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|    1 // Copyright 2014 The Chromium Authors. All rights reserved. |    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 |    2 // Use of this source code is governed by a BSD-style license that can be | 
|    3 // found in the LICENSE file. |    3 // found in the LICENSE file. | 
|    4  |    4  | 
|    5 #include <stddef.h> |    5 #include <stddef.h> | 
|    6  |    6  | 
|    7 #include "content/renderer/media/media_stream_audio_level_calculator.h" |    7 #include "content/renderer/media/mock_media_constraint_factory.h" | 
|    8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |    8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 
|    9 #include "content/renderer/media/webrtc_audio_capturer.h" |    9 #include "content/renderer/media/webrtc_audio_capturer.h" | 
|   10 #include "content/renderer/media/webrtc_local_audio_track.h" |   10 #include "content/renderer/media/webrtc_local_audio_track.h" | 
|   11 #include "testing/gmock/include/gmock/gmock.h" |   11 #include "testing/gmock/include/gmock/gmock.h" | 
|   12 #include "testing/gtest/include/gtest/gtest.h" |   12 #include "testing/gtest/include/gtest/gtest.h" | 
|   13 #include "third_party/webrtc/api/mediastreaminterface.h" |   13 #include "third_party/webrtc/api/mediastreaminterface.h" | 
|   14  |   14  | 
|   15 using ::testing::_; |   15 using ::testing::_; | 
|   16 using ::testing::AnyNumber; |   16 using ::testing::AnyNumber; | 
|   17  |   17  | 
| (...skipping 13 matching lines...) Expand all  Loading... | 
|   31 }; |   31 }; | 
|   32  |   32  | 
|   33 }  // namespace |   33 }  // namespace | 
|   34  |   34  | 
|   35 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test { |   35 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test { | 
|   36  public: |   36  public: | 
|   37   WebRtcLocalAudioTrackAdapterTest() |   37   WebRtcLocalAudioTrackAdapterTest() | 
|   38       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |   38       : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 
|   39                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480), |   39                 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480), | 
|   40         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) { |   40         adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) { | 
|   41     track_.reset(new WebRtcLocalAudioTrack(adapter_.get())); |   41     MockMediaConstraintFactory constraint_factory; | 
 |   42     capturer_ = WebRtcAudioCapturer::CreateCapturer( | 
 |   43         -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""), | 
 |   44         constraint_factory.CreateWebMediaConstraints(), NULL, NULL); | 
 |   45     track_.reset(new WebRtcLocalAudioTrack(adapter_.get(), capturer_, NULL)); | 
|   42   } |   46   } | 
|   43  |   47  | 
|   44  protected: |   48  protected: | 
|   45   void SetUp() override { |   49   void SetUp() override { | 
|   46     track_->OnSetFormat(params_); |   50     track_->OnSetFormat(params_); | 
|   47     EXPECT_TRUE(track_->GetAudioAdapter()->enabled()); |   51     EXPECT_TRUE(track_->GetAudioAdapter()->enabled()); | 
|   48   } |   52   } | 
|   49  |   53  | 
|   50   media::AudioParameters params_; |   54   media::AudioParameters params_; | 
|   51   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; |   55   scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; | 
 |   56   scoped_refptr<WebRtcAudioCapturer> capturer_; | 
|   52   scoped_ptr<WebRtcLocalAudioTrack> track_; |   57   scoped_ptr<WebRtcLocalAudioTrack> track_; | 
|   53 }; |   58 }; | 
|   54  |   59  | 
|   55 // Adds and Removes a WebRtcAudioSink to a local audio track. |   60 // Adds and Removes a WebRtcAudioSink to a local audio track. | 
|   56 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) { |   61 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) { | 
|   57   // Add a sink to the webrtc track. |   62   // Add a sink to the webrtc track. | 
|   58   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink()); |   63   scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink()); | 
|   59   webrtc::AudioTrackInterface* webrtc_track = |   64   webrtc::AudioTrackInterface* webrtc_track = | 
|   60       static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); |   65       static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); | 
|   61   webrtc_track->AddSink(sink.get()); |   66   webrtc_track->AddSink(sink.get()); | 
|   62  |   67  | 
|   63   // Send a packet via |track_| and the data should reach the sink of the |   68   // Send a packet via |track_| and the data should reach the sink of the | 
|   64   // |adapter_|. |   69   // |adapter_|. | 
|   65   const scoped_ptr<media::AudioBus> audio_bus = |   70   const scoped_ptr<media::AudioBus> audio_bus = | 
|   66       media::AudioBus::Create(params_); |   71       media::AudioBus::Create(params_); | 
|   67   // While this test is not checking the signal data being passed around, the |   72   // While this test is not checking the signal data being passed around, the | 
|   68   // implementation in WebRtcLocalAudioTrack reads the data for its signal level |   73   // implementation in WebRtcLocalAudioTrack reads the data for its signal level | 
|   69   // computation.  Initialize all samples to zero to make the memory sanitizer |   74   // computation.  Initialize all samples to zero to make the memory sanitizer | 
|   70   // happy. |   75   // happy. | 
|   71   audio_bus->Zero(); |   76   audio_bus->Zero(); | 
|   72  |   77  | 
|   73   base::TimeTicks estimated_capture_time = base::TimeTicks::Now(); |   78   base::TimeTicks estimated_capture_time = base::TimeTicks::Now(); | 
|   74   EXPECT_CALL(*sink, |   79   EXPECT_CALL(*sink, | 
|   75               OnData(_, 16, params_.sample_rate(), params_.channels(), |   80               OnData(_, 16, params_.sample_rate(), params_.channels(), | 
|   76                      params_.frames_per_buffer())); |   81                      params_.frames_per_buffer())); | 
|   77   track_->Capture(*audio_bus, estimated_capture_time); |   82   track_->Capture(*audio_bus, estimated_capture_time, false); | 
|   78  |   83  | 
|   79   // Remove the sink from the webrtc track. |   84   // Remove the sink from the webrtc track. | 
|   80   webrtc_track->RemoveSink(sink.get()); |   85   webrtc_track->RemoveSink(sink.get()); | 
|   81   sink.reset(); |   86   sink.reset(); | 
|   82  |   87  | 
|   83   // Verify that no more callback gets into the sink. |   88   // Verify that no more callback gets into the sink. | 
|   84   estimated_capture_time += |   89   estimated_capture_time += | 
|   85       params_.frames_per_buffer() * base::TimeDelta::FromSeconds(1) / |   90       params_.frames_per_buffer() * base::TimeDelta::FromSeconds(1) / | 
|   86           params_.sample_rate(); |   91           params_.sample_rate(); | 
|   87   track_->Capture(*audio_bus, estimated_capture_time); |   92   track_->Capture(*audio_bus, estimated_capture_time, false); | 
|   88 } |   93 } | 
|   89  |   94  | 
|   90 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) { |   95 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) { | 
|   91   webrtc::AudioTrackInterface* webrtc_track = |   96   webrtc::AudioTrackInterface* webrtc_track = | 
|   92       static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); |   97       static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); | 
|   93   int signal_level = -1; |   98   int signal_level = 0; | 
|   94   EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level)); |  | 
|   95   MediaStreamAudioLevelCalculator calculator; |  | 
|   96   adapter_->SetLevel(calculator.level()); |  | 
|   97   signal_level = -1; |  | 
|   98   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); |   99   EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); | 
|   99   EXPECT_EQ(0, signal_level); |  | 
|  100 } |  100 } | 
|  101  |  101  | 
|  102 }  // namespace content |  102 }  // namespace content | 
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