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

Index: webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
index 59c8f796eea777b9b94fe65e368c00677d46df84..0711a67e620d13e09154863bbb8c80482e4b35de 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
@@ -61,11 +61,6 @@ class AudioEncoderOpus final : public AudioEncoder {
   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;
 
@@ -84,6 +79,11 @@ class AudioEncoderOpus final : public AudioEncoder {
   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;