| Index: media/engine/webrtcvoiceengine.h
|
| diff --git a/media/engine/webrtcvoiceengine.h b/media/engine/webrtcvoiceengine.h
|
| index e27857696a64d207e8d5713ff2d3d11c1f9736af..87b7a947271b576e42cd48137ee998c237fec9db 100644
|
| --- a/media/engine/webrtcvoiceengine.h
|
| +++ b/media/engine/webrtcvoiceengine.h
|
| @@ -22,7 +22,6 @@
|
| #include "call/call.h"
|
| #include "media/base/rtputils.h"
|
| #include "media/engine/apm_helpers.h"
|
| -#include "media/engine/webrtccommon.h"
|
| #include "media/engine/webrtcvoe.h"
|
| #include "modules/audio_processing/include/audio_processing.h"
|
| #include "pc/channel.h"
|
| @@ -89,7 +88,6 @@ class WebRtcVoiceEngine final : public webrtc::TraceCallback {
|
| void UnregisterChannel(WebRtcVoiceMediaChannel* channel);
|
|
|
| VoEWrapper* voe() { return voe_wrapper_.get(); }
|
| - int GetLastEngineError();
|
|
|
| // Starts AEC dump using an existing file. A maximum file size in bytes can be
|
| // specified. When the maximum file size is reached, logging is stopped and
|
| @@ -249,7 +247,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
|
| bool MuteStream(uint32_t ssrc, bool mute);
|
|
|
| WebRtcVoiceEngine* engine() { return engine_; }
|
| - int GetLastEngineError() { return engine()->GetLastEngineError(); }
|
| void ChangePlayout(bool playout);
|
| int CreateVoEChannel();
|
| bool DeleteVoEChannel(int channel);
|
|
|