Added empty directory with myself as owner for new mixer.

webrtc/modules/audio_mixer/source/time_scheduler.cc

Index: webrtc/modules/audio_mixer/source/time_scheduler.cc
diff --git a/webrtc/modules/audio_mixer/source/time_scheduler.cc b/webrtc/modules/audio_mixer/source/time_scheduler.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b7c45ec66fcb521abd796f3387d1597edc89e6f7
--- /dev/null
+++ b/webrtc/modules/audio_mixer/source/time_scheduler.cc
@@ -0,0 +1,91 @@
+/*
+ *  Copyright (c) 2011 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/timeutils.h"
+#include "webrtc/modules/audio_conference_mixer/source/time_scheduler.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+
+namespace webrtc {
+TimeScheduler::TimeScheduler(const int64_t periodicityInMs)
+    : _crit(CriticalSectionWrapper::CreateCriticalSection()),
+      _isStarted(false),
+      _lastPeriodMark(),
+      _periodicityInMs(periodicityInMs),
+      _periodicityInTicks(periodicityInMs * rtc::kNumNanosecsPerMillisec),
+      _missedPeriods(0) {}
+
+TimeScheduler::~TimeScheduler() {
+  delete _crit;
+}
+
+int32_t TimeScheduler::UpdateScheduler() {
+  CriticalSectionScoped cs(_crit);
+  if (!_isStarted) {
+    _isStarted = true;
+    _lastPeriodMark = rtc::TimeNanos();
+    return 0;
+  }
+  // Don't perform any calculations until the debt of pending periods have
+  // been worked off.
+  if (_missedPeriods > 0) {
+    _missedPeriods--;
+    return 0;
+  }
+
+  // Calculate the time that has past since previous call to this function.
+  int64_t tickNow = rtc::TimeNanos();
+  int64_t amassedTicks = tickNow - _lastPeriodMark;
+  int64_t amassedMs = amassedTicks / rtc::kNumNanosecsPerMillisec;
+
+  // Calculate the number of periods the time that has passed correspond to.
+  int64_t periodsToClaim = amassedMs / _periodicityInMs;
+
+  // One period will be worked off by this call. Make sure that the number of
+  // pending periods don't end up being negative (e.g. if this function is
+  // called to often).
+  if (periodsToClaim < 1) {
+    periodsToClaim = 1;
+  }
+
+  // Update the last period mark without introducing any drifting.
+  // Note that if this fuunction is called to often _lastPeriodMark can
+  // refer to a time in the future which in turn will yield TimeToNextUpdate
+  // that is greater than the periodicity
+  for (int64_t i = 0; i < periodsToClaim; i++) {
+    _lastPeriodMark += _periodicityInTicks;
+  }
+
+  // Update the total amount of missed periods note that we have processed
+  // one period hence the - 1
+  _missedPeriods += periodsToClaim - 1;
+  return 0;
+}
+
+int32_t TimeScheduler::TimeToNextUpdate(int64_t& updateTimeInMS) const {
+  CriticalSectionScoped cs(_crit);
+  // Missed periods means that the next UpdateScheduler() should happen
+  // immediately.
+  if (_missedPeriods > 0) {
+    updateTimeInMS = 0;
+    return 0;
+  }
+
+  // Calculate the time (in ms) that has past since last call to
+  // UpdateScheduler()
+  int64_t tickNow = rtc::TimeNanos();
+  int64_t ticksSinceLastUpdate = tickNow - _lastPeriodMark;
+  const int64_t millisecondsSinceLastUpdate =
+      ticksSinceLastUpdate / rtc::kNumNanosecsPerMillisec;
+
+  updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate;
+  updateTimeInMS = (updateTimeInMS < 0) ? 0 : updateTimeInMS;
+  return 0;
+}
+}  // namespace webrtc