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| 1 // Copyright 2015 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 "content/renderer/media/audio_track_recorder.h" | |
| 6 | |
| 7 #include "base/run_loop.h" | |
| 8 #include "base/strings/utf_string_conversions.h" | |
| 9 #include "content/renderer/media/media_stream_audio_source.h" | |
| 10 #include "content/renderer/media/mock_media_constraint_factory.h" | |
| 11 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | |
| 12 #include "content/renderer/media/webrtc_local_audio_track.h" | |
| 13 #include "media/audio/simple_sources.h" | |
| 14 #include "testing/gmock/include/gmock/gmock.h" | |
| 15 #include "testing/gtest/include/gtest/gtest.h" | |
| 16 #include "third_party/WebKit/public/web/WebHeap.h" | |
| 17 | |
| 18 using ::testing::_; | |
| 19 using ::testing::DoAll; | |
| 20 using ::testing::InSequence; | |
| 21 using ::testing::Mock; | |
| 22 using ::testing::Return; | |
| 23 using ::testing::SaveArg; | |
| 24 | |
| 25 namespace { | |
| 26 | |
| 27 // Input audio format. | |
| 28 const media::AudioParameters::Format kInputFormat = | |
| 29 media::AudioParameters::AUDIO_PCM_LOW_LATENCY; | |
| 30 const int kBitsPerSample = 16; | |
| 31 const int kSamplingRate = 48000; | |
| 32 const int kFramesPerBuffer = 480; | |
| 33 | |
| 34 } // namespace | |
| 35 | |
| 36 namespace content { | |
| 37 | |
| 38 ACTION_P(RunClosure, closure) { | |
| 39 closure.Run(); | |
| 40 } | |
| 41 | |
| 42 class AudioTrackRecorderTest : public testing::Test { | |
| 43 public: | |
| 44 AudioTrackRecorderTest() | |
| 45 : params1_(kInputFormat, | |
| 46 media::CHANNEL_LAYOUT_MONO, | |
| 47 kSamplingRate, | |
| 48 kBitsPerSample, | |
| 49 kFramesPerBuffer), | |
| 50 params2_(kInputFormat, | |
| 51 media::CHANNEL_LAYOUT_STEREO, | |
| 52 kSamplingRate, | |
| 53 kBitsPerSample, | |
| 54 kFramesPerBuffer), | |
| 55 mono_source_(1 /* # channels */, 440, kSamplingRate), | |
| 56 stereo_source_(2 /* # channels */, 440, kSamplingRate) { | |
| 57 PrepareBlinkTrack(); | |
| 58 audio_track_recorder_.reset(new AudioTrackRecorder( | |
| 59 blink_track_, base::Bind(&AudioTrackRecorderTest::OnEncodedAudio, | |
| 60 base::Unretained(this)))); | |
| 61 } | |
| 62 | |
| 63 ~AudioTrackRecorderTest() { | |
| 64 audio_track_recorder_.reset(); | |
| 65 blink_track_.reset(); | |
| 66 blink::WebHeap::collectAllGarbageForTesting(); | |
| 67 } | |
| 68 | |
| 69 scoped_ptr<media::AudioBus> NextAudioBus(int num_channels, | |
| 70 const base::TimeDelta& duration) { | |
| 71 // Only supports up to two channels for now. | |
| 72 EXPECT_TRUE(num_channels == 1 || num_channels == 2); | |
| 73 const int num_samples = static_cast<int>((kSamplingRate * duration) / | |
| 74 base::TimeDelta::FromSeconds(1)); | |
| 75 scoped_ptr<media::AudioBus> bus( | |
| 76 media::AudioBus::Create(num_channels, num_samples)); | |
| 77 if (num_channels == 1) { | |
| 78 mono_source_.OnMoreData(bus.get(), 0); | |
| 79 } else { | |
| 80 // two channels | |
|
mcasas
2015/10/29 22:58:21
Nit: remove this comment and with that, also the c
ajose
2015/11/02 20:12:51
Done.
