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

webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.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.
  */
 #include <string>
@@ skipping 79 matching lines @@
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
   EXPECT_EQ(0, encoded_bytes);
   encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
+  EXPECT_GT(encoded_bytes, 0);
 
   // Call to update bandwidth estimator with real data.
   EXPECT_EQ(0, WebRtcIsac_UpdateBwEstimate(isac_codec_, bitstream_,
-                                           encoded_bytes, 1, 12345, 56789));
+                                           static_cast<size_t>(encoded_bytes),
+                                           1, 12345, 56789));
 
   // Free memory.
   EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
 }
 
 }  // namespace webrtc