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

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 114 matching lines @@
   WEBRTC_STUB_CONST(proc_sample_rate_hz, ());
   WEBRTC_STUB_CONST(proc_split_sample_rate_hz, ());
   WEBRTC_STUB_CONST(num_input_channels, ());
   WEBRTC_STUB_CONST(num_output_channels, ());
   WEBRTC_STUB_CONST(num_reverse_channels, ());
   WEBRTC_VOID_STUB(set_output_will_be_muted, (bool muted));
   WEBRTC_BOOL_STUB_CONST(output_will_be_muted, ());
   WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessStream, (
       const float* const* src,
-      int samples_per_channel,
+      size_t samples_per_channel,
       int input_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout input_layout,
       int output_sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout output_layout,
       float* const* dest));
   WEBRTC_STUB(ProcessStream,
               (const float* const* src,
                const webrtc::StreamConfig& input_config,
                const webrtc::StreamConfig& output_config,
                float* const* dest));
   WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame));
   WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));
   WEBRTC_STUB(AnalyzeReverseStream, (
       const float* const* data,
-      int samples_per_channel,
+      size_t samples_per_channel,
       int sample_rate_hz,
       webrtc::AudioProcessing::ChannelLayout layout));
   WEBRTC_STUB(ProcessReverseStream,
               (const float* const* src,
                const webrtc::StreamConfig& reverse_input_config,
                const webrtc::StreamConfig& reverse_output_config,
                float* const* dest));
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
@@ skipping 1122 matching lines @@
   DtmfInfo dtmf_info_;
   webrtc::VoEMediaProcess* media_processor_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_