/* * Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved. * * This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit). * * Use of this source code is governed by MIT-like license that can be found in the * LICENSE file in the root of the source tree. All contributing project authors * may be found in the AUTHORS file in the root of the source tree. */ #ifndef ZLMEDIAKIT_RTCPCONTEXT_H #define ZLMEDIAKIT_RTCPCONTEXT_H #include "Rtcp.h" #include #include namespace mediakit { class RtcpContext { public: using Ptr = std::shared_ptr; virtual ~RtcpContext() = default; /** * 输出或输入rtp时调用 * @param seq rtp的seq * @param stamp rtp的时间戳，单位采样数(非毫秒) * @param ntp_stamp_ms ntp时间戳 * @param rtp rtp时间戳采样率，视频一般为90000，音频一般为采样率 * @param bytes rtp数据长度 * Called when outputting or inputting rtp * @param seq rtp's seq * @param stamp rtp's timestamp, unit is sample number (not millisecond) * @param ntp_stamp_ms ntp timestamp * @param rtp rtp timestamp sampling rate, video is generally 90000, audio is generally sampling rate * @param bytes rtp data length * [AUTO-TRANSLATED:745772b5] */ virtual void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes); /** * 输入sr rtcp包 * @param rtcp 输入一个rtcp * Input sr rtcp packet * @param rtcp input an rtcp * [AUTO-TRANSLATED:46f309ec] */ virtual void onRtcp(RtcpHeader *rtcp) = 0; /** * 计算总丢包数 * Calculate the total number of lost packets * [AUTO-TRANSLATED:084f3832] */ virtual size_t getLost(); /** * 返回理应收到的rtp数 * Return the number of rtp that should be received * [AUTO-TRANSLATED:ede367a0] */ virtual size_t getExpectedPackets() const; /** * 创建SR rtcp包 * @param rtcp_ssrc rtcp的ssrc * @return rtcp包 * Create SR rtcp packet * @param rtcp_ssrc rtcp's ssrc * @return rtcp packet * [AUTO-TRANSLATED:a9ec36d0] */ virtual toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc); /** * @brief 创建xr的dlrr包，用于接收者估算rtt * * @return toolkit::Buffer::Ptr * @brief Create xr's dlrr packet, used by receiver to estimate rtt * * @return toolkit::Buffer::Ptr * [AUTO-TRANSLATED:a5094e1d] */ virtual toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc); /** * 创建RR rtcp包 * @param rtcp_ssrc rtcp的ssrc * @param rtp_ssrc rtp的ssrc * @return rtcp包 * Create RR rtcp packet * @param rtcp_ssrc rtcp's ssrc * @param rtp_ssrc rtp's ssrc * @return rtcp packet * [AUTO-TRANSLATED:81ebbf81] */ virtual toolkit::Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc); /** * 上次结果与本次结果间应收包数 * Number of packets that should be received between the last result and the current result * [AUTO-TRANSLATED:3b2846ab] */ virtual size_t getExpectedPacketsInterval(); /** * 上次结果与本次结果间丢包个数 * Number of lost packets between the last result and the current result * [AUTO-TRANSLATED:fe5ac890] */ virtual size_t getLostInterval(); protected: // 收到或发送的rtp的字节数 [AUTO-TRANSLATED:a38d88a9] // Number of bytes of rtp received or sent size_t _bytes = 0; // 收到或发送的rtp的个数 [AUTO-TRANSLATED:b28c3c90] // Number of rtp received or sent size_t _packets = 0; // 上次的rtp时间戳,毫秒 [AUTO-TRANSLATED:99eecec6] // Last rtp timestamp, milliseconds uint32_t _last_rtp_stamp = 0; uint64_t _last_ntp_stamp_ms = 0; }; class RtcpContextForSend : public RtcpContext { public: toolkit::Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc) override; void onRtcp(RtcpHeader *rtcp) override; toolkit::Buffer::Ptr createRtcpXRDLRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override; /** * 获取rtt * @param ssrc rtp ssrc * @return rtt,单位毫秒 * Get rtt * @param ssrc rtp ssrc * @return rtt, unit is millisecond * [AUTO-TRANSLATED:f0885551] */ uint32_t getRtt(uint32_t ssrc) const; private: std::map _rtt; std::map _sender_report_ntp; std::map _xr_rrtr_recv_sys_stamp; std::map _xr_xrrtr_recv_last_rr; }; class RtcpContextForRecv : public RtcpContext { public: void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) override; toolkit::Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override; size_t getExpectedPackets() const override; size_t getExpectedPacketsInterval() override; size_t getLost() override; size_t getLostInterval() override; void onRtcp(RtcpHeader *rtcp) override; private: // 时间戳抖动值 [AUTO-TRANSLATED:8100680c] // Timestamp jitter value double _jitter = 0; // 第一个seq的值 [AUTO-TRANSLATED:d893719d] // The value of the first seq uint16_t _seq_base = 0; // rtp最大seq [AUTO-TRANSLATED:5cc9f775] // Maximum rtp seq uint16_t _seq_max = 0; // rtp回环次数 [AUTO-TRANSLATED:9fe9c340] // Rtp loopback times uint16_t _seq_cycles = 0; // 上次回环发生时，记录的rtp包数 [AUTO-TRANSLATED:c32cb555] // Number of rtp packets recorded when the last loopback occurred size_t _last_cycle_packets = 0; // 上次的seq [AUTO-TRANSLATED:07364b7d] // Last seq uint16_t _last_rtp_seq = 0; // 上次的rtp的系统时间戳(毫秒)用于统计抖动 [AUTO-TRANSLATED:b1e8c89b] // Last rtp system timestamp (milliseconds) used for jitter statistics uint64_t _last_rtp_sys_stamp = 0; // 上次统计的丢包总数 [AUTO-TRANSLATED:242e75ed] // Last total number of lost packets counted size_t _last_lost = 0; // 上次统计应收rtp包总数 [AUTO-TRANSLATED:eb2d5f4d] // Last total number of rtp packets that should be received counted size_t _last_expected = 0; // 上次收到sr包时计算出的Last SR timestamp [AUTO-TRANSLATED:fdec069e] // Last SR timestamp calculated when the last SR packet was received uint32_t _last_sr_lsr = 0; // 上次收到sr时的系统时间戳,单位毫秒 [AUTO-TRANSLATED:044fa0d5] // System timestamp when the last SR was received, unit is millisecond uint64_t _last_sr_ntp_sys = 0; }; } // namespace mediakit #endif // ZLMEDIAKIT_RTCPCONTEXT_H