK_METHOD4(RenderData, void(media::AudioBus* audio_bus,
  82                                 int sample_rate,
  83                                 int audio_delay_milliseconds,
  84                                 base::TimeDelta* current_time));
  85   MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
  86 };
  87
  88 }  // namespace
  89
  90 class WebRtcAudioRendererTest : public testing::Test {
  91  protected:
  92   WebRtcAudioRendererTest()
  93       : message_loop_(new base::MessageLoopForIO),
  94         mock_ipc_(new MockAudioOutputIPC()),
  95         mock_output_device_(new FakeAudioOutputDevice(
  96             scoped_ptr<media::AudioOutputIPC>(mock_ipc_),
  97             message_loop_->task_runner())),
  98         factory_(new MockAudioDeviceFactory()),
  99         source_(new MockAudioRendererSource()),
 100         stream_(new rtc::RefCountedObject<MockMediaStream>("label")),
 101         renderer_(new WebRtcAudioRenderer(message_loop_->task_runner(),
 102                                           stream_,
 103                                           1,
 104                                           1,
 105                                           44100,
 106                                           kHardwareBufferSize)) {
 107     EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
 108         .WillOnce(Return(mock_output_device_.get()));
 109     EXPECT_CALL(*mock_output_device_.get(), Start());
 110     EXPECT_TRUE(renderer_->Initialize(source_.get()));
 111     renderer_proxy_ = renderer_->CreateSharedAudioRendererProxy(stream_);
 112   }
 113
 114   // Used to construct |mock_output_device_|.
 115   scoped_ptr<base::MessageLoopForIO> message_loop_;
 116   MockAudioOutputIPC* mock_ipc_;  // Owned by AudioOuputDevice.
 117
 118   scoped_refptr<FakeAudioOutputDevice> mock_output_device_;
 119   scoped_ptr<MockAudioDeviceFactory> factory_;
 120   scoped_ptr<MockAudioRendererSource> source_;
 121   scoped_refptr<webrtc::MediaStreamInterface> stream_;
 122   scoped_refptr<WebRtcAudioRenderer> renderer_;
 123   scoped_refptr<MediaStreamAudioRenderer> renderer_proxy_;
 124 };
 125
 126 // Verify that the renderer will be stopped if the only proxy is stopped.
 127 TEST_F(WebRtcAudioRendererTest, StopRenderer) {
 128   renderer_proxy_->Start();
 129
 130   // |renderer_| has only one proxy, stopping the proxy should stop the sink of
 131   // |renderer_|.
 132   EXPECT_CALL(*mock_output_device_.get(), Stop());
 133   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 134   renderer_proxy_->Stop();
 135 }
 136
 137 // Verify that the renderer will not be stopped unless the last proxy is
 138 // stopped.
 139 TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
 140   renderer_proxy_->Start();
 141
 142   // Create a vector of renderer proxies from the |renderer_|.
 143   std::vector<scoped_refptr<MediaStreamAudioRenderer> > renderer_proxies_;
 144   static const int kNumberOfRendererProxy = 5;
 145   for (int i = 0; i < kNumberOfRendererProxy; ++i) {
 146     scoped_refptr<MediaStreamAudioRenderer> renderer_proxy(
 147         renderer_->CreateSharedAudioRendererProxy(stream_));
 148     renderer_proxy->Start();
 149     renderer_proxies_.push_back(renderer_proxy);
 150   }
 151
 152   // Stop the |renderer_proxy_| should not stop the sink since it is used by
 153   // other proxies.
 154   EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
 155   renderer_proxy_->Stop();
 156
 157   for (int i = 0; i < kNumberOfRendererProxy; ++i) {
 158     if (i != kNumberOfRendererProxy -1) {
 159       EXPECT_CALL(*mock_output_device_.get(), Stop()).Times(0);
 160     } else {
 161       // When the last proxy is stopped, the sink will stop.
 162       EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 163       EXPECT_CALL(*mock_output_device_.get(), Stop());
 164     }
 165     renderer_proxies_[i]->Stop();
 166   }
 167 }
 168
 169 // Verify that the sink of the renderer is using the expected sample rate and
 170 // buffer size.
 171 TEST_F(WebRtcAudioRendererTest, VerifySinkParameters) {
 172   renderer_proxy_->Start();
 173 #if defined(OS_LINUX) || defined(OS_MACOSX)
 174   static const int kExpectedBufferSize = kHardwareSampleRate / 100;
 175 #elif defined(OS_ANDROID)
 176   static const int kExpectedBufferSize = 2 * kHardwareSampleRate / 100;
 177 #else
 178   // Windows.
 179   static const int kExpectedBufferSize = kHardwareBufferSize;
 180 #endif
 181   EXPECT_EQ(kExpectedBufferSize, renderer_->frames_per_buffer());
 182   EXPECT_EQ(kHardwareSampleRate, renderer_->sample_rate());
 183   EXPECT_EQ(2, renderer_->channels());
 184
 185   EXPECT_CALL(*mock_output_device_.get(), Stop());
 186   EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
 187   renderer_proxy_->Stop();
 188 }
 189
 190 }  // namespace content