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 "base/command_line.h" 6 #include "content/public/common/content_switches.h" 7 #include "content/renderer/media/mock_media_constraint_factory.h" 8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 9 #include "content/renderer/media/webrtc_local_audio_track.h" 10 #include "testing/gmock/include/gmock/gmock.h" 11 #include "testing/gtest/include/gtest/gtest.h" 12 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h" 13 14 using ::testing::_; 15 using ::testing::AnyNumber; 16 17 namespace content { 18 19 namespace { 20 21 class MockWebRtcAudioSink : public webrtc::AudioTrackSinkInterface { 22 public: 23 MockWebRtcAudioSink() {} 24 ~MockWebRtcAudioSink() {} 25 MOCK_METHOD5(OnData, void(const void* audio_data, 26 int bits_per_sample, 27 int sample_rate, 28 int number_of_channels, 29 int number_of_frames)); 30 }; 31 32 } // namespace 33 34 class WebRtcLocalAudioTrackAdapterTest : public ::testing::Test { 35 public: 36 WebRtcLocalAudioTrackAdapterTest() 37 : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, 38 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 480), 39 adapter_(WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)) { 40 MockMediaConstraintFactory constraint_factory; 41 capturer_ = WebRtcAudioCapturer::CreateCapturer( 42 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", ""), 43 constraint_factory.CreateWebMediaConstraints(), NULL, NULL); 44 track_.reset(new WebRtcLocalAudioTrack(adapter_, capturer_, NULL)); 45 } 46 47 protected: 48 virtual void SetUp() OVERRIDE { 49 track_->OnSetFormat(params_); 50 EXPECT_TRUE(track_->GetAudioAdapter()->enabled()); 51 } 52 53 media::AudioParameters params_; 54 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_; 55 scoped_refptr<WebRtcAudioCapturer> capturer_; 56 scoped_ptr<WebRtcLocalAudioTrack> track_; 57 }; 58 59 // Adds and Removes a WebRtcAudioSink to a local audio track. 60 TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) { 61 // Add a sink to the webrtc track. 62 scoped_ptr<MockWebRtcAudioSink> sink(new MockWebRtcAudioSink()); 63 webrtc::AudioTrackInterface* webrtc_track = 64 static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); 65 webrtc_track->AddSink(sink.get()); 66 67 // Send a packet via |track_| and it data should reach the sink of the 68 // |adapter_|. 69 const int length = params_.frames_per_buffer() * params_.channels(); 70 scoped_ptr<int16[]> data(new int16[length]); 71 // Initialize the data to 0 to avoid Memcheck:Uninitialized warning. 72 memset(data.get(), 0, length * sizeof(data[0])); 73 74 EXPECT_CALL(*sink, 75 OnData(_, 16, params_.sample_rate(), params_.channels(), 76 params_.frames_per_buffer())); 77 track_->Capture(data.get(), base::TimeDelta(), 255, false, false); 78 79 // Remove the sink from the webrtc track. 80 webrtc_track->RemoveSink(sink.get()); 81 sink.reset(); 82 83 // Verify that no more callback gets into the sink. 84 track_->Capture(data.get(), base::TimeDelta(), 255, false, false); 85 } 86 87 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) { 88 webrtc::AudioTrackInterface* webrtc_track = 89 static_cast<webrtc::AudioTrackInterface*>(adapter_.get()); 90 int signal_level = 0; 91 EXPECT_TRUE(webrtc_track->GetSignalLevel(&signal_level)); 92 93 // Disable the audio processing in the audio track. 94 CommandLine::ForCurrentProcess()->AppendSwitch( 95 switches::kDisableAudioTrackProcessing); 96 EXPECT_FALSE(webrtc_track->GetSignalLevel(&signal_level)); 97 } 98 99 } // namespace content 100
