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

webrtc/modules/audio_processing/logging/aec_logging.h

+/*
+ *  Copyright (c) 2013 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_
+
+
+// To enable AEC logging, run this command from trunk/ :
+// python webrtc/build/gyp_webrtc.py -Daec_debug_dump=1
+#ifdef WEBRTC_AEC_DEBUG_DUMP
+#include <stdio.h>
+#include "webrtc/modules/audio_processing/logging/aec_logging_file_handling.h"
+#endif
+
+
+#ifndef WEBRTC_AEC_DEBUG_DUMP
+
+// Dump wav data to file
+#define WEBRTC_AEC_DEBUG_WAV_WRITE(file, data, numSamples)
+
+// (Re)open wav file for writing using the specified sample rate
+#define WEBRTC_AEC_DEBUG_WAV_REOPEN(wavFile, name, seq1, seq2, sampleRate)
+
+// Close wav file
+#define WEBRTC_AEC_DEBUG_WAV_CLOSE(wavFile)
+
+// Dump raw data to file
+#define WEBRTC_AEC_DEBUG_RAW_WRITE(file, data, dataSize)
+
+// Open raw data file for writing using the specified sample rate
+#define WEBRTC_AEC_DEBUG_RAW_OPEN(file, name, instanceCtr)
+
+// Close raw data file
+#define WEBRTC_AEC_DEBUG_RAW_CLOSE(file)
+
+#else  // WEBRTC_AEC_DEBUG_DUMP
+
+// Dump wav data to file
+#define WEBRTC_AEC_DEBUG_WAV_WRITE(file, data, numSamples)  \
+  do {                                                      \
+    rtc_WavWriteSamples(file, data, numSamples);            \
+  } while (0);
+
+// (Re)open wav file for writing using the specified sample rate
+#define WEBRTC_AEC_DEBUG_WAV_REOPEN(wavFile, name, seq1, seq2, sampleRate) \
+  do {                                                                  \
+    WebRtcAec_ReopenWav((wavFile), (name), (seq1), (seq2), (sampleRate)); \
+  } while (0);
+
+// Close wav file
+#define WEBRTC_AEC_DEBUG_WAV_CLOSE(wavFile)     \
+  do {                                          \
+    rtc_WavClose((wavFile));                    \
+  } while (0);
+
+// Dump raw data to file
+#define WEBRTC_AEC_DEBUG_RAW_WRITE(file, data, dataSize)  \
+  do {                                                    \
+    (void)fwrite((data), (dataSize) , 1, (file));         \
+  } while (0);
+
+// Open raw data file for writing using the specified sample rate
+#define WEBRTC_AEC_DEBUG_RAW_OPEN(file, name, instanceCtr)  \
+  do {                                                      \
+    WebRtcAec_RawFileOpen((file), (name), (instanceCtr));   \
+  } while (0);
+
+
+// Close raw data file
+#define WEBRTC_AEC_DEBUG_RAW_CLOSE(file)        \
+  do {                                          \
+    fclose((file));                             \
+  } while (0);
+
+
+
+#endif  // WEBRTC_AEC_DEBUG_DUMP
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_LOGGING_