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

Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
index 321dac3567b64bb96ef7db2bbc7ffd369cf3f685..2e15fb562000ba40793fb36c086a43658c196e58 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t.h
@@ -23,6 +23,8 @@ 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
@@ -62,8 +64,7 @@ class AudioEncoderIsacT final : public AudioEncoder {
   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: