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); |