blob: 0e3bd39a7ef5ee1ae85dff256abbb7943268ee25 [file] [log] [blame]
* 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 "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/constructormagic.h"
#include "rtc_base/criticalsection.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 {
// Returns true if the current frame should include the playout delay
// extension
bool send_playout_delay() const {
rtc::CritScope lock(&crit_sect_);
return send_playout_delay_;
// Returns current playout delay.
PlayoutDelay playout_delay() const {
rtc::CritScope lock(&crit_sect_);
return playout_delay_;
// Updates the application requested playout delay, current ssrc
// and the current sequence number.
void UpdateRequest(uint32_t ssrc,
PlayoutDelay playout_delay,
uint16_t seq_num);
void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
// 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 current highest sequence number on which playout delay has been sent.
int64_t high_sequence_number_ RTC_GUARDED_BY(crit_sect_);
// Indicates whether the playout delay should go on the next frame.
bool send_playout_delay_ RTC_GUARDED_BY(crit_sect_);
// Sender ssrc.
uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
// Sequence number unwrapper.
SequenceNumberUnwrapper unwrapper_ RTC_GUARDED_BY(crit_sect_);
// Playout delay values on the next frame if |send_playout_delay_| is set.
PlayoutDelay playout_delay_ RTC_GUARDED_BY(crit_sect_);
} // namespace webrtc