mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-14 06:10:40 +01:00
5fe9510efb
The idea is to let the RtpRtcp and RTPSender classes be responsible for media-agnostic RTP transport, and move out the media-specific processing, such as packetization and media-specific headers. Bug: webrtc:7135 Change-Id: Ib0ce45bf06713b3eb6c06acd91c5168856874e4e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123187 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26954}
76 lines
3.1 KiB
C++
76 lines
3.1 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 "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 {
|
|
public:
|
|
PlayoutDelayOracle();
|
|
~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;
|
|
|
|
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_
|