Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.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 79 matching lines @@
 }
 
 size_t AudioEncoderCng::Max10MsFramesInAPacket() const {
   return speech_encoder_->Max10MsFramesInAPacket();
 }
 
 int AudioEncoderCng::GetTargetBitrate() const {
   return speech_encoder_->GetTargetBitrate();
 }
 
-AudioEncoder::EncodedInfo AudioEncoderCng::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
   RTC_CHECK_EQ(speech_buffer_.size(),
                rtp_timestamps_.size() * samples_per_10ms_frame);
   rtp_timestamps_.push_back(rtp_timestamp);
   RTC_DCHECK_EQ(samples_per_10ms_frame, audio.size());
   speech_buffer_.insert(speech_buffer_.end(), audio.cbegin(), audio.cend());
   const size_t frames_to_encode = speech_encoder_->Num10MsFramesInNextPacket();
@@ skipping 148 matching lines @@
     }
   }
   return info;
 }
 
 size_t AudioEncoderCng::SamplesPer10msFrame() const {
   return rtc::CheckedDivExact(10 * SampleRateHz(), 1000);
 }
 
 }  // namespace webrtc