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

webrtc/modules/audio_coding/main/test/opus_test.cc

 /*
  *  Copyright (c) 2013 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 252 matching lines @@
                                         &audio[written_samples]));
     written_samples += 480 * channels;
 
     // Sometimes we need to loop over the audio vector to produce the right
     // number of packets.
     int loop_encode = (written_samples - read_samples) /
         (channels * frame_length);
 
     if (loop_encode > 0) {
       const int kMaxBytes = 1000;  // Maximum number of bytes for one packet.
-      int16_t bitstream_len_byte;
+      size_t bitstream_len_byte;
       uint8_t bitstream[kMaxBytes];
       for (int i = 0; i < loop_encode; i++) {
         int bitstream_len_byte_int = WebRtcOpus_Encode(
             (channels == 1) ? opus_mono_encoder_ : opus_stereo_encoder_,
             &audio[read_samples], frame_length, kMaxBytes, bitstream);
         ASSERT_GE(bitstream_len_byte_int, 0);
-        bitstream_len_byte = static_cast<int16_t>(bitstream_len_byte_int);
+        bitstream_len_byte = static_cast<size_t>(bitstream_len_byte_int);
 
         // Simulate packet loss by setting |packet_loss_| to "true" in
         // |percent_loss| percent of the loops.
         // TODO(tlegrand): Move handling of loss simulation to TestPackStereo.
         if (percent_loss > 0) {
           if (counter_ == floor((100 / percent_loss) + 0.5)) {
             counter_ = 0;
             lost_packet = true;
             channel->set_lost_packet(true);
           } else {
@@ skipping 43 matching lines @@
 
     // Run received side of ACM.
     ASSERT_EQ(0, acm_receiver_->PlayoutData10Ms(out_freq_hz_b, &audio_frame));
 
     // Write output speech to file.
     out_file_.Write10MsData(
         audio_frame.data_,
         audio_frame.samples_per_channel_ * audio_frame.num_channels_);
 
     // Write stand-alone speech to file.
-    out_file_standalone_.Write10MsData(out_audio, decoded_samples * channels);
+    out_file_standalone_.Write10MsData(
+        out_audio, static_cast<size_t>(decoded_samples) * channels);
 
     if (audio_frame.timestamp_ > start_time_stamp) {
       // Number of channels should be the same for both stand-alone and
       // ACM-decoding.
       EXPECT_EQ(audio_frame.num_channels_, channels);
     }
 
     decoded_samples = 0;
   }
 
@@ skipping 16 matching lines @@
   out_file_.Open(file_name, 48000, "wb");
   file_stream.str("");
   file_name = file_stream.str();
   file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
       << test_number << ".pcm";
   file_name = file_stream.str();
   out_file_standalone_.Open(file_name, 48000, "wb");
 }
 
 }  // namespace webrtc