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

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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.
  */
 
@@ skipping 23 matching lines @@
   };
 
   ~AudioEncoderPcm() override;
 
   size_t MaxEncodedBytes() const override;
   int SampleRateHz() const override;
   size_t NumChannels() const override;
   size_t Num10MsFramesInNextPacket() const override;
   size_t Max10MsFramesInAPacket() const override;
   int GetTargetBitrate() const override;
-  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
-                             rtc::ArrayView<const int16_t> audio,
-                             size_t max_encoded_bytes,
-                             uint8_t* encoded) override;
   void Reset() override;
 
  protected:
   AudioEncoderPcm(const Config& config, int sample_rate_hz);
 
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             rtc::Buffer* encoded) override;
+
   virtual size_t EncodeCall(const int16_t* audio,
                             size_t input_len,
                             uint8_t* encoded) = 0;
 
   virtual size_t BytesPerSample() const = 0;
 
  private:
   const int sample_rate_hz_;
   const size_t num_channels_;
   const int payload_type_;
@@ skipping 45 matching lines @@
   size_t BytesPerSample() const override;
 
  private:
   static const int kSampleRateHz = 8000;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderPcmU);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_G711_AUDIO_ENCODER_PCM_H_