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

webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.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 21 matching lines @@
    protected:
     explicit Config(int pt)
         : frame_size_ms(20), num_channels(1), payload_type(pt) {}
   };
 
   ~AudioEncoderPcm() 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;
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              const int16_t* audio,
                              size_t max_encoded_bytes,
                              uint8_t* encoded) override;
 
  protected:
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
 
-  virtual int16_t EncodeCall(const int16_t* audio,
-                             size_t input_len,
-                             uint8_t* encoded) = 0;
+  virtual size_t EncodeCall(const int16_t* audio,
+                            size_t input_len,
+                            uint8_t* encoded) = 0;
 
   virtual int BytesPerSample() const = 0;
 
  private:
   const int sample_rate_hz_;
   const int num_channels_;
   const int payload_type_;
-  const int num_10ms_frames_per_packet_;
+  const size_t num_10ms_frames_per_packet_;
   const size_t full_frame_samples_;
   std::vector<int16_t> speech_buffer_;
   uint32_t first_timestamp_in_buffer_;
 };
 
 class AudioEncoderPcmA final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(8) {}
   };
 
   explicit AudioEncoderPcmA(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
 
  protected:
-  int16_t EncodeCall(const int16_t* audio,
-                     size_t input_len,
-                     uint8_t* encoded) override;
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
 
   int BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
 };
 
 class AudioEncoderPcmU final : public AudioEncoderPcm {
  public:
   struct Config : public AudioEncoderPcm::Config {
     Config() : AudioEncoderPcm::Config(0) {}
   };
 
   explicit AudioEncoderPcmU(const Config& config)
       : AudioEncoderPcm(config, kSampleRateHz) {}
 
  protected:
-  int16_t EncodeCall(const int16_t* audio,
-                     size_t input_len,
-                     uint8_t* encoded) override;
+  size_t EncodeCall(const int16_t* audio,
+                    size_t input_len,
+                    uint8_t* encoded) override;
 
   int BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
 };
 
 struct CodecInst;
 
 class AudioEncoderMutablePcmU
     : public AudioEncoderMutableImpl<AudioEncoderPcmU> {
  public:
   explicit AudioEncoderMutablePcmU(const CodecInst& codec_inst);
 };
 
 class AudioEncoderMutablePcmA
     : public AudioEncoderMutableImpl<AudioEncoderPcmA> {
  public:
   explicit AudioEncoderMutablePcmA(const CodecInst& codec_inst);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_INCLUDE_AUDIO_ENCODER_PCM_H_