Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(86)

Side by Side Diff: webrtc/video/vie_receiver.h

Issue 1735033003: Removes use of DeRegister Rtp Header Extension for video (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: 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
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 46 matching lines...) Expand 10 before | Expand all | Expand 10 after
57 57
58 uint32_t GetRemoteSsrc() const; 58 uint32_t GetRemoteSsrc() const;
59 int GetCsrcs(uint32_t* csrcs) const; 59 int GetCsrcs(uint32_t* csrcs) const;
60 60
61 void SetRtpRtcpModule(RtpRtcp* module); 61 void SetRtpRtcpModule(RtpRtcp* module);
62 62
63 RtpReceiver* GetRtpReceiver() const; 63 RtpReceiver* GetRtpReceiver() const;
64 64
65 void RegisterRtpRtcpModules(const std::vector<RtpRtcp*>& rtp_modules); 65 void RegisterRtpRtcpModules(const std::vector<RtpRtcp*>& rtp_modules);
66 66
67 bool SetReceiveTimestampOffsetStatus(bool enable, int id); 67 bool SetReceiveTimestampOffsetStatus(int id);
68 bool SetReceiveAbsoluteSendTimeStatus(bool enable, int id); 68 bool SetReceiveAbsoluteSendTimeStatus(int id);
69 bool SetReceiveVideoRotationStatus(bool enable, int id); 69 bool SetReceiveVideoRotationStatus(int id);
70 bool SetReceiveTransportSequenceNumber(bool enable, int id); 70 bool SetReceiveTransportSequenceNumber(int id);
71 71
72 void StartReceive(); 72 void StartReceive();
73 void StopReceive(); 73 void StopReceive();
74 74
75 bool DeliverRtp(const uint8_t* rtp_packet, 75 bool DeliverRtp(const uint8_t* rtp_packet,
76 size_t rtp_packet_length, 76 size_t rtp_packet_length,
77 const PacketTime& packet_time); 77 const PacketTime& packet_time);
78 bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length); 78 bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
79 79
80 // Implements RtpData. 80 // Implements RtpData.
(...skipping 38 matching lines...) Expand 10 before | Expand all | Expand 10 after
119 bool restored_packet_in_use_; 119 bool restored_packet_in_use_;
120 bool receiving_ast_enabled_; 120 bool receiving_ast_enabled_;
121 bool receiving_cvo_enabled_; 121 bool receiving_cvo_enabled_;
122 bool receiving_tsn_enabled_; 122 bool receiving_tsn_enabled_;
123 int64_t last_packet_log_ms_; 123 int64_t last_packet_log_ms_;
124 }; 124 };
125 125
126 } // namespace webrtc 126 } // namespace webrtc
127 127
128 #endif // WEBRTC_VIDEO_VIE_RECEIVER_H_ 128 #endif // WEBRTC_VIDEO_VIE_RECEIVER_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698