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/base/event.cc

@@ -0,0 +1,135 @@
+/*
+ *  Copyright 2004 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/base/event.h"
+
+#if defined(WEBRTC_WIN)
+#include <windows.h>
+#elif defined(WEBRTC_POSIX)
+#include <pthread.h>
+#include <sys/time.h>
+#include <time.h>
+#else
+#error "Must define either WEBRTC_WIN or WEBRTC_POSIX."
+#endif
+
+#include "webrtc/base/checks.h"
+
+namespace rtc {
+
+#if defined(WEBRTC_WIN)
+
+Event::Event(bool manual_reset, bool initially_signaled) {
+  event_handle_ = ::CreateEvent(NULL,                 // Security attributes.
+                                manual_reset,
+                                initially_signaled,
+                                NULL);                // Name.
+  RTC_CHECK(event_handle_);
+}
+
+Event::~Event() {
+  CloseHandle(event_handle_);
+}
+
+void Event::Set() {
+  SetEvent(event_handle_);
+}
+
+void Event::Reset() {
+  ResetEvent(event_handle_);
+}
+
+bool Event::Wait(int milliseconds) {
+  DWORD ms = (milliseconds == kForever) ? INFINITE : milliseconds;
+  return (WaitForSingleObject(event_handle_, ms) == WAIT_OBJECT_0);
+}
+
+#elif defined(WEBRTC_POSIX)
+
+Event::Event(bool manual_reset, bool initially_signaled)
+    : is_manual_reset_(manual_reset),
+      event_status_(initially_signaled) {
+  RTC_CHECK(pthread_mutex_init(&event_mutex_, NULL) == 0);
+  RTC_CHECK(pthread_cond_init(&event_cond_, NULL) == 0);
+}
+
+Event::~Event() {
+  pthread_mutex_destroy(&event_mutex_);
+  pthread_cond_destroy(&event_cond_);
+}
+
+void Event::Set() {
+  pthread_mutex_lock(&event_mutex_);
+  event_status_ = true;
+  pthread_cond_broadcast(&event_cond_);
+  pthread_mutex_unlock(&event_mutex_);
+}
+
+void Event::Reset() {
+  pthread_mutex_lock(&event_mutex_);
+  event_status_ = false;
+  pthread_mutex_unlock(&event_mutex_);
+}
+
+bool Event::Wait(int milliseconds) {
+  pthread_mutex_lock(&event_mutex_);
+  int error = 0;
+
+  if (milliseconds != kForever) {
+    // Converting from seconds and microseconds (1e-6) plus
+    // milliseconds (1e-3) to seconds and nanoseconds (1e-9).
+
+    struct timespec ts;
+#if HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
+    // Use relative time version, which tends to be more efficient for
+    // pthread implementations where provided (like on Android).
+    ts.tv_sec = milliseconds / 1000;
+    ts.tv_nsec = (milliseconds % 1000) * 1000000;
+#else
+    struct timeval tv;
+    gettimeofday(&tv, NULL);
+
+    ts.tv_sec = tv.tv_sec + (milliseconds / 1000);
+    ts.tv_nsec = tv.tv_usec * 1000 + (milliseconds % 1000) * 1000000;
+
+    // Handle overflow.
+    if (ts.tv_nsec >= 1000000000) {
+      ts.tv_sec++;
+      ts.tv_nsec -= 1000000000;
+    }
+#endif
+
+    while (!event_status_ && error == 0) {
+#if HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE
+      error = pthread_cond_timedwait_relative_np(
+          &event_cond_, &event_mutex_, &ts);
+#else
+      error = pthread_cond_timedwait(&event_cond_, &event_mutex_, &ts);
+#endif
+    }
+  } else {
+    while (!event_status_ && error == 0)
+      error = pthread_cond_wait(&event_cond_, &event_mutex_);
+  }
+
+  // NOTE(liulk): Exactly one thread will auto-reset this event. All
+  // the other threads will think it's unsignaled.  This seems to be
+  // consistent with auto-reset events in WEBRTC_WIN
+  if (error == 0 && !is_manual_reset_)
+    event_status_ = false;
+
+  pthread_mutex_unlock(&event_mutex_);
+
+  return (error == 0);
+}
+
+#endif
+
+}  // namespace rtc