AudioCodingModuleImpl::Encode: Use a Buffer instead of a stack-allocated array

webrtc/modules/audio_coding/main/acm2/acm_common_defs.h

 /*
  *  Copyright (c) 2011 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_
 
 #include "webrtc/engine_configurations.h"
 
 // Checks for enabled codecs, we prevent enabling codecs which are not
 // compatible.
 #if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
 #error iSAC and iSACFX codecs cannot be enabled at the same time
 #endif
 
 namespace webrtc {
 
-// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
-#define MAX_PAYLOAD_SIZE_BYTE 7680
-
 // General codec specific defines
 const int kIsacWbDefaultRate = 32000;
 const int kIsacSwbDefaultRate = 56000;
 const int kIsacPacSize480 = 480;
 const int kIsacPacSize960 = 960;
 const int kIsacPacSize1440 = 1440;
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_ACM2_ACM_COMMON_DEFS_H_