Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

 /*
  *  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 96 matching lines @@
 }
 
 template <typename T>
 int AudioEncoderIsacT<T>::GetTargetBitrate() const {
   if (config_.adaptive_mode)
     return -1;
   return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
 }
 
 template <typename T>
-AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   if (!packet_in_progress_) {
     // Starting a new packet; remember the timestamp for later.
     packet_in_progress_ = true;
     packet_timestamp_ = rtp_timestamp;
   }
   if (bwinfo_) {
     IsacBandwidthInfo bwinfo = bwinfo_->Get();
@@ skipping 57 matching lines @@
   // we get an encoding that isn't bit-for-bit identical with what a combined
   // encoder+decoder object produces.
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
   config_ = config;
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_