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.
  */
 
@@ skipping 11 matching lines @@
 #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 webrtc {
 
 struct AecCore;
 
+class AecDump;
 class AudioFrame;
 
 class NonlinearBeamformer;
 
 class StreamConfig;
 class ProcessingConfig;
 
 class EchoCancellation;
 class EchoControlMobile;
 class GainControl;
@@ skipping 398 matching lines @@
   virtual void set_stream_key_pressed(bool key_pressed) = 0;
 
   // Sets a delay |offset| in ms to add to the values passed in through
   // 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. 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;
-
-  // 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;
-
-  // TODO(ivoc): Remove this function after Chrome stops using it.
-  virtual int StartDebugRecording(FILE* handle) = 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;
+  // TODO(aleloi): doc
+  virtual void StartDebugRecording(std::unique_ptr<AecDump> aec_dump) = 0;
 
   // Stops recording debugging information, and closes the file. Recording
   // cannot be resumed in the same file (without overwriting it).
-  virtual int StopDebugRecording() = 0;
+  virtual void 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
   //             API.
   struct Statistic {
     int instant = 0;  // Instantaneous value.
     int average = 0;  // Long-term average.
@@ skipping 585 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_