Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(534)

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

Issue 1725363003: Move RTP module activation into PayloadRouter. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: nuke comment Created 4 years, 9 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/common_types.h ('k') | 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
(...skipping 147 matching lines...) Expand 10 before | Expand all | Expand 10 after
158 const CodecInst& voiceCodec) = 0; 158 const CodecInst& voiceCodec) = 0;
159 159
160 /* 160 /*
161 * set codec name and payload type 161 * set codec name and payload type
162 * 162 *
163 * return -1 on failure else 0 163 * return -1 on failure else 0
164 */ 164 */
165 virtual int32_t RegisterSendPayload( 165 virtual int32_t RegisterSendPayload(
166 const VideoCodec& videoCodec) = 0; 166 const VideoCodec& videoCodec) = 0;
167 167
168 virtual void RegisterVideoSendPayload(int payload_type,
169 const char* payload_name) = 0;
170
168 /* 171 /*
169 * Unregister a send payload 172 * Unregister a send payload
170 * 173 *
171 * payloadType - payload type of codec 174 * payloadType - payload type of codec
172 * 175 *
173 * return -1 on failure else 0 176 * return -1 on failure else 0
174 */ 177 */
175 virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0; 178 virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
176 179
177 /* 180 /*
(...skipping 467 matching lines...) Expand 10 before | Expand all | Expand 10 after
645 648
646 /* 649 /*
647 * send a request for a keyframe 650 * send a request for a keyframe
648 * 651 *
649 * return -1 on failure else 0 652 * return -1 on failure else 0
650 */ 653 */
651 virtual int32_t RequestKeyFrame() = 0; 654 virtual int32_t RequestKeyFrame() = 0;
652 }; 655 };
653 } // namespace webrtc 656 } // namespace webrtc
654 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ 657 #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
OLDNEW
« no previous file with comments | « webrtc/common_types.h ('k') | webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698