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

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

 /*
  *  Copyright (c) 2014 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 35 matching lines @@
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       encoder_(nullptr) {
   Reset();
 }
 
 AudioEncoderIlbc::AudioEncoderIlbc(const CodecInst& codec_inst)
     : AudioEncoderIlbc(CreateConfig(codec_inst)) {}
 
 AudioEncoderIlbc::~AudioEncoderIlbc() {
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
 size_t AudioEncoderIlbc::MaxEncodedBytes() const {
   return RequiredOutputSizeBytes();
 }
 
 int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
 
@@ skipping 20 matching lines @@
     default:
       FATAL();
   }
 }
 
 AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
-  DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
+  RTC_DCHECK_GE(max_encoded_bytes, RequiredOutputSizeBytes());
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
   std::memcpy(input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_,
               audio,
               kSampleRateHz / 100 * sizeof(audio[0]));
 
   // If we don't yet have enough buffered input for a whole packet, we're done
   // for now.
   if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
     return EncodedInfo();
   }
 
   // Encode buffered input.
-  DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
+  RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
   const int output_len = WebRtcIlbcfix_Encode(
       encoder_,
       input_buffer_,
       kSampleRateHz / 100 * num_10ms_frames_per_packet_,
       encoded);
-  CHECK_GE(output_len, 0);
+  RTC_CHECK_GE(output_len, 0);
   EncodedInfo info;
   info.encoded_bytes = static_cast<size_t>(output_len);
-  DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
+  RTC_DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = config_.payload_type;
   return info;
 }
 
 void AudioEncoderIlbc::Reset() {
   if (encoder_)
-    CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
-  CHECK(config_.IsOk());
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
+    RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
+  RTC_CHECK(config_.IsOk());
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
   const int encoder_frame_size_ms = config_.frame_size_ms > 30
                                         ? config_.frame_size_ms / 2
                                         : config_.frame_size_ms;
-  CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
+  RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
   num_10ms_frames_buffered_ = 0;
 }
 
 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc