Fix invalid output buffer usage

webrtc/voice_engine/coder.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 77 matching lines @@
   // Fake a timestamp in case audio doesn't contain a correct timestamp.
   // Make a local copy of the audio frame since audio is const
   AudioFrame audio_frame;
   audio_frame.CopyFrom(audio);
   audio_frame.timestamp_ = encode_timestamp_;
   encode_timestamp_ += static_cast<uint32_t>(audio_frame.samples_per_channel_);
 
   // For any codec with a frame size that is longer than 10 ms the encoded
   // length in bytes should be zero until a a full frame has been encoded.
   encoded_length_in_bytes_ = 0;
+  encoded_data_ = encoded_data;
   if (acm_->Add10MsData((AudioFrame&)audio_frame) == -1) {
     return -1;
   }
-  encoded_data_ = encoded_data;
+
   *encoded_length_in_bytes = encoded_length_in_bytes_;
   return 0;
 }
 
 int32_t AudioCoder::SendData(FrameType /* frame_type */,
                              uint8_t /* payload_type */,
                              uint32_t /* time_stamp */,
                              const uint8_t* payload_data,
                              size_t payload_size,
                              const RTPFragmentationHeader* /* fragmentation*/) {
   memcpy(encoded_data_, payload_data, sizeof(uint8_t) * payload_size);
   encoded_length_in_bytes_ = payload_size;
   return 0;
 }
 
 }  // namespace webrtc