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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
81 lines
2.9 KiB
C++
81 lines
2.9 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
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#define MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
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#include <stdint.h>
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#include "modules/include/module_common_types.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "rtc_base/criticalsection.h"
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#include "rtc_base/thread_annotations.h"
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namespace webrtc {
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// This class tracks the application requests to limit minimum and maximum
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// playout delay and makes a decision on whether the current RTP frame
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// should include the playout out delay extension header.
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//
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// Playout delay can be defined in terms of capture and render time as follows:
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//
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// Render time = Capture time in receiver time + playout delay
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//
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// The application specifies a minimum and maximum limit for the playout delay
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// which are both communicated to the receiver and the receiver can adapt
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// the playout delay within this range based on observed network jitter.
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class PlayoutDelayOracle {
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public:
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PlayoutDelayOracle();
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~PlayoutDelayOracle();
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// Returns true if the current frame should include the playout delay
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// extension
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bool send_playout_delay() const {
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rtc::CritScope lock(&crit_sect_);
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return send_playout_delay_;
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}
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// Returns current playout delay.
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PlayoutDelay playout_delay() const {
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rtc::CritScope lock(&crit_sect_);
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return playout_delay_;
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}
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// Updates the application requested playout delay, current ssrc
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// and the current sequence number.
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void UpdateRequest(uint32_t ssrc,
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PlayoutDelay playout_delay,
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uint16_t seq_num);
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void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
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private:
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// The playout delay information is updated from the encoder thread(s).
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// The sequence number feedback is updated from the worker thread.
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// Guards access to data across multiple threads.
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rtc::CriticalSection crit_sect_;
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// The current highest sequence number on which playout delay has been sent.
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int64_t high_sequence_number_ RTC_GUARDED_BY(crit_sect_);
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// Indicates whether the playout delay should go on the next frame.
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bool send_playout_delay_ RTC_GUARDED_BY(crit_sect_);
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// Sender ssrc.
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uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
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// Sequence number unwrapper.
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SequenceNumberUnwrapper unwrapper_ RTC_GUARDED_BY(crit_sect_);
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// Playout delay values on the next frame if |send_playout_delay_| is set.
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PlayoutDelay playout_delay_ RTC_GUARDED_BY(crit_sect_);
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RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle);
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};
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} // namespace webrtc
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#endif // MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_
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