Roll ANGLE bc75f36:ef9d63e
[chromium-blink-merge.git] / content / renderer / media / webrtc_audio_renderer_unittest.cc
blob914dd52b776a9f9b0175dc78dbe73a932955209a
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.
5 #include <vector>
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"
21 using testing::Return;
23 namespace content {
25 namespace {
27 class MockAudioOutputIPC : public media::AudioOutputIPC {
28 public:
29 MockAudioOutputIPC() {}
30 virtual ~MockAudioOutputIPC() {}
32 MOCK_METHOD3(CreateStream, void(media::AudioOutputIPCDelegate* delegate,
33 const media::AudioParameters& params,
34 int session_id));
35 MOCK_METHOD0(PlayStream, void());
36 MOCK_METHOD0(PauseStream, void());
37 MOCK_METHOD0(CloseStream, void());
38 MOCK_METHOD1(SetVolume, void(double volume));
41 class FakeAudioOutputDevice
42 : NON_EXPORTED_BASE(public media::AudioOutputDevice) {
43 public:
44 FakeAudioOutputDevice(
45 scoped_ptr<media::AudioOutputIPC> ipc,
46 const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner)
47 : AudioOutputDevice(ipc.Pass(),
48 io_task_runner) {}
49 MOCK_METHOD0(Start, void());
50 MOCK_METHOD0(Stop, void());
51 MOCK_METHOD0(Pause, void());
52 MOCK_METHOD0(Play, void());
53 MOCK_METHOD1(SetVolume, bool(double volume));
55 protected:
56 virtual ~FakeAudioOutputDevice() {}
59 class MockAudioDeviceFactory : public AudioDeviceFactory {
60 public:
61 MockAudioDeviceFactory() {}
62 virtual ~MockAudioDeviceFactory() {}
63 MOCK_METHOD1(CreateOutputDevice, media::AudioOutputDevice*(int));
64 MOCK_METHOD1(CreateInputDevice, media::AudioInputDevice*(int));
67 class MockAudioRendererSource : public WebRtcAudioRendererSource {
68 public:
69 MockAudioRendererSource() {}
70 virtual ~MockAudioRendererSource() {}
71 MOCK_METHOD4(RenderData, void(media::AudioBus* audio_bus,
72 int sample_rate,
73 int audio_delay_milliseconds,
74 base::TimeDelta* current_time));
75 MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
78 } // namespace
80 class WebRtcAudioRendererTest : public testing::Test {
81 protected:
82 WebRtcAudioRendererTest()
83 : message_loop_(new base::MessageLoopForIO),
84 mock_ipc_(new MockAudioOutputIPC()),
85 mock_output_device_(new FakeAudioOutputDevice(
86 scoped_ptr<media::AudioOutputIPC>(mock_ipc_),
87 message_loop_->message_loop_proxy())),
88 factory_(new MockAudioDeviceFactory()),
89 source_(new MockAudioRendererSource()),
90 stream_(new rtc::RefCountedObject<MockMediaStream>("label")),
91 renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100, 441)) {
92 EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
93 .WillOnce(Return(mock_output_device_));
94 EXPECT_CALL(*mock_output_device_, Start());
95 EXPECT_TRUE(renderer_->Initialize(source_.get()));
96 renderer_proxy_ = renderer_->CreateSharedAudioRendererProxy(stream_);
99 // Used to construct |mock_output_device_|.
100 scoped_ptr<base::MessageLoopForIO> message_loop_;
101 MockAudioOutputIPC* mock_ipc_; // Owned by AudioOuputDevice.
103 scoped_refptr<FakeAudioOutputDevice> mock_output_device_;
104 scoped_ptr<MockAudioDeviceFactory> factory_;
105 scoped_ptr<MockAudioRendererSource> source_;
106 scoped_refptr<webrtc::MediaStreamInterface> stream_;
107 scoped_refptr<WebRtcAudioRenderer> renderer_;
108 scoped_refptr<MediaStreamAudioRenderer> renderer_proxy_;
111 // Verify that the renderer will be stopped if the only proxy is stopped.
112 TEST_F(WebRtcAudioRendererTest, StopRenderer) {
113 renderer_proxy_->Start();
115 // |renderer_| has only one proxy, stopping the proxy should stop the sink of
116 // |renderer_|.
117 EXPECT_CALL(*mock_output_device_, Stop());
118 EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
119 renderer_proxy_->Stop();
122 // Verify that the renderer will not be stopped unless the last proxy is
123 // stopped.
124 TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
125 renderer_proxy_->Start();
127 // Create a vector of renderer proxies from the |renderer_|.
128 std::vector<scoped_refptr<MediaStreamAudioRenderer> > renderer_proxies_;
129 static const int kNumberOfRendererProxy = 5;
130 for (int i = 0; i < kNumberOfRendererProxy; ++i) {
131 scoped_refptr<MediaStreamAudioRenderer> renderer_proxy(
132 renderer_->CreateSharedAudioRendererProxy(stream_));
133 renderer_proxy->Start();
134 renderer_proxies_.push_back(renderer_proxy);
137 // Stop the |renderer_proxy_| should not stop the sink since it is used by
138 // other proxies.
139 EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
140 renderer_proxy_->Stop();
142 for (int i = 0; i < kNumberOfRendererProxy; ++i) {
143 if (i != kNumberOfRendererProxy -1) {
144 EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
145 } else {
146 // When the last proxy is stopped, the sink will stop.
147 EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
148 EXPECT_CALL(*mock_output_device_, Stop());
150 renderer_proxies_[i]->Stop();
154 } // namespace content