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Side by Side Diff: webrtc/api/rtpsender.h

Issue 1930463002: Replace scoped_ptr with unique_ptr in webrtc/api/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 months ago
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1 /* 1 /*
2 * Copyright 2015 The WebRTC project authors. All Rights Reserved. 2 * Copyright 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
11 // This file contains classes that implement RtpSenderInterface. 11 // This file contains classes that implement RtpSenderInterface.
12 // An RtpSender associates a MediaStreamTrackInterface with an underlying 12 // An RtpSender associates a MediaStreamTrackInterface with an underlying
13 // transport (provided by AudioProviderInterface/VideoProviderInterface) 13 // transport (provided by AudioProviderInterface/VideoProviderInterface)
14 14
15 #ifndef WEBRTC_API_RTPSENDER_H_ 15 #ifndef WEBRTC_API_RTPSENDER_H_
16 #define WEBRTC_API_RTPSENDER_H_ 16 #define WEBRTC_API_RTPSENDER_H_
17 17
18 #include <memory>
18 #include <string> 19 #include <string>
19 20
20 #include "webrtc/api/mediastreamprovider.h" 21 #include "webrtc/api/mediastreamprovider.h"
21 #include "webrtc/api/rtpsenderinterface.h" 22 #include "webrtc/api/rtpsenderinterface.h"
22 #include "webrtc/api/statscollector.h" 23 #include "webrtc/api/statscollector.h"
23 #include "webrtc/base/basictypes.h" 24 #include "webrtc/base/basictypes.h"
24 #include "webrtc/base/criticalsection.h" 25 #include "webrtc/base/criticalsection.h"
25 #include "webrtc/base/scoped_ptr.h" 26 #include "webrtc/base/scoped_ptr.h"
26 #include "webrtc/media/base/audiosource.h" 27 #include "webrtc/media/base/audiosource.h"
27 28
(...skipping 84 matching lines...) Expand 10 before | Expand all | Expand 10 after
112 std::string stream_id_; 113 std::string stream_id_;
113 AudioProviderInterface* provider_; 114 AudioProviderInterface* provider_;
114 StatsCollector* stats_; 115 StatsCollector* stats_;
115 rtc::scoped_refptr<AudioTrackInterface> track_; 116 rtc::scoped_refptr<AudioTrackInterface> track_;
116 uint32_t ssrc_ = 0; 117 uint32_t ssrc_ = 0;
117 bool cached_track_enabled_ = false; 118 bool cached_track_enabled_ = false;
118 bool stopped_ = false; 119 bool stopped_ = false;
119 120
120 // Used to pass the data callback from the |track_| to the other end of 121 // Used to pass the data callback from the |track_| to the other end of
121 // cricket::AudioSource. 122 // cricket::AudioSource.
122 rtc::scoped_ptr<LocalAudioSinkAdapter> sink_adapter_; 123 std::unique_ptr<LocalAudioSinkAdapter> sink_adapter_;
123 }; 124 };
124 125
125 class VideoRtpSender : public ObserverInterface, 126 class VideoRtpSender : public ObserverInterface,
126 public rtc::RefCountedObject<RtpSenderInterface> { 127 public rtc::RefCountedObject<RtpSenderInterface> {
127 public: 128 public:
128 VideoRtpSender(VideoTrackInterface* track, 129 VideoRtpSender(VideoTrackInterface* track,
129 const std::string& stream_id, 130 const std::string& stream_id,
130 VideoProviderInterface* provider); 131 VideoProviderInterface* provider);
131 132
132 // Randomly generates stream_id. 133 // Randomly generates stream_id.
(...skipping 44 matching lines...) Expand 10 before | Expand all | Expand 10 after
177 VideoProviderInterface* provider_; 178 VideoProviderInterface* provider_;
178 rtc::scoped_refptr<VideoTrackInterface> track_; 179 rtc::scoped_refptr<VideoTrackInterface> track_;
179 uint32_t ssrc_ = 0; 180 uint32_t ssrc_ = 0;
180 bool cached_track_enabled_ = false; 181 bool cached_track_enabled_ = false;
181 bool stopped_ = false; 182 bool stopped_ = false;
182 }; 183 };
183 184
184 } // namespace webrtc 185 } // namespace webrtc
185 186
186 #endif // WEBRTC_API_RTPSENDER_H_ 187 #endif // WEBRTC_API_RTPSENDER_H_
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