RTC_[D]CHECK_op: Remove superfluous casts

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.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 80 matching lines @@
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   RTC_CHECK_EQ(speech_buffer_.size(),
                rtp_timestamps_.size() * samples_per_10ms_frame);
   rtp_timestamps_.push_back(rtp_timestamp);
   RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
   speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
   if (rtp_timestamps_.size() < frames_to_encode) {
     return EncodedInfo();
   }
-  RTC_CHECK_LE(static_cast<int>(frames_to_encode * 10), kMaxFrameSizeMs)
+  RTC_CHECK_LE(frames_to_encode * 10, kMaxFrameSizeMs)
       << "Frame size cannot be larger than " << kMaxFrameSizeMs
       << " ms when using VAD/CNG.";
 
   // Group several 10 ms blocks per VAD call. Call VAD once or twice using the
   // following split sizes:
   // 10 ms = 10 + 0 ms; 20 ms = 20 + 0 ms; 30 ms = 30 + 0 ms;
   // 40 ms = 20 + 20 ms; 50 ms = 30 + 20 ms; 60 ms = 30 + 30 ms.
   size_t blocks_in_first_vad_call =
       (frames_to_encode > 3 ? 3 : frames_to_encode);
   if (frames_to_encode == 4)
@@ skipping 135 matching lines @@
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc