mirror of
https://gitee.com/xia-chu/ZLMediaKit.git
synced 2026-05-06 19:07:49 +08:00
81aef25583
startSendPassive接口指定is_udp为1时则为开启udp被动模式, 被动模式下需要对方先发数据给zlmediakit，然后才能回复rtp数据； 同时在被动模式指定src_port时，不再等待对方连接成功后再返回(而是直接返回成功和端口号，失败通知请监听on_send_rtp_stopped hook)。
470 lines
17 KiB
C++
470 lines
17 KiB
C++
/*
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* Copyright (c) 2016-present The ZLMediaKit project authors. All Rights Reserved.
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*
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* This file is part of ZLMediaKit(https://github.com/ZLMediaKit/ZLMediaKit).
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*
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* Use of this source code is governed by MIT-like license that can be found in the
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* LICENSE file in the root of the source tree. All contributing project authors
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* may be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef ZLMEDIAKIT_MEDIASOURCE_H
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#define ZLMEDIAKIT_MEDIASOURCE_H
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#include <string>
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#include <atomic>
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#include <memory>
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#include <functional>
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#include "Util/mini.h"
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#include "Network/Socket.h"
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#include "Extension/Track.h"
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#include "Record/Recorder.h"
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namespace toolkit {
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class Session;
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} // namespace toolkit
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namespace mediakit {
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enum class MediaOriginType : uint8_t {
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unknown = 0,
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rtmp_push ,
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rtsp_push,
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rtp_push,
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pull,
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ffmpeg_pull,
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mp4_vod,
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device_chn,
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rtc_push,
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srt_push
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};
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std::string getOriginTypeString(MediaOriginType type);
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class MediaSource;
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class RtpProcess;
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class MultiMediaSourceMuxer;
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class MediaSourceEvent {
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public:
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friend class MediaSource;
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class NotImplemented : public std::runtime_error {
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public:
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template<typename ...T>
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NotImplemented(T && ...args) : std::runtime_error(std::forward<T>(args)...) {}
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};
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virtual ~MediaSourceEvent() = default;
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// 获取媒体源类型
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virtual MediaOriginType getOriginType(MediaSource &sender) const { return MediaOriginType::unknown; }
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// 获取媒体源url或者文件路径
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virtual std::string getOriginUrl(MediaSource &sender) const;
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// 获取媒体源客户端相关信息
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virtual std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const { return nullptr; }
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// 通知拖动进度条
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virtual bool seekTo(MediaSource &sender, uint32_t stamp) { return false; }
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// 通知暂停或恢复
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virtual bool pause(MediaSource &sender, bool pause) { return false; }
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// 通知倍数
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virtual bool speed(MediaSource &sender, float speed) { return false; }
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// 通知其停止产生流
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virtual bool close(MediaSource &sender) { return false; }
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// 获取观看总人数，此函数一般强制重载
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virtual int totalReaderCount(MediaSource &sender) { throw NotImplemented(toolkit::demangle(typeid(*this).name()) + "::totalReaderCount not implemented"); }
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// 通知观看人数变化
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virtual void onReaderChanged(MediaSource &sender, int size);
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//流注册或注销事件
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virtual void onRegist(MediaSource &sender, bool regist) {}
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// 获取丢包率
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virtual float getLossRate(MediaSource &sender, TrackType type) { return -1; }
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// 获取所在线程, 此函数一般强制重载
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virtual toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) { throw NotImplemented(toolkit::demangle(typeid(*this).name()) + "::getOwnerPoller not implemented"); }
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////////////////////////仅供MultiMediaSourceMuxer对象继承////////////////////////
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// 开启或关闭录制
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virtual bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) { return false; };
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// 获取录制状态
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virtual bool isRecording(MediaSource &sender, Recorder::type type) { return false; }
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// 获取所有track相关信息
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virtual std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const { return std::vector<Track::Ptr>(); };
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// 获取MultiMediaSourceMuxer对象
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virtual std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) const { return nullptr; }
