Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(26)

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

Issue 1743543003: RtpRtcp allows to register header extension by name (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
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 | « no previous file | webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.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 (c) 2012 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2012 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_RTCP_H_ 11 #ifndef WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ 12 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
13 13
14 #include <set> 14 #include <set>
15 #include <string>
15 #include <utility> 16 #include <utility>
16 #include <vector> 17 #include <vector>
17 18
18 #include "webrtc/modules/include/module.h" 19 #include "webrtc/modules/include/module.h"
19 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
20 21
21 namespace webrtc { 22 namespace webrtc {
22 // Forward declarations. 23 // Forward declarations.
23 class ReceiveStatistics; 24 class ReceiveStatistics;
24 class RemoteBitrateEstimator; 25 class RemoteBitrateEstimator;
(...skipping 150 matching lines...) Expand 10 before | Expand all | Expand 10 after
175 virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; 176 virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
176 177
177 /* 178 /*
178 * (De)register RTP header extension type and id. 179 * (De)register RTP header extension type and id.
179 * 180 *
180 * return -1 on failure else 0 181 * return -1 on failure else 0
181 */ 182 */
182 virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, 183 virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
183 uint8_t id) = 0; 184 uint8_t id) = 0;
184 185
186 virtual bool RegisterRtpHeaderExtension(const std::string& type,
187 uint8_t id) = 0;
188
185 virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; 189 virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
186 190
187 /* 191 /*
188 * get start timestamp 192 * get start timestamp
189 */ 193 */
190 virtual uint32_t StartTimestamp() const = 0; 194 virtual uint32_t StartTimestamp() const = 0;
191 195
192 /* 196 /*
193 * configure start timestamp, default is a random number 197 * configure start timestamp, default is a random number
194 * 198 *
(...skipping 450 matching lines...) Expand 10 before | Expand all | Expand 10 after
645 649
646 /* 650 /*
647 * send a request for a keyframe 651 * send a request for a keyframe
648 * 652 *
649 * return -1 on failure else 0 653 * return -1 on failure else 0
650 */ 654 */
651 virtual int32_t RequestKeyFrame() = 0; 655 virtual int32_t RequestKeyFrame() = 0;
652 }; 656 };
653 } // namespace webrtc 657 } // namespace webrtc
654 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ 658 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
OLDNEW
« no previous file with comments | « no previous file | webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698