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Side by Side Diff: content/renderer/media/audio_track_recorder_unittest.cc

Issue 1721273002: MediaStream audio object graph untangling and clean-ups. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: REBASE Created 4 years, 9 months ago
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1 // Copyright 2015 The Chromium Authors. All rights reserved. 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 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 "content/renderer/media/audio_track_recorder.h" 5 #include "content/renderer/media/audio_track_recorder.h"
6 6
7 #include <stdint.h> 7 #include <stdint.h>
8 8
9 #include "base/macros.h" 9 #include "base/macros.h"
10 #include "base/run_loop.h" 10 #include "base/run_loop.h"
11 #include "base/stl_util.h" 11 #include "base/stl_util.h"
12 #include "base/strings/utf_string_conversions.h" 12 #include "base/strings/utf_string_conversions.h"
13 #include "content/renderer/media/media_stream_audio_source.h" 13 #include "content/renderer/media/media_stream_audio_source.h"
14 #include "content/renderer/media/mock_media_constraint_factory.h"
15 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 14 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
16 #include "content/renderer/media/webrtc_local_audio_track.h" 15 #include "content/renderer/media/webrtc_local_audio_track.h"
17 #include "media/audio/simple_sources.h" 16 #include "media/audio/simple_sources.h"
18 #include "testing/gmock/include/gmock/gmock.h" 17 #include "testing/gmock/include/gmock/gmock.h"
19 #include "testing/gtest/include/gtest/gtest.h" 18 #include "testing/gtest/include/gtest/gtest.h"
20 #include "third_party/WebKit/public/web/WebHeap.h" 19 #include "third_party/WebKit/public/web/WebHeap.h"
21 #include "third_party/opus/src/include/opus.h" 20 #include "third_party/opus/src/include/opus.h"
22 21
23 using ::testing::_; 22 using ::testing::_;
24 using ::testing::DoAll; 23 using ::testing::DoAll;
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202 201
203 // Decoder for verifying data was properly encoded. 202 // Decoder for verifying data was properly encoded.
204 OpusDecoder* opus_decoder_; 203 OpusDecoder* opus_decoder_;
205 scoped_ptr<float[]> buffer_; 204 scoped_ptr<float[]> buffer_;
206 205
207 private: 206 private:
208 // Prepares a blink track of a given MediaStreamType and attaches the native 207 // Prepares a blink track of a given MediaStreamType and attaches the native
209 // track, which can be used to capture audio data and pass it to the producer. 208 // track, which can be used to capture audio data and pass it to the producer.
210 // Adapted from media::WebRTCLocalAudioSourceProviderTest. 209 // Adapted from media::WebRTCLocalAudioSourceProviderTest.
211 void PrepareBlinkTrack() { 210 void PrepareBlinkTrack() {
212 MockMediaConstraintFactory constraint_factory;
213 scoped_refptr<WebRtcAudioCapturer> capturer(
214 WebRtcAudioCapturer::CreateCapturer(
215 -1, StreamDeviceInfo(),
216 constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
217 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( 211 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
218 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); 212 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
219 scoped_ptr<WebRtcLocalAudioTrack> native_track( 213 scoped_ptr<WebRtcLocalAudioTrack> native_track(
220 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); 214 new WebRtcLocalAudioTrack(adapter.get()));
221 blink::WebMediaStreamSource audio_source; 215 blink::WebMediaStreamSource audio_source;
222 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), 216 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
223 blink::WebMediaStreamSource::TypeAudio, 217 blink::WebMediaStreamSource::TypeAudio,
224 base::UTF8ToUTF16("dummy_source_name"), 218 base::UTF8ToUTF16("dummy_source_name"),
225 false /* remote */, true /* readonly */); 219 false /* remote */, true /* readonly */);
226 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), 220 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
227 audio_source); 221 audio_source);
228 blink_track_.setExtraData(native_track.release()); 222 blink_track_.setExtraData(native_track.release());
229 } 223 }
230 224
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282 base::TimeTicks::Now()); 276 base::TimeTicks::Now());
283 } 277 }
284 278
285 run_loop.Run(); 279 run_loop.Run();
286 Mock::VerifyAndClearExpectations(this); 280 Mock::VerifyAndClearExpectations(this);
287 } 281 }
288 282
289 INSTANTIATE_TEST_CASE_P(, AudioTrackRecorderTest, ValuesIn(kATRTestParams)); 283 INSTANTIATE_TEST_CASE_P(, AudioTrackRecorderTest, ValuesIn(kATRTestParams));
290 284
291 } // namespace content 285 } // namespace content
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