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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 <stddef.h> | 5 #include <stddef.h> |
6 | 6 |
7 #include "base/logging.h" | 7 #include "base/logging.h" |
8 #include "base/strings/utf_string_conversions.h" | |
9 #include "content/renderer/media/mock_media_constraint_factory.h" | |
10 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
11 #include "content/renderer/media/webrtc_audio_capturer.h" | |
12 #include "content/renderer/media/webrtc_local_audio_source_provider.h" | 9 #include "content/renderer/media/webrtc_local_audio_source_provider.h" |
13 #include "content/renderer/media/webrtc_local_audio_track.h" | 10 #include "content/renderer/media/webrtc_local_audio_track.h" |
14 #include "media/audio/audio_parameters.h" | 11 #include "media/audio/audio_parameters.h" |
15 #include "media/base/audio_bus.h" | 12 #include "media/base/audio_bus.h" |
16 #include "testing/gtest/include/gtest/gtest.h" | 13 #include "testing/gtest/include/gtest/gtest.h" |
17 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 14 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| 15 #include "third_party/WebKit/public/platform/WebString.h" |
18 #include "third_party/WebKit/public/web/WebHeap.h" | 16 #include "third_party/WebKit/public/web/WebHeap.h" |
19 | 17 |
20 namespace content { | 18 namespace content { |
21 | 19 |
22 class WebRtcLocalAudioSourceProviderTest : public testing::Test { | 20 class WebRtcLocalAudioSourceProviderTest : public testing::Test { |
23 protected: | 21 protected: |
24 void SetUp() override { | 22 void SetUp() override { |
25 source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 23 source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
26 media::CHANNEL_LAYOUT_MONO, 48000, 16, 480); | 24 media::CHANNEL_LAYOUT_MONO, 48000, 16, 480); |
27 sink_params_.Reset( | 25 sink_params_.Reset( |
28 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 26 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
29 media::CHANNEL_LAYOUT_STEREO, 44100, 16, | 27 media::CHANNEL_LAYOUT_STEREO, 44100, 16, |
30 WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize); | 28 WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize); |
31 sink_bus_ = media::AudioBus::Create(sink_params_); | 29 sink_bus_ = media::AudioBus::Create(sink_params_); |
32 MockMediaConstraintFactory constraint_factory; | |
33 scoped_refptr<WebRtcAudioCapturer> capturer( | |
34 WebRtcAudioCapturer::CreateCapturer( | |
35 -1, StreamDeviceInfo(), | |
36 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); | |
37 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 30 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
38 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 31 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
39 scoped_ptr<WebRtcLocalAudioTrack> native_track( | 32 scoped_ptr<WebRtcLocalAudioTrack> native_track( |
40 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); | 33 new WebRtcLocalAudioTrack(adapter.get())); |
41 blink::WebMediaStreamSource audio_source; | 34 blink::WebMediaStreamSource audio_source; |
42 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), | 35 audio_source.initialize(blink::WebString::fromUTF8("dummy_source_id"), |
43 blink::WebMediaStreamSource::TypeAudio, | 36 blink::WebMediaStreamSource::TypeAudio, |
44 base::UTF8ToUTF16("dummy_source_name"), | 37 blink::WebString::fromUTF8("dummy_source_name"), |
45 false /* remote */, true /* readonly */); | 38 false /* remote */, true /* readonly */); |
46 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), | 39 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), |
47 audio_source); | 40 audio_source); |
48 blink_track_.setExtraData(native_track.release()); | 41 blink_track_.setExtraData(native_track.release()); |
49 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_)); | 42 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_)); |
50 source_provider_->SetSinkParamsForTesting(sink_params_); | 43 source_provider_->SetSinkParamsForTesting(sink_params_); |
51 source_provider_->OnSetFormat(source_params_); | 44 source_provider_->OnSetFormat(source_params_); |
52 } | 45 } |
53 | 46 |
54 void TearDown() override { | 47 void TearDown() override { |
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136 // Stop the audio track. | 129 // Stop the audio track. |
137 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>( | 130 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>( |
138 MediaStreamTrack::GetTrack(blink_track_)); | 131 MediaStreamTrack::GetTrack(blink_track_)); |
139 native_track->Stop(); | 132 native_track->Stop(); |
140 | 133 |
141 // Delete the source provider. | 134 // Delete the source provider. |
142 source_provider_.reset(); | 135 source_provider_.reset(); |
143 } | 136 } |
144 | 137 |
145 } // namespace content | 138 } // namespace content |
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