mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2024-10-31 00:37:39 +08:00
增加 RtcpContextForSend/RtcpContextForRecv作为RtcpContext子类
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290b3f37a5
commit
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@ -14,12 +14,16 @@ using namespace toolkit;
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namespace mediakit {
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RtcpContext::RtcpContext(bool is_receiver) {
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_is_receiver = is_receiver;
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}
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void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) {
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if (_is_receiver) {
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++_packets;
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_bytes += bytes;
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_last_rtp_stamp = stamp;
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_last_ntp_stamp_ms = ntp_stamp_ms;
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}
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void RtcpContextForRecv::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) {
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{
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//接收者才做复杂的统计运算
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auto sys_stamp = getCurrentMillisecond();
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if (_last_rtp_sys_stamp) {
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@ -57,11 +61,7 @@ void RtcpContext::onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uin
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_last_rtp_seq = seq;
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_last_rtp_sys_stamp = sys_stamp;
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}
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++_packets;
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_bytes += bytes;
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_last_rtp_stamp = stamp;
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_last_ntp_stamp_ms = ntp_stamp_ms;
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RtcpContext::onRtp(seq, stamp, ntp_stamp_ms, sample_rate, bytes);
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}
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void RtcpContext::onRtcp(RtcpHeader *rtcp) {
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@ -115,9 +115,9 @@ uint32_t RtcpContext::getRtt(uint32_t ssrc) const {
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}
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size_t RtcpContext::getExpectedPackets() const {
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if (!_is_receiver) {
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throw std::runtime_error("rtp发送者无法统计应收包数");
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}
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throw std::runtime_error("没有实现, rtp发送者无法统计应收包数");
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}
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size_t RtcpContextForRecv::getExpectedPackets() const {
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return (_seq_cycles << 16) + _seq_max - _seq_base + 1;
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}
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@ -129,9 +129,10 @@ size_t RtcpContext::getExpectedPacketsInterval() {
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}
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size_t RtcpContext::getLost() {
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if (!_is_receiver) {
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throw std::runtime_error("rtp发送者无法统计丢包率");
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}
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throw std::runtime_error("没有实现, rtp发送者无法统计丢包率");
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}
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size_t RtcpContextForRecv::getLost() {
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return getExpectedPackets() - _packets;
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}
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@ -143,9 +144,10 @@ size_t RtcpContext::geLostInterval() {
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}
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Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
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if (_is_receiver) {
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throw std::runtime_error("rtp接收者尝试发送sr包");
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}
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throw std::runtime_error("没有实现, rtp接收者尝试发送sr包");
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}
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Buffer::Ptr RtcpContextForSend::createRtcpSR(uint32_t rtcp_ssrc) {
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auto rtcp = RtcpSR::create(0);
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rtcp->setNtpStamp(_last_ntp_stamp_ms);
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rtcp->rtpts = htonl(_last_rtp_stamp);
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@ -165,9 +167,11 @@ Buffer::Ptr RtcpContext::createRtcpSR(uint32_t rtcp_ssrc) {
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}
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Buffer::Ptr RtcpContext::createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) {
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if (!_is_receiver) {
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throw std::runtime_error("rtp发送者尝试发送rr包");
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}
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throw std::runtime_error("没有实现, rtp发送者尝试发送rr包");
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}
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Buffer::Ptr RtcpContextForRecv::createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) {
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auto rtcp = RtcpRR::create(1);
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rtcp->ssrc = htonl(rtcp_ssrc);
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@ -20,11 +20,7 @@ namespace mediakit {
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class RtcpContext {
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public:
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using Ptr = std::shared_ptr<RtcpContext>;
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/**
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* 创建rtcp上下文
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* @param is_receiver 是否为rtp接收者,接收者更消耗性能
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*/
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RtcpContext(bool is_receiver);
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virtual ~RtcpContext() = default;
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/**
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* 输出或输入rtp时调用
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@ -34,7 +30,7 @@ public:
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* @param rtp rtp时间戳采样率,视频一般为90000,音频一般为采样率
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* @param bytes rtp数据长度
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*/
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void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
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virtual void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes);
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/**
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* 输入sr rtcp包
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@ -45,19 +41,19 @@ public:
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/**
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* 计算总丢包数
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*/
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size_t getLost();
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virtual size_t getLost();
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/**
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* 返回理应收到的rtp数
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*/
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size_t getExpectedPackets() const;
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virtual size_t getExpectedPackets() const;
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/**
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* 创建SR rtcp包
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* @param rtcp_ssrc rtcp的ssrc
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* @return rtcp包
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*/
