Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size.

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
 #include <memory>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderG722 final : public AudioEncoder {
  public:
+  using AudioEncoder::EncodeInternal;
+
   struct Config {
     bool IsOk() const;
 
     int payload_type = 9;
     int frame_size_ms = 20;
     size_t num_channels = 1;
   };
 
   explicit AudioEncoderG722(const Config& config);
   explicit AudioEncoderG722(const CodecInst& codec_inst);
   ~AudioEncoderG722() override;
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   int RtpTimestampRateHz() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
+  void Reset() override;
+
+protected:
   EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
                              rtc::ArrayView<const int16_t> audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
-  void Reset() override;
+                             rtc::Buffer* encoded) override;
 
  private:
   // The encoder state for one channel.
   struct EncoderState {
     G722EncInst* encoder;
     std::unique_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
     rtc::Buffer encoded_buffer;                 // Already encoded.
     EncoderState();
     ~EncoderState();
   };
 
   size_t SamplesPerChannel() const;
 
   const size_t num_channels_;
   const int payload_type_;
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
   const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_