Implement the NackModule as part of the new jitter buffer.

webrtc/modules/video_coding/histogram.cc

+/*
+ *  Copyright (c) 2016 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/video_coding/histogram.h"
+
+#include <algorithm>
+
+#include "webrtc/base/mod_ops.h"
+
+namespace webrtc {
+namespace video_coding {
+Histogram::Histogram(size_t num_buckets, size_t max_num_values) {
+  RTC_DCHECK_LE(0u, num_buckets);
+  RTC_DCHECK_LE(0u, max_num_values);
+  buckets_.resize(num_buckets);
+  values_.reserve(max_num_values);
+  index_ = 0;
+}
+
+void Histogram::Add(size_t value) {
+  RTC_DCHECK_GE(value, 0u);
+  value = std::min<size_t>(value, buckets_.size() - 1);
+  if (index_ < values_.size()) {
+    --buckets_[values_[index_]];
+    RTC_DCHECK_LT(values_[index_], buckets_.size());
+    values_[index_] = value;
+  } else {
+    values_.emplace_back(value);
+  }
+
+  ++buckets_[value];
+  index_ = (index_ + 1) % values_.capacity();
+}
+
+size_t Histogram::InverseCdf(float probability) const {
+  RTC_DCHECK_GE(probability, 0.f);
+  RTC_DCHECK_LE(probability, 1.f);
+  RTC_DCHECK_GT(values_.size(), 0ul);
+
+  size_t bucket = 0;
+  float accumulated_probability = 0;
+  while (accumulated_probability < probability && bucket < buckets_.size()) {
+    accumulated_probability +=
+        static_cast<float>(buckets_[bucket]) / values_.size();
+    ++bucket;
+  }
+  return bucket;
+}
+
+size_t Histogram::NumValues() const {
+  return values_.size();
+}
+
+}  // namespace video_coding
+}  // namespace webrtc