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 59 matching lines @@
   WEBRTC_STUB(Initialize, (
       const webrtc::ProcessingConfig& processing_config));
 
   WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) {
     experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled;
   }
 
   WEBRTC_STUB_CONST(input_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; }

[perkj_webrtc, 2016/01/07 09:32:32]

I am not sure I understand the purpose of WEBRTC_STUB_CONST.. Should there be a
macro for size_t as well to keep the same pattern?
solenberg - I guess that is your decision if you know own the voice engine
implementation.

[Peter Kasting, 2016/01/07 09:47:50]

AFAICT it's just to reduce verbosity.  I don't see much benefit to adding
another macro form for "const function which returns size_t" when that would
literally just be used here.

Frankly, given the style guide's direction to avoid macros when possible, I
think all the macros here should be removed and replaced with normal function
declarations like we use almost everywhere else, but I'm not going to try to
write that change and push it through.

[the sun, 2016/01/07 12:27:59]

Don't worry; this is going away. Don't create any new macros.

+  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_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,
       float* const* dest));
@@ skipping 721 matching lines @@
   int playout_fail_channel_;
   int send_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_