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

webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h

 /*
  *  Copyright (c) 2015 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 24 matching lines @@
   // Inserts a new packet with |rtp_header| and |payload| of
   // |payload_size_bytes| bytes. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   virtual void InsertPacket(WebRtcRTPHeader rtp_header, const uint8_t* payload,
                             size_t payload_size_bytes,
                             uint32_t receive_timestamp);
 
   // Get 10 ms of audio data. The data is written to |output|, which can hold
   // (at least) |max_length| elements. Returns number of samples.
-  int GetOutputAudio(size_t max_length, int16_t* output,
-                     NetEqOutputType* output_type);
+  size_t GetOutputAudio(size_t max_length, int16_t* output,
+                        NetEqOutputType* output_type);
 
   NetEq* neteq() { return neteq_.get(); }
 
  private:
   NetEqDecoder codec_;
   AudioDecoder* decoder_;
   int sample_rate_hz_;
   int channels_;
   rtc::scoped_ptr<NetEq> neteq_;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EXTERNAL_DECODER_TEST_H_