Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility.

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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 @@
   // 4 bits/sample, 16000 samples/s/channel.
   return static_cast<int>(64000 * NumChannels());
 }
 
 void AudioEncoderG722::Reset() {
   num_10ms_frames_buffered_ = 0;
   for (size_t i = 0; i < num_channels_; ++i)
     RTC_CHECK_EQ(0, WebRtcG722_EncoderInit(encoders_[i].encoder));
 }
 
-AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
+AudioEncoder::EncodedInfo AudioEncoderG722::EncodeImpl(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
     rtc::Buffer* encoded) {
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Deinterleave samples and save them in each channel's buffer.
   const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
   for (size_t i = 0; i < kSampleRateHz / 100; ++i)
     for (size_t j = 0; j < num_channels_; ++j)
@@ skipping 47 matching lines @@
 
 AudioEncoderG722::EncoderState::~EncoderState() {
   RTC_CHECK_EQ(0, WebRtcG722_FreeEncoder(encoder));
 }
 
 size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 }  // namespace webrtc