Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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 59 matching lines @@
 
 size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
   return static_cast<int>(
       8 * BytesPerSample() * SampleRateHz() * NumChannels());
 }
 
-AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
   speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
@@ skipping 36 matching lines @@
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 }  // namespace webrtc