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 1260 matching lines @@
   if (voe_wrapper_->processing()->StartDebugRecording(aec_dump_file_stream) !=
       webrtc::AudioProcessing::kNoError) {
     LOG_RTCERR0(StartDebugRecording);
     fclose(aec_dump_file_stream);
     return false;
   }
   is_dumping_aec_ = true;
   return true;
 }
 
+bool WebRtcVoiceEngine::StartAecDump(rtc::PlatformFile file,
+                                     int 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,
+                                                      max_size_bytes) !=
+      webrtc::AudioProcessing::kNoError) {
+    LOG_RTCERR0(StartDebugRecording);
+    fclose(aec_dump_file_stream);
+    return false;
+  }
+  is_dumping_aec_ = true;
+  return true;
+}

[kwiberg-webrtc, 2015/10/25 02:29:12]

This duplicates a lot of the preceding function. Can you make one of them call
the other, or extract a new method for both of them to call?

[the sun, 2015/10/26 10:37:20]

See previous comment in PeerConnectionFactory: we can make do with just this
method here.

[ivoc, 2015/11/05 13:14:45]

Done.

+
 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) {
       LOG_RTCERR1(StartDebugRecording, filename.c_str());
     } else {
       is_dumping_aec_ = true;
     }
   }
@@ skipping 1704 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE