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 84 matching lines @@
   int ProcessReverseStream(const float* const* src,
                            const StreamConfig& reverse_input_config,
                            const StreamConfig& reverse_output_config,
                            float* const* dest) override;
   int set_stream_delay_ms(int delay) override;
   int stream_delay_ms() const override;
   bool was_stream_delay_set() const override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
-  int StartDebugRecording(const char filename[kMaxFilenameSize]) override;
-  int StartDebugRecording(FILE* handle) 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 StartDebugRecordingForPlatformFile(rtc::PlatformFile handle) override;
   int StopDebugRecording() override;
   void UpdateHistogramsOnCallEnd() override;
   EchoCancellation* echo_cancellation() const override;
   EchoControlMobile* echo_control_mobile() const override;
   GainControl* gain_control() const override;
   HighPassFilter* high_pass_filter() const override;
   LevelEstimator* level_estimator() const override;
   NoiseSuppression* noise_suppression() const override;
   VoiceDetection* voice_detection() const override;
@@ skipping 50 matching lines @@
   // it is different from the last saved one; if |forced|, writes the config
   // regardless of the last saved.
   int WriteConfigMessage(bool forced);
 
   rtc::scoped_ptr<FileWrapper> debug_file_;
   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_;
+
+  // 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 nr_bytes_left_for_log;

[hlundin-webrtc, 2015/11/11 15:55:05]

Trailing underscore on member variable names.

[ivoc, 2015/11/11 16:44:31]

Right, fixed.

 #endif
 
   // Format of processing streams at input/output call sites.
   ProcessingConfig api_format_;
 
   // Only the rate and samples fields of fwd_proc_format_ are used because the
   // forward processing number of channels is mutable and is tracked by the
   // capture_audio_.
   StreamConfig fwd_proc_format_;
   StreamConfig rev_proc_format_;
@@ skipping 22 matching lines @@
   rtc::scoped_ptr<Beamformer<float>> beamformer_;
   const std::vector<Point> array_geometry_;
 
   bool intelligibility_enabled_;
   rtc::scoped_ptr<IntelligibilityEnhancer> intelligibility_enhancer_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_