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 386 matching lines @@
   // set_stream_delay_ms(). May be positive or negative.
   //
   // Note that this could cause an otherwise valid value passed to
   // set_stream_delay_ms() to return an error.
   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.
+  // An already existing file will be overwritten without warning. A maximum
+  // file size (in bytes) for the log can be specified. The logging is stopped
+  // once the limit has been reached. If max_log_size_bytes is set to a value
+  // <= 0, no limit will be used.
   static const size_t kMaxFilenameSize = 1024;
-  virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
+  virtual int StartDebugRecording(const char filename[kMaxFilenameSize],
+                                  int64_t 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;
+  virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
+
+  // TODO(ivoc): Remove this function after Chromium switches to the one above.
+  virtual int StartDebugRecording(FILE* handle) {
+    return StartDebugRecording(handle, -1);
+  }
 
   // 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).
@@ skipping 520 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_