Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(430)

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

Issue 1373903003: Unify newapi::RtcpMode and RTCPMethod. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: ehm, compile the code Created 5 years, 2 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 108 matching lines...) Expand 10 before | Expand all | Expand 10 after
119 int64_t capture_time_ms, 119 int64_t capture_time_ms,
120 bool retransmission) override; 120 bool retransmission) override;
121 121
122 // Returns the number of padding bytes actually sent, which can be more or 122 // Returns the number of padding bytes actually sent, which can be more or
123 // less than |bytes|. 123 // less than |bytes|.
124 size_t TimeToSendPadding(size_t bytes) override; 124 size_t TimeToSendPadding(size_t bytes) override;
125 125
126 // RTCP part. 126 // RTCP part.
127 127
128 // Get RTCP status. 128 // Get RTCP status.
129 RTCPMethod RTCP() const override; 129 RtcpMode RTCP() const override;
130 130
131 // Configure RTCP status i.e on/off. 131 // Configure RTCP status i.e on/off.
132 void SetRTCPStatus(RTCPMethod method) override; 132 void SetRTCPStatus(RtcpMode method) override;
133 133
134 // Set RTCP CName. 134 // Set RTCP CName.
135 int32_t SetCNAME(const char* c_name) override; 135 int32_t SetCNAME(const char* c_name) override;
136 136
137 // Get remote CName. 137 // Get remote CName.
138 int32_t RemoteCNAME(uint32_t remote_ssrc, 138 int32_t RemoteCNAME(uint32_t remote_ssrc,
139 char c_name[RTCP_CNAME_SIZE]) const override; 139 char c_name[RTCP_CNAME_SIZE]) const override;
140 140
141 // Get remote NTP. 141 // Get remote NTP.
142 int32_t RemoteNTP(uint32_t* received_ntp_secs, 142 int32_t RemoteNTP(uint32_t* received_ntp_secs,
(...skipping 236 matching lines...) Expand 10 before | Expand all | Expand 10 after
379 PacketLossStats receive_loss_stats_; 379 PacketLossStats receive_loss_stats_;
380 380
381 // The processed RTT from RtcpRttStats. 381 // The processed RTT from RtcpRttStats.
382 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_; 382 rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtt_;
383 int64_t rtt_ms_; 383 int64_t rtt_ms_;
384 }; 384 };
385 385
386 } // namespace webrtc 386 } // namespace webrtc
387 387
388 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ 388 #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
OLDNEW
« no previous file with comments | « webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.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