Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(99)

Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

Issue 1864313003: Move Ownership of RtpModules to VideoSendStream. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebased Created 4 years, 8 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 57 matching lines...) Expand 10 before | Expand all | Expand 10 after
68 uint32_t StartTimestamp() const override; 68 uint32_t StartTimestamp() const override;
69 69
70 // Configure start timestamp, default is a random number. 70 // Configure start timestamp, default is a random number.
71 void SetStartTimestamp(uint32_t timestamp) override; 71 void SetStartTimestamp(uint32_t timestamp) override;
72 72
73 uint16_t SequenceNumber() const override; 73 uint16_t SequenceNumber() const override;
74 74
75 // Set SequenceNumber, default is a random number. 75 // Set SequenceNumber, default is a random number.
76 void SetSequenceNumber(uint16_t seq) override; 76 void SetSequenceNumber(uint16_t seq) override;
77 77
78 bool SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) override; 78 void SetRtpState(const RtpState& rtp_state) override;
79 bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) override; 79 void SetRtxState(const RtpState& rtp_state) override;
80 RtpState GetRtpState() const override;
81 RtpState GetRtxState() const override;
80 82
81 uint32_t SSRC() const override; 83 uint32_t SSRC() const override;
82 84
83 // Configure SSRC, default is a random number. 85 // Configure SSRC, default is a random number.
84 void SetSSRC(uint32_t ssrc) override; 86 void SetSSRC(uint32_t ssrc) override;
85 87
86 void SetCsrcs(const std::vector<uint32_t>& csrcs) override; 88 void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
87 89
88 RTCPSender::FeedbackState GetFeedbackState(); 90 RTCPSender::FeedbackState GetFeedbackState();
89 91
(...skipping 272 matching lines...) Expand 10 before | Expand all | Expand 10 after
362 PacketLossStats receive_loss_stats_; 364 PacketLossStats receive_loss_stats_;
363 365
364 // The processed RTT from RtcpRttStats. 366 // The processed RTT from RtcpRttStats.
365 rtc::CriticalSection critical_section_rtt_; 367 rtc::CriticalSection critical_section_rtt_;
366 int64_t rtt_ms_; 368 int64_t rtt_ms_;
367 }; 369 };
368 370
369 } // namespace webrtc 371 } // namespace webrtc
370 372
371 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 373 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender.cc ('k') | webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698