Update a ton of audio code to use size_t more correctly and in general reduce

webrtc/modules/audio_coding/codecs/opus/interface/audio_encoder_opus.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 32 matching lines @@
     int complexity;
     bool dtx_enabled;
   };
 
   explicit AudioEncoderOpus(const Config& config);
   ~AudioEncoderOpus() override;
 
   int SampleRateHz() const override;
   int NumChannels() const override;
   size_t MaxEncodedBytes() 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;
 
   double packet_loss_rate() const { return packet_loss_rate_; }
   ApplicationMode application() const { return application_; }
   bool dtx_enabled() const { return dtx_enabled_; }
 
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
  private:
-  const int num_10ms_frames_per_packet_;
+  const size_t num_10ms_frames_per_packet_;
   const int num_channels_;
   const int payload_type_;
   const ApplicationMode application_;
   int bitrate_bps_;
   const bool dtx_enabled_;
-  const int samples_per_10ms_frame_;
+  const size_t samples_per_10ms_frame_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   double packet_loss_rate_;
 };
 
 struct CodecInst;
 
 class AudioEncoderMutableOpus
     : public AudioEncoderMutableImpl<AudioEncoderOpus> {
@@ skipping 17 matching lines @@
     return encoder()->packet_loss_rate();
   }
   bool dtx_enabled() const {
     CriticalSectionScoped cs(encoder_lock_.get());
     return encoder()->dtx_enabled();
   }
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_INTERFACE_AUDIO_ENCODER_OPUS_H_