Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(163)

Side by Side Diff: webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h

Issue 1942823002: Remove webrtc/base/scoped_ptr.h (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fix additional scoped_ptr uses that had been added 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
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 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 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
13 13
14 #include <map> 14 #include <map>
15 #include <memory> 15 #include <memory>
16 16
17 #include "webrtc/base/criticalsection.h" 17 #include "webrtc/base/criticalsection.h"
18 #include "webrtc/base/scoped_ptr.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h" 18 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
20 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
21 20
22 namespace webrtc { 21 namespace webrtc {
23 22
24 // This strategy deals with the audio/video-specific aspects 23 // This strategy deals with the audio/video-specific aspects
25 // of payload handling. 24 // of payload handling.
26 class RTPPayloadStrategy { 25 class RTPPayloadStrategy {
27 public: 26 public:
28 virtual ~RTPPayloadStrategy() {} 27 virtual ~RTPPayloadStrategy() {}
(...skipping 149 matching lines...) Expand 10 before | Expand all | Expand 10 after
178 std::map<int, int> rtx_payload_type_map_; 177 std::map<int, int> rtx_payload_type_map_;
179 // When true, use rtx_payload_type_map_ when restoring RTX packets to get the 178 // When true, use rtx_payload_type_map_ when restoring RTX packets to get the
180 // correct payload type. 179 // correct payload type.
181 bool use_rtx_payload_mapping_on_restore_; 180 bool use_rtx_payload_mapping_on_restore_;
182 uint32_t ssrc_rtx_; 181 uint32_t ssrc_rtx_;
183 }; 182 };
184 183
185 } // namespace webrtc 184 } // namespace webrtc
186 185
187 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_ 186 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_PAYLOAD_REGISTRY_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h ('k') | webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698