| |
| 81 stereo_source_.OnMoreData(bus.get(), 0); | |
| 82 } | |
| 83 return bus.Pass(); | |
| 84 } | |
| 85 | |
| 86 MOCK_METHOD3(DoOnEncodedAudio, | |
| 87 void(const media::AudioParameters& params, | |
| 88 std::string encoded_data, | |
| 89 base::TimeTicks timestamp)); | |
| 90 | |
| 91 void OnEncodedAudio(const media::AudioParameters& params, | |
| 92 scoped_ptr<std::string> encoded_data, | |
| 93 base::TimeTicks timestamp) { | |
| 94 EXPECT_TRUE(!encoded_data->empty()); | |
| 95 DoOnEncodedAudio(params, *encoded_data, timestamp); | |
| 96 } | |
| 97 | |
| 98 const base::MessageLoop message_loop_; | |
| 99 | |
| 100 // ATR and WebMediaStreamTrack for fooling it. | |
| 101 scoped_ptr<AudioTrackRecorder> audio_track_recorder_; | |
| 102 blink::WebMediaStreamTrack blink_track_; | |
| 103 | |
| 104 // Two different sets of AudioParameters for testing re-init of ATR. | |
| 105 media::AudioParameters params1_; | |
|
mcasas
2015/10/29 22:58:21
nit: I think |params1|, |params2|, |mono_source||
ajose
2015/11/02 20:12:51
media::SineWaveAudioSource doesn't like being cons
| |
| 106 media::AudioParameters params2_; | |
| 107 | |
| 108 // AudioSources for creating AudioBuses. | |
| 109 media::SineWaveAudioSource mono_source_; | |
| 110 media::SineWaveAudioSource stereo_source_; | |
| 111 | |
| 112 private: | |
| 113 // Prepares a blink track of a given MediaStreamType and attaches the native | |
| 114 // track, which can be used to capture audio data and pass it to the producer. | |
| 115 // Adapted from media::WebRTCLocalAudioSourceProviderTest. | |
| 116 void PrepareBlinkTrack() { | |
| 117 MockMediaConstraintFactory constraint_factory; | |
| 118 scoped_refptr<WebRtcAudioCapturer> capturer( | |
| 119 WebRtcAudioCapturer::CreateCapturer( | |
| 120 -1, StreamDeviceInfo(), | |
| 121 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); | |
| 122 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | |
| 123 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | |
| 124 scoped_ptr<WebRtcLocalAudioTrack> native_track( | |
| 125 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | |
| 126 blink::WebMediaStreamSource audio_source; | |
| 127 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | |
| 128 blink::WebMediaStreamSource::TypeAudio, | |
| 129 base::UTF8ToUTF16("dummy_source_name"), | |
| 130 false /* remote */, true /* readonly */); | |
| 131 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), | |
| 132 audio_source); | |
| 133 blink_track_.setExtraData(native_track.release()); | |
| 134 } | |
| 135 | |
| 136 DISALLOW_COPY_AND_ASSIGN(AudioTrackRecorderTest); | |
| 137 }; | |
| 138 | |
| 139 TEST_F(AudioTrackRecorderTest, OnData) { | |
| 140 InSequence s; | |
| 141 base::RunLoop run_loop; | |
| 142 base::Closure quit_closure = run_loop.QuitClosure(); | |
| 143 | |
| 144 // Give ATR initial audio parameters. | |
| 145 audio_track_recorder_->OnSetFormat(params1_); | |
| 146 // TODO(ajose): consider adding WillOnce(SaveArg...) and inspecting, as done | |
| 147 // in VTR unittests. http://crbug.com/548856 | |
| 148 const base::TimeTicks time1 = base::TimeTicks::Now(); | |
| 149 EXPECT_CALL(*this, DoOnEncodedAudio(_, _, time1)).Times(1); | |
| 150 audio_track_recorder_->OnData( | |
| 151 *NextAudioBus(params1_.channels(), base::TimeDelta::FromMilliseconds(10)), | |
| 152 time1); | |
| 153 | |
| 154 // Send more audio. | |
| 155 const base::TimeTicks time2 = base::TimeTicks::Now(); | |
| 156 EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _)).Times(1); | |
| 157 audio_track_recorder_->OnData( | |
| 158 *NextAudioBus(params1_.channels(), base::TimeDelta::FromMilliseconds(10)), | |
| 159 time2); | |
| 160 | |
| 161 // Give ATR new audio parameters. | |
| 162 audio_track_recorder_->OnSetFormat(params2_); | |
| 163 // Send audio with different params. | |
| 164 const base::TimeTicks time3 = base::TimeTicks::Now(); | |
| 165 EXPECT_CALL(*this, DoOnEncodedAudio(_, _, _)) | |
| 166 .Times(1) | |
| 167 .WillOnce(RunClosure(quit_closure)); | |
| 168 audio_track_recorder_->OnData( | |
| 169 *NextAudioBus(params2_.channels(), base::TimeDelta::FromMilliseconds(10)), | |
| 170 time3); | |
| 171 | |
| 172 run_loop.Run(); | |
| 173 Mock::VerifyAndClearExpectations(this); | |
| 174 } | |
| 175 | |
| 176 } // namespace content | |
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