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.cc

 /*
  *  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 25 matching lines @@
 void NetEqExternalDecoderTest::InsertPacket(WebRtcRTPHeader rtp_header,
                                             const uint8_t* payload,
                                             size_t payload_size_bytes,
                                             uint32_t receive_timestamp) {
   ASSERT_EQ(
       NetEq::kOK,
       neteq_->InsertPacket(
           rtp_header, payload, payload_size_bytes, receive_timestamp));
 }
 
-int NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
-                                             int16_t* output,
-                                             NetEqOutputType* output_type) {
+size_t NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
+                                                int16_t* output,
+                                                NetEqOutputType* output_type) {
   // Get audio from regular instance.
-  int samples_per_channel;
+  size_t samples_per_channel;
   int num_channels;
   EXPECT_EQ(NetEq::kOK,
             neteq_->GetAudio(max_length,
                              output,
                              &samples_per_channel,
                              &num_channels,
                              output_type));
   EXPECT_EQ(channels_, num_channels);
-  EXPECT_EQ(kOutputLengthMs * sample_rate_hz_ / 1000, samples_per_channel);
+  EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * sample_rate_hz_ / 1000),
+            samples_per_channel);
   return samples_per_channel;
 }
 
 }  // namespace test
 }  // namespace webrtc