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

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.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_OPUS_AUDIO_ENCODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 
 #include <vector>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
 
 struct CodecInst;
 
 class AudioEncoderOpus final : public AudioEncoder {
  public:
+  using AudioEncoder::EncodeInternal;
+
   enum ApplicationMode {
     kVoip = 0,
     kAudio = 1,
   };
 
   struct Config {
     bool IsOk() const;
     int frame_size_ms = 20;
     size_t num_channels = 1;
     int payload_type = 120;
@@ skipping 18 matching lines @@
   explicit AudioEncoderOpus(const CodecInst& codec_inst);
   ~AudioEncoderOpus() 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;
   bool SetFec(bool enable) override;
 
   // Set Opus DTX. Once enabled, Opus stops transmission, when it detects voice
   // being inactive. During that, it still sends 2 packets (one for content, one
   // for signaling) about every 400 ms.
   bool SetDtx(bool enable) override;
 
   bool SetApplication(Application application) override;
   void SetMaxPlaybackRate(int frequency_hz) override;
   void SetProjectedPacketLossRate(double fraction) override;
   void SetTargetBitrate(int target_bps) override;
 
   // Getters for testing.
   double packet_loss_rate() const { return packet_loss_rate_; }
   ApplicationMode application() const { return config_.application; }
   bool dtx_enabled() const { return config_.dtx_enabled; }
 
+protected:
+  EncodedInfo EncodeInternal(uint32_t rtp_timestamp,
+                             rtc::ArrayView<const int16_t> audio,
+                             rtc::Buffer* encoded) override;
+
  private:
   size_t Num10msFramesPerPacket() const;
   size_t SamplesPer10msFrame() const;
   bool RecreateEncoderInstance(const Config& config);
 
   Config config_;
   double packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_