Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index eb3b4d3b18abd0e1c9ca7cb7d9296b2d63eeddaa..65ba927cc55f677ed892497a288f273afd8e29d7 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -77,10 +77,10 @@ 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_proc_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_proc_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_STUB(ProcessStream, (webrtc::AudioFrame* frame)); |
WEBRTC_STUB(ProcessStream, ( |