mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-20 00:57:49 +01:00
71f94c93a6
There's now one const method PlayoutDelayToSend to produce the delay values to insert into outgoing packets, and two update methods, OnSentPacket, and OnReceivedAck, to observe outgoing packets and acks, respectively. Bug: webrtc:7135 Change-Id: I07498c30f0de87ae0113f7e2eb6357a091a1f0af Reviewed-on: https://webrtc-review.googlesource.com/c/120603 Commit-Queue: Niels Moller <nisse@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26474}
75 lines
3 KiB
C++
75 lines
3 KiB
C++
/*
|
|
* 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.
|
|
*/
|
|
|
|
#ifndef MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
|
|
#define MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
|
|
|
|
#include <stdint.h>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "common_types.h" // NOLINT(build/include)
|
|
#include "modules/include/module_common_types_public.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:
|
|
PlayoutDelayOracle();
|
|
~PlayoutDelayOracle();
|
|
|
|
// 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);
|
|
|
|
private:
|
|
// 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};
|
|
|
|
RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle);
|
|
};
|
|
|
|
} // namespace webrtc
|
|
|
|
#endif // MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
|