Remove RentACodec::GetEncoderStack

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.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 233 matching lines @@
   rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
   int id_;  // TODO(henrik.lundin) Make const.
   uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
   uint32_t expected_in_ts_ GUARDED_BY(acm_crit_sect_);
   ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
   AcmReceiver receiver_;  // AcmReceiver has it's own internal lock.
   ChangeLogger bitrate_logger_ GUARDED_BY(acm_crit_sect_);
   CodecManager codec_manager_ GUARDED_BY(acm_crit_sect_);
   RentACodec rent_a_codec_ GUARDED_BY(acm_crit_sect_);
 
+  // Last encoder stack obtained from rent_a_codec_.RentEncoderStack.
+  AudioEncoder* encoder_stack_ GUARDED_BY(acm_crit_sect_);
+
   // This is to keep track of CN instances where we can send DTMFs.
   uint8_t previous_pltype_ GUARDED_BY(acm_crit_sect_);
 
   // Used when payloads are pushed into ACM without any RTP info
   // One example is when pre-encoded bit-stream is pushed from
   // a file.
   // IMPORTANT: this variable is only used in IncomingPayload(), therefore,
   // no lock acquired when interacting with this variable. If it is going to
   // be used in other methods, locks need to be taken.
   rtc::scoped_ptr<WebRtcRTPHeader> aux_rtp_header_;
@@ skipping 10 matching lines @@
   mutable rtc::CriticalSection callback_crit_sect_;
   AudioPacketizationCallback* packetization_callback_
       GUARDED_BY(callback_crit_sect_);
   ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
 };
 
 }  // namespace acm2
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_