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

Issue 1335923002: Add RTC_ prefix to (D)CHECKs and related macros. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Rebase. Created 5 years, 3 months ago
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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
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468 468
469 // Store new. 469 // Store new.
470 packet_overhead_ = packet_overhead; 470 packet_overhead_ = packet_overhead;
471 471
472 uint16_t length = 472 uint16_t length =
473 rtp_sender_.MaxPayloadLength() - packet_over_head_diff; 473 rtp_sender_.MaxPayloadLength() - packet_over_head_diff;
474 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_); 474 return rtp_sender_.SetMaxPayloadLength(length, packet_overhead_);
475 } 475 }
476 476
477 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) { 477 int32_t ModuleRtpRtcpImpl::SetMaxTransferUnit(const uint16_t mtu) {
478 DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu; 478 RTC_DCHECK_LE(mtu, IP_PACKET_SIZE) << "Invalid mtu: " << mtu;
479 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_, 479 return rtp_sender_.SetMaxPayloadLength(mtu - packet_overhead_,
480 packet_overhead_); 480 packet_overhead_);
481 } 481 }
482 482
483 RTCPMethod ModuleRtpRtcpImpl::RTCP() const { 483 RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
484 if (rtcp_sender_.Status() != kRtcpOff) { 484 if (rtcp_sender_.Status() != kRtcpOff) {
485 return rtcp_receiver_.Status(); 485 return rtcp_receiver_.Status();
486 } 486 }
487 return kRtcpOff; 487 return kRtcpOff;
488 } 488 }
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993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback( 993 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
994 StreamDataCountersCallback* callback) { 994 StreamDataCountersCallback* callback) {
995 rtp_sender_.RegisterRtpStatisticsCallback(callback); 995 rtp_sender_.RegisterRtpStatisticsCallback(callback);
996 } 996 }
997 997
998 StreamDataCountersCallback* 998 StreamDataCountersCallback*
999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const { 999 ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
1000 return rtp_sender_.GetRtpStatisticsCallback(); 1000 return rtp_sender_.GetRtpStatisticsCallback();
1001 } 1001 }
1002 } // namespace webrtc 1002 } // namespace webrtc
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