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.cc

 /*
  *  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 11 matching lines @@
 
 namespace webrtc {
 namespace test {
 
 AcmSendTest::AcmSendTest(InputAudioFile* audio_source,
                          int source_rate_hz,
                          int test_duration_ms)
     : clock_(0),
       audio_source_(audio_source),
       source_rate_hz_(source_rate_hz),
-      input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),
+      input_block_size_samples_(
+          static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),
       codec_registered_(false),
       test_duration_ms_(test_duration_ms),
       frame_type_(kAudioFrameSpeech),
       payload_type_(0),
       timestamp_(0),
       sequence_number_(0) {
   webrtc::AudioCoding::Config config;
   config.clock = &clock_;
   config.transport = this;
   acm_.reset(webrtc::AudioCoding::Create(config));
@@ skipping 89 matching lines @@
          last_payload_vec_.size());
   Packet* packet =
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds());
   assert(packet);
   assert(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc