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

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc

 /*
  *  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 62 matching lines @@
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
   return static_cast<int>(
       8 * BytesPerSample() * SampleRateHz() * NumChannels());
 }
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
     rtc::ArrayView<const int16_t> audio,
-    size_t max_encoded_bytes,
-    uint8_t* encoded) {
+    rtc::Buffer* encoded) {
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
   speech_buffer_.insert(speech_buffer_.end(), audio.begin(), audio.end());
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
-  RTC_CHECK_GE(max_encoded_bytes, full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.encoded_bytes =
-      EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
+      encoded->AppendData(MaxEncodedBytes(),
+                          [&] (rtc::ArrayView<uint8_t> encoded) {
+                            return EncodeCall(&speech_buffer_[0],
+                                              full_frame_samples_,
+                                              encoded.data());
+                          });
   speech_buffer_.clear();
   return info;
 }
 
 void AudioEncoderPcm::Reset() {
   speech_buffer_.clear();
 }
 
 AudioEncoderPcmA::AudioEncoderPcmA(const CodecInst& codec_inst)
     : AudioEncoderPcmA(CreateConfig<AudioEncoderPcmA>(codec_inst)) {}
@@ skipping 15 matching lines @@
                                     size_t input_len,
                                     uint8_t* encoded) {
   return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 size_t AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 }  // namespace webrtc