Revert of Remove the deprecated EncodeInternal interface from AudioEncoder

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 34 matching lines @@
           config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
       first_timestamp_in_buffer_(0) {
   RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
   RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   speech_buffer_.reserve(full_frame_samples_);
 }
 
 AudioEncoderPcm::~AudioEncoderPcm() = default;
 
+size_t AudioEncoderPcm::MaxEncodedBytes() const {
+  return full_frame_samples_ * BytesPerSample();
+}
+
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
 size_t AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
@@ skipping 17 matching lines @@
   }
   speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes =
-      encoded->AppendData(full_frame_samples_ * BytesPerSample(),
+      encoded->AppendData(MaxEncodedBytes(),
                           [&] (rtc::ArrayView<uint8_t> encoded) {
                             return EncodeCall(&speech_buffer_[0],
                                               full_frame_samples_,
                                               encoded.data());
                           });
   speech_buffer_.clear();
   return info;
 }
 
 void AudioEncoderPcm::Reset() {
@@ skipping 20 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