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

webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_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 47 matching lines @@
   ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
             loop_length_samples_);
   fclose(fp);
 
   // Add an extra block length of samples to the end of the array, starting
   // over again from the beginning of the array. This is done to simplify
   // the reading process when reading over the end of the loop.
   memcpy(&in_data_[loop_length_samples_], &in_data_[0],
          input_length_sample_ * channels_ * sizeof(int16_t));
 
-  max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+  max_bytes_ =
+      static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
   out_data_.reset(new int16_t[output_length_sample_ * channels_]);
   bit_stream_.reset(new uint8_t[max_bytes_]);
 
   if (save_out_data_) {
     std::string out_filename =
         ::testing::UnitTest::GetInstance()->current_test_info()->name();
 
     // Erase '/'
     size_t found;
     while ((found = out_filename.find('/')) != std::string::npos)
@@ skipping 36 matching lines @@
         loop_length_samples_;
     time_now_ms += block_duration_ms_;
   }
 
   printf("Encoding: %.2f%% real time,\nDecoding: %.2f%% real time.\n",
          (encoding_time_ms_ / audio_duration_sec) / 10.0,
          (decoding_time_ms_ / audio_duration_sec) / 10.0);
 }
 
 }  // namespace webrtc