| Index: content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| diff --git a/content/renderer/media/webrtc_audio_renderer_unittest.cc b/content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| new file mode 100644
|
| index 0000000000000000000000000000000000000000..34d0c7582fca8182e677002653aa3cd34b89b6d3
|
| --- /dev/null
|
| +++ b/content/renderer/media/webrtc_audio_renderer_unittest.cc
|
| @@ -0,0 +1,153 @@
|
| +// 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 <vector>
|
| +
|
| +#include "content/renderer/media/audio_device_factory.h"
|
| +#include "content/renderer/media/audio_message_filter.h"
|
| +#include "content/renderer/media/media_stream_audio_renderer.h"
|
| +#include "content/renderer/media/mock_media_stream_dependency_factory.h"
|
| +#include "content/renderer/media/webrtc_audio_device_impl.h"
|
| +#include "content/renderer/media/webrtc_audio_renderer.h"
|
| +#include "media/audio/audio_output_device.h"
|
| +#include "media/audio/audio_output_ipc.h"
|
| +#include "media/base/audio_bus.h"
|
| +#include "media/base/mock_audio_renderer_sink.h"
|
| +#include "testing/gmock/include/gmock/gmock.h"
|
| +#include "testing/gtest/include/gtest/gtest.h"
|
| +#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
|
| +
|
| +using testing::Return;
|
| +
|
| +namespace content {
|
| +
|
| +namespace {
|
| +
|
| +class MockAudioOutputIPC : public media::AudioOutputIPC {
|
| + public:
|
| + MockAudioOutputIPC() {}
|
| + virtual ~MockAudioOutputIPC() {}
|
| +
|
| + MOCK_METHOD3(CreateStream, void(media::AudioOutputIPCDelegate* delegate,
|
| + const media::AudioParameters& params,
|
| + int session_id));
|
| + MOCK_METHOD0(PlayStream, void());
|
| + MOCK_METHOD0(PauseStream, void());
|
| + MOCK_METHOD0(CloseStream, void());
|
| + MOCK_METHOD1(SetVolume, void(double volume));
|
| +};
|
| +
|
| +class FakeAudioOutputDevice
|
| + : NON_EXPORTED_BASE(public media::AudioOutputDevice) {
|
| + public:
|
| + FakeAudioOutputDevice(
|
| + scoped_ptr<media::AudioOutputIPC> ipc,
|
| + const scoped_refptr<base::SingleThreadTaskRunner>& io_task_runner)
|
| + : AudioOutputDevice(ipc.Pass(),
|
| + io_task_runner) {}
|
| + MOCK_METHOD0(Start, void());
|
| + MOCK_METHOD0(Stop, void());
|
| + MOCK_METHOD0(Pause, void());
|
| + MOCK_METHOD0(Play, void());
|
| + MOCK_METHOD1(SetVolume, bool(double volume));
|
| +
|
| + protected:
|
| + virtual ~FakeAudioOutputDevice() {}
|
| +};
|
| +
|
| +class MockAudioDeviceFactory : public AudioDeviceFactory {
|
| + public:
|
| + MockAudioDeviceFactory() {}
|
| + virtual ~MockAudioDeviceFactory() {}
|
| + MOCK_METHOD1(CreateOutputDevice, media::AudioOutputDevice*(int));
|
| + MOCK_METHOD1(CreateInputDevice, media::AudioInputDevice*(int));
|
| +};
|
| +
|
| +class MockAudioRendererSource : public WebRtcAudioRendererSource {
|
| + public:
|
| + MockAudioRendererSource() {}
|
| + virtual ~MockAudioRendererSource() {}
|
| + MOCK_METHOD3(RenderData, void(media::AudioBus* audio_bus,
|
| + int sample_rate,
|
| + int audio_delay_milliseconds));
|
| + MOCK_METHOD1(RemoveAudioRenderer, void(WebRtcAudioRenderer* renderer));
|
| +};
|
| +
|
| +} // namespace
|
| +
|
