Replaced the wav file dumping functionality in aec_core.c with the newly added corresponding macros

webrtc/modules/audio_processing/logging/aec_logging.h

Index: webrtc/modules/audio_processing/logging/aec_logging.h
diff --git a/webrtc/modules/audio_processing/logging/aec_logging.h b/webrtc/modules/audio_processing/logging/aec_logging.h
new file mode 100644
index 0000000000000000000000000000000000000000..c6c6cbf0438a41a214820575ee616ff9e44bd0a0
--- /dev/null
+++ b/webrtc/modules/audio_processing/logging/aec_logging.h
@@ -0,0 +1,86 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_
+
+#include <stdio.h>
+
+#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
+
+// To enable AEC logging, invoke GYP with -Daec_debug_dump=1.
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+// Dumps a wav data to file.
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+    rtc_WavWriteSamples((file), (data), (num_samples));  \
+  } while (0)
+
+// (Re)opens a wav file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_WAV_REOPEN(name, instance_index, process_rate, \
+                                 sample_rate, wav_file)              \
+  do {                                                               \
+    WebRtcAec_ReopenWav((name), (instance_index), (process_rate),    \
+                        (sample_rate), (wav_file));                  \
+  } while (0)
+
+// Closes a wav file.
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+    rtc_WavClose((wav_file));             \
+  } while (0)
+
+// Dumps a raw data to file.
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+    (void) fwrite((data), (data_size), 1, (file));     \
+  } while (0)
+
+// Opens a raw data file for writing using the specified sample rate.
+#define RTC_AEC_DEBUG_RAW_OPEN(name, instance_counter, file)   \
+  do {                                                         \
+    WebRtcAec_RawFileOpen((name), (instance_counter), (file)); \
+  } while (0)
+
+// Closes a raw data file.
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+    fclose((file));                   \
+  } while (0)
+
+#else  // RTC_AEC_DEBUG_DUMP
+#define RTC_AEC_DEBUG_WAV_WRITE(file, data, num_samples) \
+  do {                                                   \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_REOPEN(wav_file, name, instance_index, process_rate, \
+                                 sample_rate)                                  \
+  do {                                                                         \
+  } while (0)
+
+#define RTC_AEC_DEBUG_WAV_CLOSE(wav_file) \
+  do {                                    \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_WRITE(file, data, data_size) \
+  do {                                                 \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_OPEN(file, name, instance_counter) \
+  do {                                                       \
+  } while (0)
+
+#define RTC_AEC_DEBUG_RAW_CLOSE(file) \
+  do {                                \
+  } while (0)
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_