Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(265)

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

Issue 503683003: Remove implicit conversions from scoped_refptr to T* in content/renderer/media/webrtc* (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 6 years, 3 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
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 "base/strings/utf_string_conversions.h" 6 #include "base/strings/utf_string_conversions.h"
7 #include "content/renderer/media/mock_media_constraint_factory.h" 7 #include "content/renderer/media/mock_media_constraint_factory.h"
8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" 8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h"
9 #include "content/renderer/media/webrtc_audio_capturer.h" 9 #include "content/renderer/media/webrtc_audio_capturer.h"
10 #include "content/renderer/media/webrtc_local_audio_source_provider.h" 10 #include "content/renderer/media/webrtc_local_audio_source_provider.h"
(...skipping 19 matching lines...) Expand all
30 source_data_.reset(new int16[length]); 30 source_data_.reset(new int16[length]);
31 sink_bus_ = media::AudioBus::Create(sink_params_); 31 sink_bus_ = media::AudioBus::Create(sink_params_);
32 MockMediaConstraintFactory constraint_factory; 32 MockMediaConstraintFactory constraint_factory;
33 scoped_refptr<WebRtcAudioCapturer> capturer( 33 scoped_refptr<WebRtcAudioCapturer> capturer(
34 WebRtcAudioCapturer::CreateCapturer( 34 WebRtcAudioCapturer::CreateCapturer(
35 -1, StreamDeviceInfo(), 35 -1, StreamDeviceInfo(),
36 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); 36 constraint_factory.CreateWebMediaConstraints(), NULL, NULL));
37 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( 37 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter(
38 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); 38 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL));
39 scoped_ptr<WebRtcLocalAudioTrack> native_track( 39 scoped_ptr<WebRtcLocalAudioTrack> native_track(
40 new WebRtcLocalAudioTrack(adapter, capturer, NULL)); 40 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL));
41 blink::WebMediaStreamSource audio_source; 41 blink::WebMediaStreamSource audio_source;
42 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), 42 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"),
43 blink::WebMediaStreamSource::TypeAudio, 43 blink::WebMediaStreamSource::TypeAudio,
44 base::UTF8ToUTF16("dummy_source_name")); 44 base::UTF8ToUTF16("dummy_source_name"));
45 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), 45 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"),
46 audio_source); 46 audio_source);
47 blink_track_.setExtraData(native_track.release()); 47 blink_track_.setExtraData(native_track.release());
48 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_)); 48 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_));
49 source_provider_->SetSinkParamsForTesting(sink_params_); 49 source_provider_->SetSinkParamsForTesting(sink_params_);
50 source_provider_->OnSetFormat(source_params_); 50 source_provider_->OnSetFormat(source_params_);
(...skipping 78 matching lines...) Expand 10 before | Expand all | Expand 10 after
129 // Stop the audio track. 129 // Stop the audio track.
130 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>( 130 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>(
131 MediaStreamTrack::GetTrack(blink_track_)); 131 MediaStreamTrack::GetTrack(blink_track_));
132 native_track->Stop(); 132 native_track->Stop();
133 133
134 // Delete the source provider. 134 // Delete the source provider.
135 source_provider_.reset(); 135 source_provider_.reset();
136 } 136 }
137 137
138 } // namespace content 138 } // namespace content
OLDNEW
« no previous file with comments | « content/renderer/media/webrtc_local_audio_renderer.cc ('k') | content/renderer/media/webrtc_local_audio_track.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698