Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h

 /*
  *  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 22 matching lines @@
 
   // Caller keeps ownership of the AudioEncoder object.
   explicit AudioEncoderCopyRed(const Config& config);
 
   ~AudioEncoderCopyRed() override;
 
   int SampleRateHz() const override;
   int NumChannels() const override;
   size_t MaxEncodedBytes() const override;
   int RtpTimestampRateHz() const override;
-  int Num10MsFramesInNextPacket() const override;
-  int Max10MsFramesInAPacket() const override;
+  size_t Num10MsFramesInNextPacket() const override;
+  size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
   void SetTargetBitrate(int bits_per_second) override;
   void SetProjectedPacketLossRate(double fraction) override;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
  private:
   AudioEncoder* speech_encoder_;
   int red_payload_type_;
   rtc::Buffer secondary_encoded_;
   EncodedInfoLeaf secondary_info_;
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_RED_AUDIO_ENCODER_COPY_RED_H_