Replaced the data logging functionality in the AEC with a generic logging functionality

webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc

Index: webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
diff --git a/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc b/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
deleted file mode 100644
index 3a434714e1e754bd6fb6c7b2dfd850dac37cf735..0000000000000000000000000000000000000000
--- a/webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- *  Copyright (c) 2015 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.
- */
-
-#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
-
-#include <stdint.h>
-#include <stdio.h>
-
-#include "webrtc/base/checks.h"
-#include "webrtc/base/stringutils.h"
-#include "webrtc/common_audio/wav_file.h"
-#include "webrtc/typedefs.h"
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-void WebRtcAec_ReopenWav(const char* name,
-                         int instance_index,
-                         int process_rate,
-                         int sample_rate,
-                         rtc_WavWriter** wav_file) {
-  if (*wav_file) {
-    if (rtc_WavSampleRate(*wav_file) == sample_rate)
-      return;
-    rtc_WavClose(*wav_file);
-  }
-  char filename[64];
-  int written = rtc::sprintfn(filename, sizeof(filename), "%s%d-%d.wav", name,
-                              instance_index, process_rate);
-
-  // Ensure there was no buffer output error.
-  RTC_DCHECK_GE(written, 0);
-  // Ensure that the buffer size was sufficient.
-  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
-
-  *wav_file = rtc_WavOpen(filename, sample_rate, 1);
-}
-
-void WebRtcAec_RawFileOpen(const char* name, int instance_index, FILE** file) {
-  char filename[64];
-  int written = rtc::sprintfn(filename, sizeof(filename), "%s_%d.dat", name,
-                              instance_index);
-
-  // Ensure there was no buffer output error.
-  RTC_DCHECK_GE(written, 0);
-  // Ensure that the buffer size was sufficient.
-  RTC_DCHECK_LT(static_cast<size_t>(written), sizeof(filename));
-
-  *file = fopen(filename, "wb");
-}
-
-#endif  // WEBRTC_AEC_DEBUG_DUMP