Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(774)

Side by Side Diff: webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

Issue 2628563003: Propagate packet pacing information to SenTimeHistory (Closed)
Patch Set: Rebase Created 3 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
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 96 matching lines...) Expand 10 before | Expand all | Expand 10 after
107 const uint8_t* payload_data, 107 const uint8_t* payload_data,
108 size_t payload_size, 108 size_t payload_size,
109 const RTPFragmentationHeader* fragmentation, 109 const RTPFragmentationHeader* fragmentation,
110 const RTPVideoHeader* rtp_video_header, 110 const RTPVideoHeader* rtp_video_header,
111 uint32_t* frame_id_out)); 111 uint32_t* frame_id_out));
112 MOCK_METHOD5(TimeToSendPacket, 112 MOCK_METHOD5(TimeToSendPacket,
113 bool(uint32_t ssrc, 113 bool(uint32_t ssrc,
114 uint16_t sequence_number, 114 uint16_t sequence_number,
115 int64_t capture_time_ms, 115 int64_t capture_time_ms,
116 bool retransmission, 116 bool retransmission,
117 int probe_cluster_id)); 117 const PacedPacketInfo& pacing_info));
118 MOCK_METHOD2(TimeToSendPadding, size_t(size_t bytes, int probe_cluster_id)); 118 MOCK_METHOD2(TimeToSendPadding,
119 size_t(size_t bytes, const PacedPacketInfo& pacing_info));
119 MOCK_METHOD2(RegisterRtcpObservers, 120 MOCK_METHOD2(RegisterRtcpObservers,
120 void(RtcpIntraFrameObserver* intra_frame_callback, 121 void(RtcpIntraFrameObserver* intra_frame_callback,
121 RtcpBandwidthObserver* bandwidth_callback)); 122 RtcpBandwidthObserver* bandwidth_callback));
122 MOCK_CONST_METHOD0(RTCP, RtcpMode()); 123 MOCK_CONST_METHOD0(RTCP, RtcpMode());
123 MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method)); 124 MOCK_METHOD1(SetRTCPStatus, void(RtcpMode method));
124 MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE])); 125 MOCK_METHOD1(SetCNAME, int32_t(const char cname[RTCP_CNAME_SIZE]));
125 MOCK_CONST_METHOD2(RemoteCNAME, 126 MOCK_CONST_METHOD2(RemoteCNAME,
126 int32_t(uint32_t remote_ssrc, 127 int32_t(uint32_t remote_ssrc,
127 char cname[RTCP_CNAME_SIZE])); 128 char cname[RTCP_CNAME_SIZE]));
128 MOCK_CONST_METHOD5(RemoteNTP, 129 MOCK_CONST_METHOD5(RemoteNTP,
(...skipping 75 matching lines...) Expand 10 before | Expand all | Expand 10 after
204 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback, 205 MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
205 StreamDataCountersCallback*(void)); 206 StreamDataCountersCallback*(void));
206 MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&)); 207 MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&));
207 // Members. 208 // Members.
208 unsigned int remote_ssrc_; 209 unsigned int remote_ssrc_;
209 }; 210 };
210 211
211 } // namespace webrtc 212 } // namespace webrtc
212 213
213 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_ 214 #endif // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698