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

webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc

 /*
  *  Copyright (c) 2012 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 127 matching lines @@
                    : last_rtp_timestamp_ +
                          rtc::CheckedDivExact(
                              input_data.input_timestamp - last_timestamp_,
                              static_cast<uint32_t>(rtc::CheckedDivExact(
                                  encoder_stack_->SampleRateHz(),
                                  encoder_stack_->RtpTimestampRateHz())));
   last_timestamp_ = input_data.input_timestamp;
   last_rtp_timestamp_ = rtp_timestamp;
   first_frame_ = false;
 
-  encode_buffer_.SetSize(encoder_stack_->MaxEncodedBytes());
+  // Clear the buffer before reuse - encoded data will get appended.
+  encode_buffer_.Clear();
   encoded_info = encoder_stack_->Encode(
       rtp_timestamp, rtc::ArrayView<const int16_t>(
                          input_data.audio, input_data.audio_channel *
                                                input_data.length_per_channel),
-      encode_buffer_.size(), encode_buffer_.data());
-  encode_buffer_.SetSize(encoded_info.encoded_bytes);
+      &encode_buffer_);
+
   bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
   if (encode_buffer_.size() == 0 && !encoded_info.send_even_if_empty) {
     // Not enough data.
     return 0;
   }
   previous_pltype = previous_pltype_;  // Read it while we have the critsect.
 
   RTPFragmentationHeader my_fragmentation;
   ConvertEncodedInfoToFragmentationHeader(encoded_info, &my_fragmentation);
   FrameType frame_type;
@@ skipping 651 matching lines @@
   return receiver_.LeastRequiredDelayMs();
 }
 
 void AudioCodingModuleImpl::GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const {
   receiver_.GetDecodingCallStatistics(call_stats);
 }
 
 }  // namespace acm2
 }  // namespace webrtc