Revert of Remove the deprecated EncodeInternal interface from AudioEncoder

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 42 matching lines @@
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
   Reset();
 }
 
 AudioEncoderG722::AudioEncoderG722(const CodecInst& codec_inst)
     : AudioEncoderG722(CreateConfig(codec_inst)) {}
 
 AudioEncoderG722::~AudioEncoderG722() = default;
 
+size_t AudioEncoderG722::MaxEncodedBytes() const {
+  return SamplesPerChannel() / 2 * num_channels_;
+}
+
 int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 size_t AudioEncoderG722::NumChannels() const {
   return num_channels_;
 }
 
 int AudioEncoderG722::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
@@ skipping 81 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