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

webrtc/modules/audio_processing/include/audio_processing.h

 /*
  *  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 375 matching lines @@
   virtual void set_delay_offset_ms(int offset) = 0;
   virtual int delay_offset_ms() const = 0;
 
   // Starts recording debugging information to a file specified by |filename|,
   // a NULL-terminated string. If there is an ongoing recording, the old file
   // will be closed, and recording will continue in the newly specified file.
   // An already existing file will be overwritten without warning.
   static const size_t kMaxFilenameSize = 1024;
   virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
 
+  // Same as above, but a maximum file size (in bytes) for the log can be
+  // specified. The logging is stopped once the limit has been reached.
+  virtual int StartDebugRecording(const char filename[kMaxFilenameSize],

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

Consolidate to one interface, but keep a forwarding impl of the old version
until Chromium code has been updated.

[ivoc, 2015/11/05 13:14:46]

Done.

+                                  int max_log_size_bytes) = 0;
+
   // Same as above but uses an existing file handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
   virtual int StartDebugRecording(FILE* handle) = 0;
 
+  // Same as above, but a maximum file size (in bytes) for the log can be
+  // specified. The logging is stopped once the limit has been reached.
+  virtual int StartDebugRecording(FILE* handle, int max_log_size_bytes) = 0;
+
   // Same as above but uses an existing PlatformFile handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
   // TODO(xians): Make this interface pure virtual.
   virtual int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) {
       return -1;
   }
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;
@@ skipping 519 matching lines @@
   // This does not impact the size of frames passed to |ProcessStream()|.
   virtual int set_frame_size_ms(int size) = 0;
   virtual int frame_size_ms() const = 0;
 
  protected:
   virtual ~VoiceDetection() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_