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

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 139 matching lines @@
                const webrtc::StreamConfig& reverse_input_config,
                const webrtc::StreamConfig& reverse_output_config,
                float* const* dest));
   WEBRTC_STUB(set_stream_delay_ms, (int delay));
   WEBRTC_STUB_CONST(stream_delay_ms, ());
   WEBRTC_BOOL_STUB_CONST(was_stream_delay_set, ());
   WEBRTC_VOID_STUB(set_stream_key_pressed, (bool key_pressed));
   WEBRTC_VOID_STUB(set_delay_offset_ms, (int offset));
   WEBRTC_STUB_CONST(delay_offset_ms, ());
   WEBRTC_STUB(StartDebugRecording, (const char filename[kMaxFilenameSize]));
+  WEBRTC_STUB(StartDebugRecording,
+              (const char filename[kMaxFilenameSize], int max_log_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle, int max_log_size_bytes));
   WEBRTC_STUB(StopDebugRecording, ());
   WEBRTC_VOID_STUB(UpdateHistogramsOnCallEnd, ());
   webrtc::EchoCancellation* echo_cancellation() const override { return NULL; }
   webrtc::EchoControlMobile* echo_control_mobile() const override {
     return NULL;
   }
   webrtc::GainControl* gain_control() const override { return NULL; }
   webrtc::HighPassFilter* high_pass_filter() const override { return NULL; }
   webrtc::LevelEstimator* level_estimator() const override { return NULL; }
   webrtc::NoiseSuppression* noise_suppression() const override { return NULL; }
@@ skipping 826 matching lines @@
   }
   WEBRTC_FUNC(GetEcMetricsStatus, (bool& enabled)) {
     enabled = ec_metrics_enabled_;
     return 0;
   }
   WEBRTC_STUB(GetEchoMetrics, (int& ERL, int& ERLE, int& RERL, int& A_NLP));
   WEBRTC_STUB(GetEcDelayMetrics, (int& delay_median, int& delay_std,
       float& fraction_poor_delays));
 
   WEBRTC_STUB(StartDebugRecording, (const char* fileNameUTF8));
+  WEBRTC_STUB(StartDebugRecording,
+              (const char* fileNameUTF8, int max_size_bytes));
   WEBRTC_STUB(StartDebugRecording, (FILE* handle));
+  WEBRTC_STUB(StartDebugRecording, (FILE * handle, int max_size_bytes));
   WEBRTC_STUB(StopDebugRecording, ());
 
   WEBRTC_FUNC(SetTypingDetectionStatus, (bool enable)) {
     typing_detection_enabled_ = enable;
     return 0;
   }
   WEBRTC_FUNC(GetTypingDetectionStatus, (bool& enabled)) {
     enabled = typing_detection_enabled_;
     return 0;
   }
@@ skipping 100 matching lines @@
   int playout_sample_rate_;
   DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_