[NOT FOR REVIEW] Convert channel counts to size_t.

talk/media/webrtc/fakewebrtcvoiceengine.h

Index: talk/media/webrtc/fakewebrtcvoiceengine.h
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h
index 3344b4190d2fb5e21759aeef18e00bd7c98f3465..d237b2760a6bdf82f6cd83a1fccb7601f341fd10 100644
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h
@@ -124,9 +124,9 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {
   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));