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

webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace webrtc {
 
 class MockAudioEncoder : public AudioEncoder {
  public:
   virtual ~MockAudioEncoder() { Die(); }
   MOCK_METHOD0(Die, void());
   MOCK_CONST_METHOD0(SampleRateHz, int());
   MOCK_CONST_METHOD0(NumChannels, int());
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
-  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, int());
-  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, int());
+  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
+  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
   MOCK_CONST_METHOD0(GetTargetBitrate, int());
   MOCK_METHOD1(SetTargetBitrate, void(int));
   MOCK_METHOD1(SetProjectedPacketLossRate, void(double));
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD4(EncodeInternal,
                EncodedInfo(uint32_t timestamp,
                            const int16_t* audio,
                            size_t max_encoded_bytes,
                            uint8_t* encoded));
 };
 
 class MockAudioEncoderMutable : public AudioEncoderMutable {
  public:
   MOCK_CONST_METHOD0(SampleRateHz, int());
   MOCK_CONST_METHOD0(NumChannels, int());
   MOCK_CONST_METHOD0(MaxEncodedBytes, size_t());
-  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, int());
-  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, int());
+  MOCK_CONST_METHOD0(Num10MsFramesInNextPacket, size_t());
+  MOCK_CONST_METHOD0(Max10MsFramesInAPacket, size_t());
   MOCK_CONST_METHOD0(GetTargetBitrate, int());
   MOCK_METHOD1(SetTargetBitrate, void(int));
   MOCK_METHOD1(SetProjectedPacketLossRate, void(double));
   // Note, we explicitly chose not to create a mock for the Encode method.
   MOCK_METHOD4(EncodeInternal,
                EncodedInfo(uint32_t timestamp,
                            const int16_t* audio,
                            size_t max_encoded_bytes,
                            uint8_t* encoded));
 
   MOCK_METHOD0(Reset, void());
   MOCK_METHOD1(SetFec, bool(bool enable));
   MOCK_METHOD1(SetDtx, bool(bool enable));
   MOCK_METHOD1(SetApplication, bool(Application application));
   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
   MOCK_METHOD1(SetMaxRate, void(int max_rate_bps));
   MOCK_METHOD1(SetMaxPlaybackRate, bool(int frequency_hz));
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_MOCK_MOCK_AUDIO_ENCODER_H_