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_impl.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.
  */
 
@@ skipping 99 matching lines @@
 int AudioEncoderIsacT<T>::GetTargetBitrate() const {
   if (config_.adaptive_mode)
     return -1;
   return config_.bit_rate == 0 ? kDefaultBitRate : config_.bit_rate;
 }
 
 template <typename T>
 AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
+    rtc::Buffer* encoded) {
   if (!packet_in_progress_) {
     // Starting a new packet; remember the timestamp for later.
     packet_in_progress_ = true;
     packet_timestamp_ = rtp_timestamp;
   }
   if (bwinfo_) {
     IsacBandwidthInfo bwinfo = bwinfo_->Get();
     T::SetBandwidthInfo(isac_state_, &bwinfo);
   }
-  int r = T::Encode(isac_state_, audio.data(), encoded);
-  RTC_CHECK_GE(r, 0) << "Encode failed (error code "
-                     << T::GetErrorCode(isac_state_) << ")";
 
-  // T::Encode doesn't allow us to tell it the size of the output
-  // buffer. All we can do is check for an overrun after the fact.
-  RTC_CHECK_LE(static_cast<size_t>(r), max_encoded_bytes);
+  size_t encoded_bytes = encoded->AppendData(
+      kSufficientEncodeBufferSizeBytes,
+      [&] (rtc::ArrayView<uint8_t> encoded) {
+        int r = T::Encode(isac_state_, audio.data(), encoded.data());
 
-  if (r == 0)
+        RTC_CHECK_GE(r, 0) << "Encode failed (error code "
+                           << T::GetErrorCode(isac_state_) << ")";
+
+        return static_cast<size_t>(r);
+      });
+
+  if (encoded_bytes == 0)
     return EncodedInfo();
 
   // Got enough input to produce a packet. Return the saved timestamp from
   // the first chunk of input that went into the packet.
   packet_in_progress_ = false;
   EncodedInfo info;
-  info.encoded_bytes = r;
+  info.encoded_bytes = encoded_bytes;
   info.encoded_timestamp = packet_timestamp_;
   info.payload_type = config_.payload_type;
   return info;
 }
 
 template <typename T>
 void AudioEncoderIsacT<T>::Reset() {
   RecreateEncoderInstance(config_);
 }
 
@@ skipping 25 matching lines @@
   // we get an encoding that isn't bit-for-bit identical with what a combined
   // encoder+decoder object produces.
   RTC_CHECK_EQ(0, T::SetDecSampRate(isac_state_, config.sample_rate_hz));
 
   config_ = config;
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_