Add RTC_ prefix to (D)CHECKs and related macros.

webrtc/modules/rtp_rtcp/source/rtp_sender.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 341 matching lines @@
   return payload_type_;
 }
 
 int RTPSender::SendPayloadFrequency() const {
   return audio_ != NULL ? audio_->AudioFrequency() : kVideoPayloadTypeFrequency;
 }
 
 int32_t RTPSender::SetMaxPayloadLength(size_t max_payload_length,
                                        uint16_t packet_over_head) {
   // Sanity check.
-  DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
+  RTC_DCHECK(max_payload_length >= 100 && max_payload_length <= IP_PACKET_SIZE)
       << "Invalid max payload length: " << max_payload_length;
   CriticalSectionScoped cs(send_critsect_.get());
   max_payload_length_ = max_payload_length;
   packet_over_head_ = packet_over_head;
   return 0;
 }
 
 size_t RTPSender::MaxDataPayloadLength() const {
   int rtx;
   {
@@ skipping 31 matching lines @@
 }
 
 uint32_t RTPSender::RtxSsrc() const {
   CriticalSectionScoped cs(send_critsect_.get());
   return ssrc_rtx_;
 }
 
 void RTPSender::SetRtxPayloadType(int payload_type,
                                   int associated_payload_type) {
   CriticalSectionScoped cs(send_critsect_.get());
-  DCHECK_LE(payload_type, 127);
-  DCHECK_LE(associated_payload_type, 127);
+  RTC_DCHECK_LE(payload_type, 127);
+  RTC_DCHECK_LE(associated_payload_type, 127);
   if (payload_type < 0) {
     LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
     return;
   }
 
   rtx_payload_type_map_[associated_payload_type] = payload_type;
   rtx_payload_type_ = payload_type;
 }
 
 std::pair<int, int> RTPSender::RtxPayloadType() const {
@@ skipping 1359 matching lines @@
 int32_t RTPSender::SendRTPIntraRequest() {
   if (audio_configured_) {
     return -1;
   }
   return video_->SendRTPIntraRequest();
 }
 
 void RTPSender::SetGenericFECStatus(bool enable,
                                     uint8_t payload_type_red,
                                     uint8_t payload_type_fec) {
-  DCHECK(!audio_configured_);
+  RTC_DCHECK(!audio_configured_);
   video_->SetGenericFECStatus(enable, payload_type_red, payload_type_fec);
 }
 
 void RTPSender::GenericFECStatus(bool* enable,
                                     uint8_t* payload_type_red,
                                     uint8_t* payload_type_fec) const {
-  DCHECK(!audio_configured_);
+  RTC_DCHECK(!audio_configured_);
   video_->GenericFECStatus(*enable, *payload_type_red, *payload_type_fec);
 }
 
 int32_t RTPSender::SetFecParameters(
     const FecProtectionParams *delta_params,
     const FecProtectionParams *key_params) {
   if (audio_configured_) {
     return -1;
   }
   video_->SetFecParameters(delta_params, key_params);
@@ skipping 89 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc