Misc. small cleanups

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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 119 matching lines @@
   return config_.bitrate_bps;
 }
 
 AudioEncoder::EncodedInfo AudioEncoderOpus::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   if (input_buffer_.empty())
     first_timestamp_in_buffer_ = rtp_timestamp;
-  RTC_DCHECK_EQ(static_cast<size_t>(SamplesPer10msFrame()), audio.size());
+  RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
-      (static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame())) {
+      (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
-  RTC_CHECK_EQ(
-      input_buffer_.size(),
-      static_cast<size_t>(Num10msFramesPerPacket()) * SamplesPer10msFrame());
+  RTC_CHECK_EQ(input_buffer_.size(),
+               Num10msFramesPerPacket() * SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(input_buffer_.size(),
                            static_cast<size_t>(config_.num_channels)),
       rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
   RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
   info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
@@ skipping 48 matching lines @@
   }
 }
 
 void AudioEncoderOpus::SetTargetBitrate(int bits_per_second) {
   config_.bitrate_bps =
       std::max(std::min(bits_per_second, kMaxBitrateBps), kMinBitrateBps);
   RTC_DCHECK(config_.IsOk());
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.bitrate_bps));
 }
 
-int AudioEncoderOpus::Num10msFramesPerPacket() const {
-  return rtc::CheckedDivExact(config_.frame_size_ms, 10);
+size_t AudioEncoderOpus::Num10msFramesPerPacket() const {
+  return static_cast<size_t>(rtc::CheckedDivExact(config_.frame_size_ms, 10));
 }
 
-int AudioEncoderOpus::SamplesPer10msFrame() const {
+size_t AudioEncoderOpus::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
 }
 
 // If the given config is OK, recreate the Opus encoder instance with those
 // settings, save the config, and return true. Otherwise, do nothing and return
 // false.
 bool AudioEncoderOpus::RecreateEncoderInstance(const Config& config) {
   if (!config.IsOk())
     return false;
   if (inst_)
@@ skipping 17 matching lines @@
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
   return true;
 }
 
 }  // namespace webrtc