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// 获取RtpProcess对象
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virtual std::shared_ptr<RtpProcess> getRtpProcess(MediaSource &sender) const { return nullptr; }
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class SendRtpArgs {
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public:
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enum DataType {
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kRtpES = 0, // 发送ES流
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kRtpPS = 1, // 发送PS流
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kRtpTS = 2 // 发送TS流
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};
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enum ConType {
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kTcpActive = 0, // tcp主动模式，tcp客户端主动连接对方并发送rtp
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kUdpActive = 1, // udp主动模式，主动发送数据给对方
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kTcpPassive = 2, // tcp被动模式，tcp服务器，等待对方连接并回复rtp
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kUdpPassive = 3 // udp被动方式，等待对方发送nat打洞包，然后回复rtp至打洞包源地址
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};
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// rtp类型
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DataType data_type = kRtpPS;
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// 连接类型
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ConType con_type = kUdpActive;
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// 发送es流时指定是否只发送纯音频流
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bool only_audio = false;
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// rtp payload type
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uint8_t pt = 96;
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// 是否支持同ssrc多服务器发送
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bool ssrc_multi_send = false;
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// 指定rtp ssrc
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std::string ssrc;
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// 指定本地发送端口
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uint16_t src_port = 0;
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// 发送目标端口
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uint16_t dst_port;
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// 发送目标主机地址，可以是ip或域名
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std::string dst_url;
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// udp发送时，是否开启rr rtcp接收超时判断
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bool udp_rtcp_timeout = false;
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// passive被动、tcp主动模式超时时间
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uint32_t close_delay_ms = 0;
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// udp 发送时，rr rtcp包接收超时时间，单位毫秒
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uint32_t rtcp_timeout_ms = 30 * 1000;
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// udp 发送时，发送sr rtcp包间隔，单位毫秒
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uint32_t rtcp_send_interval_ms = 5 * 1000;
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// 发送rtp同时接收，一般用于双向语言对讲, 如果不为空，说明开启接收
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std::string recv_stream_id;
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};
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// 开始发送ps-rtp
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virtual void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) { cb(0, toolkit::SockException(toolkit::Err_other, "not implemented"));};
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// 停止发送ps-rtp
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virtual bool stopSendRtp(MediaSource &sender, const std::string &ssrc) {return false; }
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private:
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toolkit::Timer::Ptr _async_close_timer;
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};
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template <typename MAP, typename KEY, typename TYPE>
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static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
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auto val = ((MAP &)allArgs)[key];
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if (!val.empty()) {
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value = (TYPE)val;
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}
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}
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template <typename KEY, typename TYPE>
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static void getArgsValue(const toolkit::mINI &allArgs, const KEY &key, TYPE &value) {
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auto it = allArgs.find(key);
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if (it != allArgs.end()) {
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value = (TYPE)it->second;
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}
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}
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class ProtocolOption {
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public:
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ProtocolOption();
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enum {
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kModifyStampOff = 0, // 采用源视频流绝对时间戳，不做任何改变
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kModifyStampSystem = 1, // 采用zlmediakit接收数据时的系统时间戳(有平滑处理)
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kModifyStampRelative = 2 // 采用源视频流时间戳相对时间戳(增长量)，有做时间戳跳跃和回退矫正
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};
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// 时间戳类型
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int modify_stamp;
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//转协议是否开启音频
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bool enable_audio;
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//添加静音音频，在关闭音频时，此开关无效
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bool add_mute_audio;
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// 无人观看时，是否直接关闭(而不是通过on_none_reader hook返回close)
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// 此配置置1时，此流如果无人观看，将不触发on_none_reader hook回调，
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// 而是将直接关闭流
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bool auto_close;
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//断连续推延时，单位毫秒，默认采用配置文件
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uint32_t continue_push_ms;
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// 平滑发送定时器间隔，单位毫秒，置0则关闭；开启后影响cpu性能同时增加内存
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// 该配置开启后可以解决一些流发送不平滑导致zlmediakit转发也不平滑的问题
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uint32_t paced_sender_ms;
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//是否开启转换为hls(mpegts)
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bool enable_hls;
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//是否开启转换为hls(fmp4)
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bool enable_hls_fmp4;
