1 files changed, 64 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
new file mode 100644
index 0000000000..2c42d0d13f
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2012 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 "modules/audio_coding/neteq/buffer_level_filter.h"
+
+#include <stdint.h>
+
+#include <algorithm>
+
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+BufferLevelFilter::BufferLevelFilter() {
+  Reset();
+}
+
+void BufferLevelFilter::Reset() {
+  filtered_current_level_ = 0;
+  level_factor_ = 253;
+}
+
+void BufferLevelFilter::Update(size_t buffer_size_samples,
+                               int time_stretched_samples) {
+  // Filter:
+  // `filtered_current_level_` = `level_factor_` * `filtered_current_level_` +
+  //                            (1 - `level_factor_`) * `buffer_size_samples`
+  // `level_factor_` and `filtered_current_level_` are in Q8.
+  // `buffer_size_samples` is in Q0.
+  const int64_t filtered_current_level =
+      (level_factor_ * int64_t{filtered_current_level_} >> 8) +
+      (256 - level_factor_) * rtc::dchecked_cast<int64_t>(buffer_size_samples);
+
+  // Account for time-scale operations (accelerate and pre-emptive expand) and
+  // make sure that the filtered value remains non-negative.
+  filtered_current_level_ = rtc::saturated_cast<int>(std::max<int64_t>(
+      0, filtered_current_level - int64_t{time_stretched_samples} * (1 << 8)));
+}
+
+void BufferLevelFilter::SetFilteredBufferLevel(int buffer_size_samples) {
+  filtered_current_level_ =
+      rtc::saturated_cast<int>(int64_t{buffer_size_samples} * 256);
+}
+
+void BufferLevelFilter::SetTargetBufferLevel(int target_buffer_level_ms) {
+  if (target_buffer_level_ms <= 20) {
+    level_factor_ = 251;
+  } else if (target_buffer_level_ms <= 60) {
+    level_factor_ = 252;
+  } else if (target_buffer_level_ms <= 140) {
+    level_factor_ = 253;
+  } else {
+    level_factor_ = 254;
+  }
+}
+
+}  // namespace webrtc