Repo created

TMessagesProj/jni/voip/webrtc/modules/pacing/rtp_packet_pacer.h

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+/*
+ *  Copyright (c) 2019 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_PACING_RTP_PACKET_PACER_H_
+#define MODULES_PACING_RTP_PACKET_PACER_H_
+
+#include <stdint.h>
+
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/units/data_rate.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
+
+namespace webrtc {
+
+class RtpPacketPacer {
+ public:
+  virtual ~RtpPacketPacer() = default;
+
+  virtual void CreateProbeClusters(
+      std::vector<ProbeClusterConfig> probe_cluster_configs) = 0;
+
+  // Temporarily pause all sending.
+  virtual void Pause() = 0;
+
+  // Resume sending packets.
+  virtual void Resume() = 0;
+
+  virtual void SetCongested(bool congested) = 0;
+
+  // Sets the pacing rates. Must be called once before packets can be sent.
+  virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0;
+
+  // Time since the oldest packet currently in the queue was added.
+  virtual TimeDelta OldestPacketWaitTime() const = 0;
+
+  // Sum of payload + padding bytes of all packets currently in the pacer queue.
+  virtual DataSize QueueSizeData() const = 0;
+
+  // Returns the time when the first packet was sent.
+  virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
+
+  // Returns the expected number of milliseconds it will take to send the
+  // current packets in the queue, given the current size and bitrate, ignoring
+  // priority.
+  virtual TimeDelta ExpectedQueueTime() const = 0;
+
+  // Set the average upper bound on pacer queuing delay. The pacer may send at
+  // a higher rate than what was configured via SetPacingRates() in order to
+  // keep ExpectedQueueTimeMs() below `limit_ms` on average.
+  virtual void SetQueueTimeLimit(TimeDelta limit) = 0;
+
+  // Currently audio traffic is not accounted by pacer and passed through.
+  // With the introduction of audio BWE audio traffic will be accounted for
+  // the pacer budget calculation. The audio traffic still will be injected
+  // at high priority.
+  virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
+  virtual void SetIncludeOverhead() = 0;
+  virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_PACING_RTP_PACKET_PACER_H_