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

Index: webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
index 52c34bb79c85b3ce4de30b6837f069c76198d4f7..49750c26c8702fcc881e24147f69135b2b6caf89 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
@@ -43,11 +43,11 @@ void NetEqExternalDecoderTest::InsertPacket(WebRtcRTPHeader rtp_header,
           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,
@@ -56,7 +56,8 @@ int NetEqExternalDecoderTest::GetOutputAudio(size_t max_length,
                              &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;
 }