summaryrefslogtreecommitdiffstats
path: root/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
diff options
context:
space:
mode:
Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h')
-rw-r--r--third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h50
1 files changed, 50 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
new file mode 100644
index 0000000000..b37ce18795
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/underrun_optimizer.h
@@ -0,0 +1,50 @@
+/*
+ * Copyright (c) 2021 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.
+ */
+
+#ifndef MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+#define MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_
+
+#include <memory>
+
+#include "absl/types/optional.h"
+#include "api/neteq/tick_timer.h"
+#include "modules/audio_coding/neteq/histogram.h"
+
+namespace webrtc {
+
+// Estimates probability of buffer underrun due to late packet arrival.
+// The optimal delay is decided such that the probability of underrun is lower
+// than 1 - `histogram_quantile`.
+class UnderrunOptimizer {
+ public:
+ UnderrunOptimizer(const TickTimer* tick_timer,
+ int histogram_quantile,
+ int forget_factor,
+ absl::optional<int> start_forget_weight,
+ absl::optional<int> resample_interval_ms);
+
+ void Update(int relative_delay_ms);
+
+ absl::optional<int> GetOptimalDelayMs() const { return optimal_delay_ms_; }
+
+ void Reset();
+
+ private:
+ const TickTimer* tick_timer_;
+ Histogram histogram_;
+ const int histogram_quantile_; // In Q30.
+ const absl::optional<int> resample_interval_ms_;
+ std::unique_ptr<TickTimer::Stopwatch> resample_stopwatch_;
+ int max_delay_in_interval_ms_ = 0;
+ absl::optional<int> optimal_delay_ms_;
+};
+
+} // namespace webrtc
+#endif // MODULES_AUDIO_CODING_NETEQ_UNDERRUN_OPTIMIZER_H_