[NOT FOR REVIEW] Convert channel counts to size_t.

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 106 matching lines @@
 
   WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
   WEBRTC_STUB(set_sample_rate_hz, (int rate));
   WEBRTC_STUB_CONST(input_sample_rate_hz, ());
   WEBRTC_STUB_CONST(sample_rate_hz, ());
   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, ());
+  size_t num_input_channels() const override { return 0; }
+  size_t num_output_channels() const override { return 0; }
+  size_t num_reverse_channels() const override { return 0; }
   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,
       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,
@@ skipping 1140 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_