| /* |
| * 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_ |
| |
| #include <stdint.h> |
| |
| #include "webrtc/modules/include/module_common_types.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/rtc_base/criticalsection.h" |
| #include "webrtc/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(); |
| |
| // 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); |
| |
| 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 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_); |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(PlayoutDelayOracle); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_PLAYOUT_DELAY_ORACLE_H_ |