Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1424)

Side by Side Diff: webrtc/media/engine/fakewebrtccall.h

Issue 1728503002: Replace scoped_ptr with unique_ptr in webrtc/media/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@up1
Patch Set: Created 4 years, 10 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
« no previous file with comments | « webrtc/media/devices/mobiledevicemanager.cc ('k') | webrtc/media/engine/fakewebrtccall.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
(...skipping 78 matching lines...) Expand 10 before | Expand all | Expand 10 after
89 return true; 89 return true;
90 } 90 }
91 91
92 // webrtc::AudioReceiveStream implementation. 92 // webrtc::AudioReceiveStream implementation.
93 webrtc::AudioReceiveStream::Stats GetStats() const override; 93 webrtc::AudioReceiveStream::Stats GetStats() const override;
94 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override; 94 void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
95 95
96 webrtc::AudioReceiveStream::Config config_; 96 webrtc::AudioReceiveStream::Config config_;
97 webrtc::AudioReceiveStream::Stats stats_; 97 webrtc::AudioReceiveStream::Stats stats_;
98 int received_packets_; 98 int received_packets_;
99 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_; 99 std::unique_ptr<webrtc::AudioSinkInterface> sink_;
100 }; 100 };
101 101
102 class FakeVideoSendStream final : public webrtc::VideoSendStream, 102 class FakeVideoSendStream final : public webrtc::VideoSendStream,
103 public webrtc::VideoCaptureInput { 103 public webrtc::VideoCaptureInput {
104 public: 104 public:
105 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config, 105 FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
106 const webrtc::VideoEncoderConfig& encoder_config); 106 const webrtc::VideoEncoderConfig& encoder_config);
107 webrtc::VideoSendStream::Config GetConfig() const; 107 webrtc::VideoSendStream::Config GetConfig() const;
108 webrtc::VideoEncoderConfig GetEncoderConfig() const; 108 webrtc::VideoEncoderConfig GetEncoderConfig() const;
109 std::vector<webrtc::VideoStream> GetVideoStreams(); 109 std::vector<webrtc::VideoStream> GetVideoStreams();
(...skipping 134 matching lines...) Expand 10 before | Expand all | Expand 10 after
244 std::vector<FakeAudioSendStream*> audio_send_streams_; 244 std::vector<FakeAudioSendStream*> audio_send_streams_;
245 std::vector<FakeVideoReceiveStream*> video_receive_streams_; 245 std::vector<FakeVideoReceiveStream*> video_receive_streams_;
246 std::vector<FakeAudioReceiveStream*> audio_receive_streams_; 246 std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
247 247
248 int num_created_send_streams_; 248 int num_created_send_streams_;
249 int num_created_receive_streams_; 249 int num_created_receive_streams_;
250 }; 250 };
251 251
252 } // namespace cricket 252 } // namespace cricket
253 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_ 253 #endif // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_
OLDNEW
« no previous file with comments | « webrtc/media/devices/mobiledevicemanager.cc ('k') | webrtc/media/engine/fakewebrtccall.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698