Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these

webrtc/modules/audio_processing/logging/aec_logging_file_handling.cc

@@ -0,0 +1,57 @@
+/*
+ *  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