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

Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index d611226d7597bb45951883bf8d6c311c2d633f26..0d7b919aa23d6e4b3eb9337b3a90112019003832 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -23,6 +23,8 @@ struct CodecInst;
 
 class AudioEncoderG722 final : public AudioEncoder {
  public:
+  using AudioEncoder::EncodeInternal;
+
   struct Config {
     bool IsOk() const;
 
@@ -42,11 +44,12 @@ class AudioEncoderG722 final : public AudioEncoder {
   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.