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* 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 <stdint.h>
#include "absl/types/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
// This class tracks the application requests to limit minimum and maximum
// playout delay and makes a decision on whether the current RTP frame
// should include the playout out delay extension header.
// Playout delay can be defined in terms of capture and render time as follows:
// Render time = Capture time in receiver time + playout delay
// The application specifies a minimum and maximum limit for the playout delay
// which are both communicated to the receiver and the receiver can adapt
// the playout delay within this range based on observed network jitter.
class PlayoutDelayOracle : public RtcpAckObserver {
~PlayoutDelayOracle() override;
// The playout delay to be added to a packet. The input delays are provided by
// the application, with -1 meaning unchanged/unspecified. The output delay
// are the values to be attached to packets on the wire. Presence and value
// depends on the current input, previous inputs, and received acks from the
// remote end.
absl::optional<PlayoutDelay> PlayoutDelayToSend(
PlayoutDelay requested_delay) const;
void OnSentPacket(uint16_t sequence_number,
absl::optional<PlayoutDelay> playout_delay);
void OnReceivedAck(int64_t extended_highest_sequence_number) override;
// The playout delay information is updated from the encoder thread(s).
// The sequence number feedback is updated from the worker thread.
// Guards access to data across multiple threads.
rtc::CriticalSection crit_sect_;
// The oldest sequence number on which the current playout delay values have
// been sent. When set, it means we need to attach extension to sent packets.
absl::optional<int64_t> unacked_sequence_number_ RTC_GUARDED_BY(crit_sect_);
// Sequence number unwrapper for sent packets.
// TODO(nisse): Could potentially get out of sync with the unwrapper used by
// the caller of OnReceivedAck.
SequenceNumberUnwrapper unwrapper_ RTC_GUARDED_BY(crit_sect_);
// Playout delay values on the next frame if |send_playout_delay_| is set.
PlayoutDelay latest_delay_ RTC_GUARDED_BY(crit_sect_) = {-1, -1};
} // namespace webrtc