| 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, ());
|
|
|