Fix the maximum native sample rate in AudioProcessing

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 476 matching lines @@
     rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
     rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
     rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
     rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
     TelephoneEventHandler* telephone_event_handler_;
     rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
     rtc::scoped_ptr<AudioCodingModule> audio_coding_;
     AudioLevel _outputAudioLevel;
     bool _externalTransport;
     AudioFrame _audioFrame;
-    rtc::scoped_ptr<int16_t[]> mono_recording_audio_;
     // Downsamples to the codec rate if necessary.
     PushResampler<int16_t> input_resampler_;
     FilePlayer* _inputFilePlayerPtr;
     FilePlayer* _outputFilePlayerPtr;
     FileRecorder* _outputFileRecorderPtr;
     int _inputFilePlayerId;
     int _outputFilePlayerId;
     int _outputFileRecorderId;
     bool _outputFileRecording;
     DtmfInbandQueue _inbandDtmfQueue;
@@ skipping 70 matching lines @@
     rtc::scoped_ptr<NetworkPredictor> network_predictor_;
     // An associated send channel.
     rtc::scoped_ptr<CriticalSectionWrapper> assoc_send_channel_lock_;
     ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_