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 85 matching lines @@
                            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(const char filename[kMaxFilenameSize],

[the sun, 2015/10/26 10:37:20]

I think it makes sense to keep just the interface with a given size here as
well.

[ivoc, 2015/11/05 13:14:46]

Done.

+                          int max_log_size_bytes) override;
   int StartDebugRecording(FILE* handle) override;
+  int StartDebugRecording(FILE* handle, int 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 -1 indicates that no limit is used.
+  int nr_bytes_left_for_log;

[kwiberg-webrtc, 2015/10/25 02:29:12]

Split comment into separate sentences at the comma. And underscore at the end of
the name.

Oh, and using int means that you can't give a limit larger than 2 GB - 1 byte.
Probably not an immediate concern, but probably worth fixing now anyway while
it's easy.

[hlundin-webrtc, 2015/10/30 08:02:21]

Yes, and in general, variables dealing with number of bytes, indices, and counts
should be size_t. But I do realize you get a problem with the semantics for not
setting a limit. Let's discuss offline.

[kwiberg-webrtc, 2015/10/30 09:53:21]

Variables dealing with number of bytes, indices, and counts *in RAM* should be
size_t. For counting the same things *on disk*, use int64_t or uint64_t
(opinions vary on which is the right one; either one will work just fine).
Otherwise, 32-bit systems won't be able to handle files larger than 4 GB.

[hlundin-webrtc, 2015/10/30 09:59:14]

Ack. I stand corrected.

[ivoc, 2015/11/05 13:14:46]

I went for int64_t, hope that's okay.

 #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_