Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc

 /*
  *  Copyright (c) 2014 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 71 matching lines @@
       // 38 bytes per frame of 20 ms => 15200 bits/s.
       return 15200;
     case 3: case 6:
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
       return 13333;
     default:
       FATAL();
   }
 }
 
-AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoderIlbc::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
 
   // Save timestamp if starting a new packet.
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Buffer input.
   std::copy(audio.cbegin(), audio.cend(),
@@ skipping 47 matching lines @@
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
     case 4:   return 2 * 38;
     case 6:   return 2 * 50;
     default:  FATAL();
   }
 }
 
 }  // namespace webrtc