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

webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.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 53 matching lines @@
 
   // Creates a Packet object from the last packet produced by ACM (and received
   // through the SendData method as a callback). Ownership of the new Packet
   // object is transferred to the caller.
   Packet* CreatePacket();
 
   SimulatedClock clock_;
   rtc::scoped_ptr<AudioCodingModule> acm_;
   InputAudioFile* audio_source_;
   int source_rate_hz_;
-  const int input_block_size_samples_;
+  const size_t input_block_size_samples_;
   AudioFrame input_frame_;
   bool codec_registered_;
   int test_duration_ms_;
   // The following member variables are set whenever SendData() is called.
   FrameType frame_type_;
   int payload_type_;
   uint32_t timestamp_;
   uint16_t sequence_number_;
   std::vector<uint8_t> last_payload_vec_;
   bool data_to_send_;
 
   DISALLOW_COPY_AND_ASSIGN(AcmSendTestOldApi);
 };
 
 }  // namespace test
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_SEND_TEST_H_