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

webrtc/modules/audio_processing/audio_processing_impl.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 39 matching lines @@
   int Initialize() override;
   int Initialize(int input_sample_rate_hz,
                  int output_sample_rate_hz,
                  int reverse_sample_rate_hz,
                  ChannelLayout input_layout,
                  ChannelLayout output_layout,
                  ChannelLayout reverse_layout) override;
   int Initialize(const ProcessingConfig& processing_config) override;
   void SetExtraOptions(const Config& config) override;
   void UpdateHistogramsOnCallEnd() override;
-  int StartDebugRecording(const char filename[kMaxFilenameSize],
-                          int64_t max_log_size_bytes) override;
-  int StartDebugRecording(FILE* handle, int64_t max_log_size_bytes) override;
+  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
+  int StartDebugRecording(FILE* handle) override;
   int StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
 
   // Capture-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the capture lock.
   int ProcessStream(AudioFrame* frame) override;
   int ProcessStream(const float* const* src,
                     size_t samples_per_channel,
                     int input_sample_rate_hz,
                     ChannelLayout input_layout,
@@ skipping 63 matching lines @@
     ApmDebugDumpThreadState() : event_msg(new audioproc::Event()) {}
     rtc::scoped_ptr<audioproc::Event> event_msg;  // Protobuf message.
     std::string event_str;  // Memory for protobuf serialization.
 
     // Serialized string of last saved APM configuration.
     std::string last_serialized_config;
   };
 
   struct ApmDebugDumpState {
     ApmDebugDumpState() : debug_file(FileWrapper::Create()) {}
-    // Number of bytes that can still be written to the log before the maximum
-    // size is reached. A value of <= 0 indicates that no limit is used.
-    int64_t num_bytes_left_for_log_ = -1;
     rtc::scoped_ptr<FileWrapper> debug_file;
     ApmDebugDumpThreadState render;
     ApmDebugDumpThreadState capture;
   };
 #endif
 
   // Method for modifying the formats struct that are called from both
   // the render and capture threads. The check for whether modifications
   // are needed is done while holding the render lock only, thereby avoiding
   // that the capture thread blocks the render thread.
@@ skipping 54 matching lines @@
       EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
   bool is_rev_processed() const EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
   int ProcessReverseStreamLocked() EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
 
 // Debug dump methods that are internal and called without locks.
 // TODO(peah): Make thread safe.
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   // TODO(andrew): make this more graceful. Ideally we would split this stuff
   // out into a separate class with an "enabled" and "disabled" implementation.
   static int WriteMessageToDebugFile(FileWrapper* debug_file,
-                                     int64_t* filesize_limit_bytes,
                                      rtc::CriticalSection* crit_debug,
                                      ApmDebugDumpThreadState* debug_state);
   int WriteInitMessage() EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
 
   // Writes Config message. If not |forced|, only writes the current config if
   // it is different from the last saved one; if |forced|, writes the config
   // regardless of the last saved.
   int WriteConfigMessage(bool forced) EXCLUSIVE_LOCKS_REQUIRED(crit_capture_)
       EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
 
@@ skipping 98 matching lines @@
 
   struct ApmRenderState {
     rtc::scoped_ptr<AudioConverter> render_converter;
     rtc::scoped_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_