Added option to specify a maximum file size when recording an AEC dump.

webrtc/voice_engine/voe_audio_processing_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 906 matching lines @@
 }
 
 int VoEAudioProcessingImpl::StartDebugRecording(const char* fileNameUTF8) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "StartDebugRecording()");
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
 
-  return _shared->audio_processing()->StartDebugRecording(fileNameUTF8);
+  return _shared->audio_processing()->StartDebugRecording(fileNameUTF8, -1);
 }
 
 int VoEAudioProcessingImpl::StartDebugRecording(FILE* file_handle) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "StartDebugRecording()");
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
 
-  return _shared->audio_processing()->StartDebugRecording(file_handle);
+  return _shared->audio_processing()->StartDebugRecording(file_handle, -1);
 }
 
 int VoEAudioProcessingImpl::StopDebugRecording() {
   WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_shared->instance_id(), -1),
                "StopDebugRecording()");
   if (!_shared->statistics().Initialized()) {
     _shared->SetLastError(VE_NOT_INITED, kTraceError);
     return -1;
   }
 
@@ skipping 89 matching lines @@
   _shared->transmit_mixer()->EnableStereoChannelSwapping(enable);
 }
 
 bool VoEAudioProcessingImpl::IsStereoChannelSwappingEnabled() {
   return _shared->transmit_mixer()->IsStereoChannelSwappingEnabled();
 }
 
 #endif  // #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
 
 }  // namespace webrtc