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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 "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.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_ GUARDED_BY(crit_sect_);
// Indicates whether the playout delay should go on the next frame.
bool send_playout_delay_ GUARDED_BY(crit_sect_);
// Sender ssrc.
uint32_t ssrc_ GUARDED_BY(crit_sect_);
// Sequence number unwrapper.
SequenceNumberUnwrapper unwrapper_ GUARDED_BY(crit_sect_);
// Playout delay values on the next frame if |send_playout_delay_| is set.
PlayoutDelay playout_delay_ GUARDED_BY(crit_sect_);
} // namespace webrtc