Index: content/renderer/media/webrtc_audio_capturer_unittest.cc |
diff --git a/content/renderer/media/webrtc_audio_capturer_unittest.cc b/content/renderer/media/webrtc_audio_capturer_unittest.cc |
deleted file mode 100644 |
index 306ca9882d773e0e80188b1b052a290482f209d1..0000000000000000000000000000000000000000 |
--- a/content/renderer/media/webrtc_audio_capturer_unittest.cc |
+++ /dev/null |
@@ -1,149 +0,0 @@ |
-// Copyright 2013 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 "base/logging.h" |
-#include "build/build_config.h" |
-#include "content/public/renderer/media_stream_audio_sink.h" |
-#include "content/renderer/media/mock_media_constraint_factory.h" |
-#include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
-#include "content/renderer/media/webrtc_audio_capturer.h" |
-#include "content/renderer/media/webrtc_local_audio_track.h" |
-#include "media/audio/audio_parameters.h" |
-#include "media/base/audio_bus.h" |
-#include "testing/gmock/include/gmock/gmock.h" |
-#include "testing/gtest/include/gtest/gtest.h" |
-#include "third_party/WebKit/public/platform/WebMediaConstraints.h" |
- |
-using ::testing::_; |
-using ::testing::AtLeast; |
- |
-namespace content { |
- |
-namespace { |
- |
-class MockCapturerSource : public media::AudioCapturerSource { |
- public: |
- MockCapturerSource() {} |
- MOCK_METHOD3(Initialize, void(const media::AudioParameters& params, |
- CaptureCallback* callback, |
- int session_id)); |
- MOCK_METHOD0(Start, void()); |
- MOCK_METHOD0(Stop, void()); |
- MOCK_METHOD1(SetVolume, void(double volume)); |
- MOCK_METHOD1(SetAutomaticGainControl, void(bool enable)); |
- |
- protected: |
- ~MockCapturerSource() override {} |
-}; |
- |
-class MockMediaStreamAudioSink : public MediaStreamAudioSink { |
- public: |
- MockMediaStreamAudioSink() {} |
- ~MockMediaStreamAudioSink() override {} |
- void OnData(const media::AudioBus& audio_bus, |
- base::TimeTicks estimated_capture_time) override { |
- EXPECT_EQ(audio_bus.channels(), params_.channels()); |
- EXPECT_EQ(audio_bus.frames(), params_.frames_per_buffer()); |
- EXPECT_FALSE(estimated_capture_time.is_null()); |
- OnDataCallback(); |
- } |
- MOCK_METHOD0(OnDataCallback, void()); |
- void OnSetFormat(const media::AudioParameters& params) override { |
- params_ = params; |
- FormatIsSet(); |
- } |
- MOCK_METHOD0(FormatIsSet, void()); |
- |
- private: |
- media::AudioParameters params_; |
-}; |
- |
-} // namespace |
- |
-class WebRtcAudioCapturerTest : public testing::Test { |
- protected: |
- WebRtcAudioCapturerTest() |
-#if defined(OS_ANDROID) |
- : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) { |
- // Android works with a buffer size bigger than 20ms. |
-#else |
- : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
- media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) { |
-#endif |
- } |
- |
- void VerifyAudioParams(const blink::WebMediaConstraints& constraints, |
- bool need_audio_processing) { |
- capturer_ = WebRtcAudioCapturer::CreateCapturer( |
- -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
- params_.sample_rate(), params_.channel_layout(), |
- params_.frames_per_buffer()), |
- constraints, NULL, NULL); |
- capturer_source_ = new MockCapturerSource(); |
- EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
- EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
- EXPECT_CALL(*capturer_source_.get(), Start()); |
- capturer_->SetCapturerSource(capturer_source_, params_); |
- |
- scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
- WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
- track_.reset(new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
- track_->Start(); |
- |
- // Connect a mock sink to the track. |
- scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
- track_->AddSink(sink.get()); |
- |
- int delay_ms = 65; |
- bool key_pressed = true; |
- double volume = 0.9; |
- |
- scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_); |
- audio_bus->Zero(); |
- |
- media::AudioCapturerSource::CaptureCallback* callback = |
- static_cast<media::AudioCapturerSource::CaptureCallback*>( |
- capturer_.get()); |
- |
- // Verify the sink is getting the correct values. |
- EXPECT_CALL(*sink, FormatIsSet()); |
- EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1)); |
- callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed); |
- |
- track_->RemoveSink(sink.get()); |
- EXPECT_CALL(*capturer_source_.get(), Stop()); |
- capturer_->Stop(); |
- } |
- |
- media::AudioParameters params_; |
- scoped_refptr<MockCapturerSource> capturer_source_; |
- scoped_refptr<WebRtcAudioCapturer> capturer_; |
- scoped_ptr<WebRtcLocalAudioTrack> track_; |
-}; |
- |
-TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { |
- // Turn off the default constraints to verify that the sink will get packets |
- // with a buffer size smaller than 10ms. |
- MockMediaConstraintFactory constraint_factory; |
- constraint_factory.DisableDefaultAudioConstraints(); |
- VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false); |
-} |
- |
-TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) { |
- MockMediaConstraintFactory constraint_factory; |
- const std::string dummy_constraint = "dummy"; |
- constraint_factory.AddMandatory(dummy_constraint, true); |
- |
- scoped_refptr<WebRtcAudioCapturer> capturer( |
- WebRtcAudioCapturer::CreateCapturer( |
- 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
- params_.sample_rate(), params_.channel_layout(), |
- params_.frames_per_buffer()), |
- constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
- EXPECT_TRUE(capturer.get() == NULL); |
-} |
- |
- |
-} // namespace content |