Test submission of complete AEC-dump refactoring.

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 #define WEBRTC_MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
 
 // MSVC++ requires this to be set before any other includes to get M_PI.
 #define _USE_MATH_DEFINES
 
 #include <math.h>
 #include <stddef.h>  // size_t
 #include <stdio.h>  // FILE
 #include <vector>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/platform_file.h"
 #include "webrtc/modules/audio_processing/beamformer/array_util.h"
 #include "webrtc/modules/audio_processing/include/config.h"
 #include "webrtc/typedefs.h"
 
+namespace rtc {
+class TaskQueue;
+}  // namespace rtc
+
 namespace webrtc {
 
 struct AecCore;
 
 class AudioFrame;
 
 class NonlinearBeamformer;
 
 class StreamConfig;
 class ProcessingConfig;
@@ skipping 411 matching lines @@
 
   // 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. 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],
-                                  int64_t max_log_size_bytes) = 0;
+                                  int64_t max_log_size_bytes,
+                                  rtc::TaskQueue* worker_queue) = 0;
 
   // Same as above but uses an existing file handle. Takes ownership
   // of |handle| and closes it at StopDebugRecording().
-  virtual int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) = 0;
+  virtual int StartDebugRecording(FILE* handle,
+                                  int64_t max_log_size_bytes,
+                                  rtc::TaskQueue* worker_queue) = 0;
 
   // TODO(ivoc): Remove this function after Chrome stops using it.
-  virtual int StartDebugRecording(FILE* handle) = 0;
+  virtual int StartDebugRecording(FILE* handle,
+                                  rtc::TaskQueue* worker_queue) = 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) = 0;
+  virtual int StartDebugRecordingForPlatformFile(
+      rtc::PlatformFile handle,
+      rtc::TaskQueue* worker_queue) = 0;
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
   virtual int StopDebugRecording() = 0;
 
   // Use to send UMA histograms at end of a call. Note that all histogram
   // specific member variables are reset.
   virtual void UpdateHistogramsOnCallEnd() = 0;
 
   // TODO(ivoc): Remove when the calling code no longer uses the old Statistics
@@ skipping 589 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_