Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index bf22a290b89673cc245ff05d717ac94d161433e3..5fc2f485190c14b9ca5a2211f2f7f40b273e71dc 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -77,9 +77,9 @@ class FakeAudioProcessing : public webrtc::AudioProcessing { |
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_S
Peter Kasting
2016/01/07 09:47:50
AFAICT it's just to reduce verbosity. I don't see
the sun
2016/01/07 12:27:59
Don't worry; this is going away. Don't create any
|
+ 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, ( |