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

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 1241 matching lines @@
     adm_->Release();
     adm_ = NULL;
   }
   if (adm) {
     adm_ = adm;
     adm_->AddRef();
   }
   return true;
 }
 
-bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file) {
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
+                                     int64_t max_size_bytes) {
   FILE* aec_dump_file_stream = rtc::FdopenPlatformFileForWriting(file);
   if (!aec_dump_file_stream) {
     LOG(LS_ERROR) << "Could not open AEC dump file stream.";
     if (!rtc::ClosePlatformFile(file))
       LOG(LS_WARNING) << "Could not close file.";
     return false;
   }
   StopAecDump();
-  if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
+  if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
+          aec_dump_file_stream, max_size_bytes) !=
       webrtc::AudioProcessing::kNoError) {
     LOG_RTCERR0(StartDebugRecording);
     fclose(aec_dump_file_stream);
     return false;
   }
   is_dumping_aec_ = true;
   return true;
 }
 
 void WebRtcVoiceEngine::StartAecDump(const std::string& filename) {
   if (!is_dumping_aec_) {
     // Start dumping AEC when we are not dumping.
-    if (voe_wrapper_->processing()->StartDebugRecording(
-        filename.c_str()) != webrtc::AudioProcessing::kNoError) {
+    if (voe_wrapper_->base()->audio_processing()->StartDebugRecording(
+            filename.c_str(), -1) != webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR1(StartDebugRecording, filename.c_str());
     } else {
       is_dumping_aec_ = true;
     }
   }
 }
 
 void WebRtcVoiceEngine::StopAecDump() {
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
-    if (voe_wrapper_->processing()->StopDebugRecording() !=
+    if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }
     is_dumping_aec_ = false;
   }
 }
 
 bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
   return voe_wrapper_->codec()->GetEventLog()->StartLogging(file);
 }
@@ skipping 1688 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE