Clean out unused methods from VoiceEngine and VoEBase.

media/engine/webrtcvoiceengine.h

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 #define MEDIA_ENGINE_WEBRTCVOICEENGINE_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "api/audio_codecs/audio_encoder_factory.h"
 #include "api/rtpreceiverinterface.h"
 #include "call/audio_state.h"
 #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"
 #include "rtc_base/buffer.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/networkroute.h"
 #include "rtc_base/scoped_ref_ptr.h"
 #include "rtc_base/task_queue.h"
 #include "rtc_base/thread_checker.h"
 
@@ skipping 46 matching lines @@
   const std::vector<AudioCodec>& recv_codecs() const;
   RtpCapabilities GetCapabilities() const;
 
   // For tracking WebRtc channels. Needed because we have to pause them
   // all when switching devices.
   // May only be called by WebRtcVoiceMediaChannel.
   void RegisterChannel(WebRtcVoiceMediaChannel* channel);
   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
   // the file is closed. If max_size_bytes is set to <= 0, no limit will be
   // used.
   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
 
   // Stops AEC dump.
   void StopAecDump();
 
@@ skipping 139 matching lines @@
   int GetSendChannelId(uint32_t ssrc) const;
 
  private:
   bool SetOptions(const AudioOptions& options);
   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
   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);
   bool SetMaxSendBitrate(int bps);
   bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
   void SetupRecording();
   // Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
   // unsignaled anymore (i.e. it is now removed, or signaled), and return true.
   bool MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc);
 
@@ skipping 39 matching lines @@
   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
 
   rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
       send_codec_spec_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
 };
 }  // namespace cricket
 
 #endif  // MEDIA_ENGINE_WEBRTCVOICEENGINE_H_