| +class WebRtcAudioRendererTest : public testing::Test {
|
| + protected:
|
| + WebRtcAudioRendererTest()
|
| + : message_loop_(new base::MessageLoopForIO),
|
| + mock_ipc_(new MockAudioOutputIPC()),
|
| + mock_output_device_(new FakeAudioOutputDevice(
|
| + scoped_ptr<media::AudioOutputIPC>(mock_ipc_),
|
| + message_loop_->message_loop_proxy())),
|
| + factory_(new MockAudioDeviceFactory()),
|
| + source_(new MockAudioRendererSource()),
|
| + stream_(new talk_base::RefCountedObject<MockMediaStream>("label")),
|
| + renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100, 441)) {
|
| + EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
|
| + .WillOnce(Return(mock_output_device_));
|
| + EXPECT_CALL(*mock_output_device_, Start());
|
| + EXPECT_TRUE(renderer_->Initialize(source_.get()));
|
| + renderer_proxy_ = renderer_->CreateSharedAudioRendererProxy(stream_);
|
| + }
|
| +
|
| + // Used to construct |mock_output_device_|.
|
| + scoped_ptr<base::MessageLoopForIO> message_loop_;
|
| + MockAudioOutputIPC* mock_ipc_; // Owned by AudioOuputDevice.
|
| +
|
| + scoped_refptr<FakeAudioOutputDevice> mock_output_device_;
|
| + scoped_ptr<MockAudioDeviceFactory> factory_;
|
| + scoped_ptr<MockAudioRendererSource> source_;
|
| + scoped_refptr<webrtc::MediaStreamInterface> stream_;
|
| + scoped_refptr<WebRtcAudioRenderer> renderer_;
|
| + scoped_refptr<MediaStreamAudioRenderer> renderer_proxy_;
|
| +};
|
| +
|
| +// Verify that the renderer will be stopped if the only proxy is stopped.
|
| +TEST_F(WebRtcAudioRendererTest, StopRenderer) {
|
| + renderer_proxy_->Start();
|
| +
|
| + // |renderer_| has only one proxy, stopping the proxy should stop the sink of
|
| + // |renderer_|.
|
| + EXPECT_CALL(*mock_output_device_, Stop());
|
| + EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
|
| + renderer_proxy_->Stop();
|
| +}
|
| +
|
| +// Verify that the renderer will not be stopped unless the last proxy is
|
| +// stopped.
|
| +TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
|
| + renderer_proxy_->Start();
|
| +
|
| + // Create a vector of renderer proxies from the |renderer_|.
|
| + std::vector<scoped_refptr<MediaStreamAudioRenderer> > renderer_proxies_;
|
| + static const int kNumberOfRendererProxy = 5;
|
| + for (int i = 0; i < kNumberOfRendererProxy; ++i) {
|
| + scoped_refptr<MediaStreamAudioRenderer> renderer_proxy(
|
| + renderer_->CreateSharedAudioRendererProxy(stream_));
|
| + renderer_proxy->Start();
|
| + renderer_proxies_.push_back(renderer_proxy);
|
| + }
|
| +
|
| + // Stop the |renderer_proxy_| should not stop the sink since it is used by
|
| + // other proxies.
|
| + EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
|
| + renderer_proxy_->Stop();
|
| +
|
| + for (int i = 0; i < kNumberOfRendererProxy; ++i) {
|
| + if (i != kNumberOfRendererProxy -1) {
|
| + EXPECT_CALL(*mock_output_device_, Stop()).Times(0);
|
| + } else {
|
| + // When the last proxy is stopped, the sink will stop.
|
| + EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
|
| + EXPECT_CALL(*mock_output_device_, Stop());
|
| + }
|
| + renderer_proxies_[i]->Stop();
|
| + }
|
| +}
|
| +
|
| +} // namespace content
|
|
|