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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 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 "base/logging.h" | 5 #include "base/logging.h" |
6 #include "build/build_config.h" | 6 #include "build/build_config.h" |
7 #include "content/public/renderer/media_stream_audio_sink.h" | 7 #include "content/public/renderer/media_stream_audio_sink.h" |
8 #include "content/renderer/media/mock_constraint_factory.h" | 8 #include "content/renderer/media/mock_constraint_factory.h" |
9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 9 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
10 #include "content/renderer/media/webrtc_audio_capturer.h" | 10 #include "content/renderer/media/webrtc_audio_capturer.h" |
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69 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) { | 69 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 960) { |
70 // Android works with a buffer size bigger than 20ms. | 70 // Android works with a buffer size bigger than 20ms. |
71 #else | 71 #else |
72 : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 72 : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
73 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) { | 73 media::CHANNEL_LAYOUT_STEREO, 48000, 16, 128) { |
74 #endif | 74 #endif |
75 } | 75 } |
76 | 76 |
77 void VerifyAudioParams(const blink::WebMediaConstraints& constraints, | 77 void VerifyAudioParams(const blink::WebMediaConstraints& constraints, |
78 bool need_audio_processing) { | 78 bool need_audio_processing) { |
79 const scoped_ptr<WebRtcAudioCapturer> capturer = | 79 capturer_ = WebRtcAudioCapturer::CreateCapturer( |
80 WebRtcAudioCapturer::CreateCapturer( | 80 -1, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
81 -1, StreamDeviceInfo( | 81 params_.sample_rate(), params_.channel_layout(), |
82 MEDIA_DEVICE_AUDIO_CAPTURE, "", "", params_.sample_rate(), | 82 params_.frames_per_buffer()), |
83 params_.channel_layout(), params_.frames_per_buffer()), | 83 constraints, NULL, NULL); |
84 constraints, nullptr, nullptr); | 84 capturer_source_ = new MockCapturerSource(); |
85 const scoped_refptr<MockCapturerSource> capturer_source( | 85 EXPECT_CALL(*capturer_source_.get(), Initialize(_, capturer_.get(), -1)); |
86 new MockCapturerSource()); | 86 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); |
87 EXPECT_CALL(*capturer_source.get(), Initialize(_, capturer.get(), -1)); | 87 EXPECT_CALL(*capturer_source_.get(), Start()); |
88 EXPECT_CALL(*capturer_source.get(), SetAutomaticGainControl(true)); | 88 capturer_->SetCapturerSource(capturer_source_, params_); |
89 EXPECT_CALL(*capturer_source.get(), Start()); | |
90 capturer->SetCapturerSource(capturer_source, params_); | |
91 | 89 |
92 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 90 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
93 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 91 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
94 const scoped_ptr<WebRtcLocalAudioTrack> track( | 92 track_.reset(new WebRtcLocalAudioTrack(adapter.get(), capturer_, NULL)); |
95 new WebRtcLocalAudioTrack(adapter.get())); | 93 track_->Start(); |
96 capturer->AddTrack(track.get()); | |
97 | 94 |
98 // Connect a mock sink to the track. | 95 // Connect a mock sink to the track. |
99 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); | 96 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); |
100 track->AddSink(sink.get()); | 97 track_->AddSink(sink.get()); |
101 | 98 |
102 int delay_ms = 65; | 99 int delay_ms = 65; |
103 bool key_pressed = true; | 100 bool key_pressed = true; |
104 double volume = 0.9; | 101 double volume = 0.9; |
105 | 102 |
106 scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_); | 103 scoped_ptr<media::AudioBus> audio_bus = media::AudioBus::Create(params_); |
107 audio_bus->Zero(); | 104 audio_bus->Zero(); |
108 | 105 |
109 media::AudioCapturerSource::CaptureCallback* callback = | 106 media::AudioCapturerSource::CaptureCallback* callback = |
110 static_cast<media::AudioCapturerSource::CaptureCallback*>( | 107 static_cast<media::AudioCapturerSource::CaptureCallback*>( |
111 capturer.get()); | 108 capturer_.get()); |
112 | 109 |
113 // Verify the sink is getting the correct values. | 110 // Verify the sink is getting the correct values. |
114 EXPECT_CALL(*sink, FormatIsSet()); | 111 EXPECT_CALL(*sink, FormatIsSet()); |
115 EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1)); | 112 EXPECT_CALL(*sink, OnDataCallback()).Times(AtLeast(1)); |
116 callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed); | 113 callback->Capture(audio_bus.get(), delay_ms, volume, key_pressed); |
117 | 114 |
118 track->RemoveSink(sink.get()); | 115 track_->RemoveSink(sink.get()); |
119 EXPECT_CALL(*capturer_source.get(), Stop()); | 116 EXPECT_CALL(*capturer_source_.get(), Stop()); |
120 capturer->Stop(); | 117 capturer_->Stop(); |
121 } | 118 } |
122 | 119 |
123 media::AudioParameters params_; | 120 media::AudioParameters params_; |
| 121 scoped_refptr<MockCapturerSource> capturer_source_; |
| 122 scoped_refptr<WebRtcAudioCapturer> capturer_; |
| 123 scoped_ptr<WebRtcLocalAudioTrack> track_; |
124 }; | 124 }; |
125 | 125 |
126 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { | 126 TEST_F(WebRtcAudioCapturerTest, VerifyAudioParamsWithAudioProcessing) { |
127 // Turn off the default constraints to verify that the sink will get packets | 127 // Turn off the default constraints to verify that the sink will get packets |
128 // with a buffer size smaller than 10ms. | 128 // with a buffer size smaller than 10ms. |
129 MockConstraintFactory constraint_factory; | 129 MockConstraintFactory constraint_factory; |
130 constraint_factory.DisableDefaultAudioConstraints(); | 130 constraint_factory.DisableDefaultAudioConstraints(); |
131 VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false); | 131 VerifyAudioParams(constraint_factory.CreateWebMediaConstraints(), false); |
132 } | 132 } |
133 | 133 |
134 TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) { | 134 TEST_F(WebRtcAudioCapturerTest, FailToCreateCapturerWithWrongConstraints) { |
135 MockConstraintFactory constraint_factory; | 135 MockConstraintFactory constraint_factory; |
136 const std::string dummy_constraint = "dummy"; | 136 const std::string dummy_constraint = "dummy"; |
137 // Set a non-audio constraint. | 137 // Set a non-audio constraint. |
138 constraint_factory.basic().width.setExact(240); | 138 constraint_factory.basic().width.setExact(240); |
139 | 139 |
140 scoped_ptr<WebRtcAudioCapturer> capturer(WebRtcAudioCapturer::CreateCapturer( | 140 scoped_refptr<WebRtcAudioCapturer> capturer( |
141 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", | 141 WebRtcAudioCapturer::CreateCapturer( |
142 params_.sample_rate(), params_.channel_layout(), | 142 0, StreamDeviceInfo(MEDIA_DEVICE_AUDIO_CAPTURE, "", "", |
143 params_.frames_per_buffer()), | 143 params_.sample_rate(), params_.channel_layout(), |
144 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); | 144 params_.frames_per_buffer()), |
| 145 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
145 EXPECT_TRUE(capturer.get() == NULL); | 146 EXPECT_TRUE(capturer.get() == NULL); |
146 } | 147 } |
147 | 148 |
148 | 149 |
149 } // namespace content | 150 } // namespace content |
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