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

Issue 883293005: Cast: Basic cast_receiver API for chrome. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: removed extra BUILD.gn line Created 5 years, 10 months ago
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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/synchronization/waitable_event.h" 5 #include "base/synchronization/waitable_event.h"
6 #include "base/test/test_timeouts.h" 6 #include "base/test/test_timeouts.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/media_stream_audio_source.h" 8 #include "content/renderer/media/media_stream_audio_source.h"
9 #include "content/renderer/media/mock_media_constraint_factory.h" 9 #include "content/renderer/media/mock_media_constraint_factory.h"
10 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 10 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
(...skipping 153 matching lines...) Expand 10 before | Expand all | Expand 10 after
164 std::string(), std::string()); 164 std::string(), std::string());
165 capturer_ = WebRtcAudioCapturer::CreateCapturer( 165 capturer_ = WebRtcAudioCapturer::CreateCapturer(
166 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL, 166 -1, device, constraint_factory.CreateWebMediaConstraints(), NULL,
167 audio_source); 167 audio_source);
168 audio_source->SetAudioCapturer(capturer_.get()); 168 audio_source->SetAudioCapturer(capturer_.get());
169 capturer_source_ = new MockCapturerSource(capturer_.get()); 169 capturer_source_ = new MockCapturerSource(capturer_.get());
170 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1)) 170 EXPECT_CALL(*capturer_source_.get(), OnInitialize(_, capturer_.get(), -1))
171 .WillOnce(Return()); 171 .WillOnce(Return());
172 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true)); 172 EXPECT_CALL(*capturer_source_.get(), SetAutomaticGainControl(true));
173 EXPECT_CALL(*capturer_source_.get(), OnStart()); 173 EXPECT_CALL(*capturer_source_.get(), OnStart());
174 capturer_->SetCapturerSourceForTesting(capturer_source_, params_); 174 capturer_->SetCapturerSource(capturer_source_, params_);
175 } 175 }
176 176
177 void TearDown() override { 177 void TearDown() override {
178 blink_source_.reset(); 178 blink_source_.reset();
179 blink::WebHeap::collectAllGarbageForTesting(); 179 blink::WebHeap::collectAllGarbageForTesting();
180 } 180 }
181 181
182 media::AudioParameters params_; 182 media::AudioParameters params_;
183 blink::WebMediaStreamSource blink_source_; 183 blink::WebMediaStreamSource blink_source_;
184 scoped_refptr<MockCapturerSource> capturer_source_; 184 scoped_refptr<MockCapturerSource> capturer_source_;
(...skipping 236 matching lines...) Expand 10 before | Expand all | Expand 10 after
421 NULL)); 421 NULL));
422 scoped_refptr<MockCapturerSource> new_source( 422 scoped_refptr<MockCapturerSource> new_source(
423 new MockCapturerSource(new_capturer.get())); 423 new MockCapturerSource(new_capturer.get()));
424 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1)); 424 EXPECT_CALL(*new_source.get(), OnInitialize(_, new_capturer.get(), -1));
425 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true)); 425 EXPECT_CALL(*new_source.get(), SetAutomaticGainControl(true));
426 EXPECT_CALL(*new_source.get(), OnStart()); 426 EXPECT_CALL(*new_source.get(), OnStart());
427 427
428 media::AudioParameters new_param( 428 media::AudioParameters new_param(
429 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, 429 media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
430 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441); 430 media::CHANNEL_LAYOUT_MONO, 44100, 16, 441);
431 new_capturer->SetCapturerSourceForTesting(new_source, new_param); 431 new_capturer->SetCapturerSource(new_source, new_param);
432 432
433 // Setup the second audio track, connect it to the new capturer and start it. 433 // Setup the second audio track, connect it to the new capturer and start it.
434 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2( 434 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter_2(
435 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); 435 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
436 scoped_ptr<WebRtcLocalAudioTrack> track_2( 436 scoped_ptr<WebRtcLocalAudioTrack> track_2(
437 new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL)); 437 new WebRtcLocalAudioTrack(adapter_2.get(), new_capturer, NULL));
438 track_2->Start(); 438 track_2->Start();
439 439
440 // Verify the data flow by connecting the |sink_2| to |track_2|. 440 // Verify the data flow by connecting the |sink_2| to |track_2|.
441 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink()); 441 scoped_ptr<MockMediaStreamAudioSink> sink_2(new MockMediaStreamAudioSink());
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475 "", "", params.sample_rate(), 475 "", "", params.sample_rate(),
476 params.channel_layout(), 476 params.channel_layout(),
477 params.frames_per_buffer()), 477 params.frames_per_buffer()),
478 factory.CreateWebMediaConstraints(), 478 factory.CreateWebMediaConstraints(),
479 NULL, NULL)); 479 NULL, NULL));
480 scoped_refptr<MockCapturerSource> source( 480 scoped_refptr<MockCapturerSource> source(
481 new MockCapturerSource(capturer.get())); 481 new MockCapturerSource(capturer.get()));
482 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1)); 482 EXPECT_CALL(*source.get(), OnInitialize(_, capturer.get(), -1));
483 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true)); 483 EXPECT_CALL(*source.get(), SetAutomaticGainControl(true));
484 EXPECT_CALL(*source.get(), OnStart()); 484 EXPECT_CALL(*source.get(), OnStart());
485 capturer->SetCapturerSourceForTesting(source, params); 485 capturer->SetCapturerSource(source, params);
486 486
487 // Setup a audio track, connect it to the capturer and start it. 487 // Setup a audio track, connect it to the capturer and start it.
488 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( 488 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
489 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); 489 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
490 scoped_ptr<WebRtcLocalAudioTrack> track( 490 scoped_ptr<WebRtcLocalAudioTrack> track(
491 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); 491 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
492 track->Start(); 492 track->Start();
493 493
494 // Verify the data flow by connecting the |sink| to |track|. 494 // Verify the data flow by connecting the |sink| to |track|.
495 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink()); 495 scoped_ptr<MockMediaStreamAudioSink> sink(new MockMediaStreamAudioSink());
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510 // Stopping the new source will stop the second track. 510 // Stopping the new source will stop the second track.
511 EXPECT_CALL(*source.get(), OnStop()).Times(1); 511 EXPECT_CALL(*source.get(), OnStop()).Times(1);
512 capturer->Stop(); 512 capturer->Stop();
513 513
514 // Even though this test don't use |capturer_source_| it will be stopped 514 // Even though this test don't use |capturer_source_| it will be stopped
515 // during teardown of the test harness. 515 // during teardown of the test harness.
516 EXPECT_CALL(*capturer_source_.get(), OnStop()); 516 EXPECT_CALL(*capturer_source_.get(), OnStop());
517 } 517 }
518 518
519 } // namespace content 519 } // namespace content
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