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Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
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virtual Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc);
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/**
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* 创建RR rtcp包
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@ -65,7 +61,7 @@ public:
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* @param rtp_ssrc rtp的ssrc
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* @return rtcp包
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*/
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Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
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virtual Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc);
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/**
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* 获取rtt
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@ -84,9 +80,7 @@ public:
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*/
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size_t geLostInterval();
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private:
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//是否为接收者
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bool _is_receiver;
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protected:
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//时间戳抖动值
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double _jitter = 0;
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//收到或发送的rtp的字节数
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@ -120,5 +114,17 @@ private:
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map<uint32_t/*last_sr_lsr*/, uint64_t/*ntp stamp*/> _sender_report_ntp;
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};
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class RtcpContextForSend : public RtcpContext {
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public:
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Buffer::Ptr createRtcpSR(uint32_t rtcp_ssrc) override;
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};
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class RtcpContextForRecv : public RtcpContext {
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public:
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void onRtp(uint16_t seq, uint32_t stamp, uint64_t ntp_stamp_ms, uint32_t sample_rate, size_t bytes) override;
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Buffer::Ptr createRtcpRR(uint32_t rtcp_ssrc, uint32_t rtp_ssrc) override;
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size_t getExpectedPackets() const override;
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size_t getLost() override;
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};
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}//namespace mediakit
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#endif //ZLMEDIAKIT_RTCPCONTEXT_H
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@ -23,11 +23,11 @@ RtpServer::~RtpServer() {
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}
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}
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class RtcpHelper : public RtcpContext, public std::enable_shared_from_this<RtcpHelper> {
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class RtcpHelper : public RtcpContextForRecv, public std::enable_shared_from_this<RtcpHelper> {
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public:
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using Ptr = std::shared_ptr<RtcpHelper>;
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RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) : RtcpContext(true){
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RtcpHelper(Socket::Ptr rtcp_sock, uint32_t sample_rate) {
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_rtcp_sock = std::move(rtcp_sock);
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_sample_rate = sample_rate;
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}
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@ -206,7 +206,7 @@ void RtspPlayer::handleResDESCRIBE(const Parser& parser) {
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}
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_rtcp_context.clear();
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for (auto &track : _sdp_track) {
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_rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
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_rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
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}
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sendSetup(0);
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}
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@ -179,7 +179,7 @@ void RtspPusher::sendAnnounce() {
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}
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_rtcp_context.clear();
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for (auto &track : _track_vec) {
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_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
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_rtcp_context.emplace_back(std::make_shared<RtcpContextForSend>());
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}
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_on_res_func = std::bind(&RtspPusher::handleResAnnounce, this, placeholders::_1);
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sendRtspRequest("ANNOUNCE", _url, {}, src->getSdp());
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@ -257,7 +257,7 @@ void RtspSession::handleReq_ANNOUNCE(const Parser &parser) {
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}
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_rtcp_context.clear();
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for (auto &track : _sdp_track) {
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_rtcp_context.emplace_back(std::make_shared<RtcpContext>(true));
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_rtcp_context.emplace_back(std::make_shared<RtcpContextForRecv>());
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}
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_push_src = std::make_shared<RtspMediaSourceImp>(_media_info._vhost, _media_info._app, _media_info._streamid);
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_push_src->setListener(dynamic_pointer_cast<MediaSourceEvent>(shared_from_this()));
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@ -418,7 +418,7 @@ void RtspSession::onAuthSuccess() {
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}
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strongSelf->_rtcp_context.clear();
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for (auto &track : strongSelf->_sdp_track) {
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strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContext>(false));
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strongSelf->_rtcp_context.emplace_back(std::make_shared<RtcpContextForSend>());
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}
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strongSelf->_sessionid = makeRandStr(12);
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strongSelf->_play_src = rtsp_src;
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@ -420,7 +420,7 @@ void WebRtcTransportImp::onStartWebRTC() {
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track->offer_ssrc_rtx = m_offer->getRtxSSRC();
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track->plan_rtp = &m_answer.plan[0];;
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track->plan_rtx = m_answer.getRelatedRtxPlan(track->plan_rtp->pt);
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track->rtcp_context_send = std::make_shared<RtcpContext>(false);
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track->rtcp_context_send = std::make_shared<RtcpContextForSend>();
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//send ssrc --> MediaTrack
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_ssrc_to_track[track->answer_ssrc_rtp] = track;
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@ -656,7 +656,7 @@ private:
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private:
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NackContext _nack_ctx;
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RtcpContext _rtcp_context{true};
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RtcpContextForRecv _rtcp_context;
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EventPoller::Ptr _poller;
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DelayTask::Ptr _delay_task;
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function<void(const FCI_NACK &nack)> _on_nack;
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