Index: talk/media/webrtc/fakewebrtcvoiceengine.h |
diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h |
index 50cdd144ee248a1a468c7a9e7ab92064d7e29085..419170b24dc471bdb2c1a8c2ec9ee7fe6fde6a5c 100644 |
--- a/talk/media/webrtc/fakewebrtcvoiceengine.h |
+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h |
@@ -112,8 +112,6 @@ |
webrtc::AudioProcessing::ChannelLayout input_layout, |
webrtc::AudioProcessing::ChannelLayout output_layout, |
webrtc::AudioProcessing::ChannelLayout reverse_layout)); |
- WEBRTC_STUB(Initialize, ( |
- const webrtc::ProcessingConfig& processing_config)); |
WEBRTC_VOID_FUNC(SetExtraOptions, (const webrtc::Config& config)) { |
experimental_ns_enabled_ = config.Get<webrtc::ExperimentalNs>().enabled; |
@@ -138,20 +136,12 @@ |
int output_sample_rate_hz, |
webrtc::AudioProcessing::ChannelLayout output_layout, |
float* const* dest)); |
- WEBRTC_STUB(ProcessStream, |
- (const float* const* src, |
- const webrtc::StreamConfig& input_config, |
- const webrtc::StreamConfig& output_config, |
- float* const* dest)); |
WEBRTC_STUB(AnalyzeReverseStream, (webrtc::AudioFrame* frame)); |
WEBRTC_STUB(AnalyzeReverseStream, ( |
const float* const* data, |
int samples_per_channel, |
int sample_rate_hz, |
webrtc::AudioProcessing::ChannelLayout layout)); |
- WEBRTC_STUB(AnalyzeReverseStream, ( |
- const float* const* data, |
- const webrtc::StreamConfig& reverse_config)); |
WEBRTC_STUB(set_stream_delay_ms, (int delay)); |
WEBRTC_STUB_CONST(stream_delay_ms, ()); |
WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ()); |