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

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.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_ISAC_AUDIO_ENCODER_ISAC_T_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_
 
 #include <vector>
 
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 template <typename T>
 class AudioEncoderIsacT final : public AudioEncoder {
  public:
+  using AudioEncoder::EncodeInternal;
+
   // Allowed combinations of sample rate, frame size, and bit rate are
   //  - 16000 Hz, 30 ms, 10000-32000 bps
   //  - 16000 Hz, 60 ms, 10000-32000 bps
   //  - 32000 Hz, 30 ms, 10000-56000 bps (if T has super-wideband support)
   struct Config {
     bool IsOk() const;
 
     LockedIsacBandwidthInfo* bwinfo = nullptr;
 
     int payload_type = 103;
@@ skipping 19 matching lines @@
   ~AudioEncoderIsacT() 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;
+                             rtc::Buffer* encoded) override;
   void Reset() override;
 
  private:
   // This value is taken from STREAM_SIZE_MAX_60 for iSAC float (60 ms) and
   // STREAM_MAXW16_60MS for iSAC fix (60 ms).
   static const size_t kSufficientEncodeBufferSizeBytes = 400;
 
   static const int kDefaultBitRate = 32000;
 
   // Recreate the iSAC encoder instance with the given settings, and save them.
@@ skipping 11 matching lines @@
 
   // Timestamp of the previously encoded packet.
   uint32_t last_encoded_timestamp_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderIsacT);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_H_