Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(306)

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

Issue 1965063003: Revert of Remove webrtc/base/scoped_ptr.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 7 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/api/peerconnectioninterface_unittest.cc ('k') | webrtc/api/statstypes.h » ('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 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 <memory>
19 #include <string> 19 #include <string>
20 20
21 #include "webrtc/api/mediastreamprovider.h" 21 #include "webrtc/api/mediastreamprovider.h"
22 #include "webrtc/api/rtpsenderinterface.h" 22 #include "webrtc/api/rtpsenderinterface.h"
23 #include "webrtc/api/statscollector.h" 23 #include "webrtc/api/statscollector.h"
24 #include "webrtc/base/basictypes.h" 24 #include "webrtc/base/basictypes.h"
25 #include "webrtc/base/criticalsection.h" 25 #include "webrtc/base/criticalsection.h"
26 #include "webrtc/base/scoped_ptr.h"
26 #include "webrtc/media/base/audiosource.h" 27 #include "webrtc/media/base/audiosource.h"
27 28
28 namespace webrtc { 29 namespace webrtc {
29 30
30 // LocalAudioSinkAdapter receives data callback as a sink to the local 31 // LocalAudioSinkAdapter receives data callback as a sink to the local
31 // AudioTrack, and passes the data to the sink of AudioSource. 32 // AudioTrack, and passes the data to the sink of AudioSource.
32 class LocalAudioSinkAdapter : public AudioTrackSinkInterface, 33 class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
33 public cricket::AudioSource { 34 public cricket::AudioSource {
34 public: 35 public:
35 LocalAudioSinkAdapter(); 36 LocalAudioSinkAdapter();
(...skipping 141 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_
OLDNEW
« no previous file with comments | « webrtc/api/peerconnectioninterface_unittest.cc ('k') | webrtc/api/statstypes.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698