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//是否开启MP4录制
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bool enable_mp4;
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//是否开启转换为rtsp/webrtc
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bool enable_rtsp;
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//是否开启转换为rtmp/flv
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bool enable_rtmp;
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//是否开启转换为http-ts/ws-ts
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bool enable_ts;
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//是否开启转换为http-fmp4/ws-fmp4
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bool enable_fmp4;
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// hls协议是否按需生成，如果hls.segNum配置为0(意味着hls录制)，那么hls将一直生成(不管此开关)
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bool hls_demand;
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// rtsp[s]协议是否按需生成
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bool rtsp_demand;
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// rtmp[s]、http[s]-flv、ws[s]-flv协议是否按需生成
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bool rtmp_demand;
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// http[s]-ts协议是否按需生成
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bool ts_demand;
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// http[s]-fmp4、ws[s]-fmp4协议是否按需生成
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bool fmp4_demand;
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//是否将mp4录制当做观看者
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bool mp4_as_player;
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//mp4切片大小，单位秒
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size_t mp4_max_second;
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//mp4录制保存路径
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std::string mp4_save_path;
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//hls录制保存路径
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std::string hls_save_path;
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// 支持通过on_publish返回值替换stream_id
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std::string stream_replace;
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// 最大track数
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size_t max_track = 2;
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template <typename MAP>
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ProtocolOption(const MAP &allArgs) : ProtocolOption() {
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load(allArgs);
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}
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template <typename MAP>
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void load(const MAP &allArgs) {
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#define GET_OPT_VALUE(key) getArgsValue(allArgs, #key, key)
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GET_OPT_VALUE(modify_stamp);
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GET_OPT_VALUE(enable_audio);
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GET_OPT_VALUE(add_mute_audio);
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GET_OPT_VALUE(auto_close);
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GET_OPT_VALUE(continue_push_ms);
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GET_OPT_VALUE(paced_sender_ms);
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GET_OPT_VALUE(enable_hls);
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GET_OPT_VALUE(enable_hls_fmp4);
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GET_OPT_VALUE(enable_mp4);
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GET_OPT_VALUE(enable_rtsp);
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GET_OPT_VALUE(enable_rtmp);
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GET_OPT_VALUE(enable_ts);
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GET_OPT_VALUE(enable_fmp4);
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GET_OPT_VALUE(hls_demand);
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GET_OPT_VALUE(rtsp_demand);
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GET_OPT_VALUE(rtmp_demand);
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GET_OPT_VALUE(ts_demand);
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GET_OPT_VALUE(fmp4_demand);
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GET_OPT_VALUE(mp4_max_second);
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GET_OPT_VALUE(mp4_as_player);
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GET_OPT_VALUE(mp4_save_path);
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GET_OPT_VALUE(hls_save_path);
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GET_OPT_VALUE(stream_replace);
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GET_OPT_VALUE(max_track);
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}
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};
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//该对象用于拦截感兴趣的MediaSourceEvent事件
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class MediaSourceEventInterceptor : public MediaSourceEvent {
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public:
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void setDelegate(const std::weak_ptr<MediaSourceEvent> &listener);
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std::shared_ptr<MediaSourceEvent> getDelegate() const;
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MediaOriginType getOriginType(MediaSource &sender) const override;
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std::string getOriginUrl(MediaSource &sender) const override;
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std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const override;
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bool seekTo(MediaSource &sender, uint32_t stamp) override;
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bool pause(MediaSource &sender, bool pause) override;
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bool speed(MediaSource &sender, float speed) override;
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bool close(MediaSource &sender) override;
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int totalReaderCount(MediaSource &sender) override;
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void onReaderChanged(MediaSource &sender, int size) override;
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void onRegist(MediaSource &sender, bool regist) override;
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bool setupRecord(MediaSource &sender, Recorder::type type, bool start, const std::string &custom_path, size_t max_second) override;
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bool isRecording(MediaSource &sender, Recorder::type type) override;
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std::vector<Track::Ptr> getMediaTracks(MediaSource &sender, bool trackReady = true) const override;
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void startSendRtp(MediaSource &sender, const SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb) override;
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bool stopSendRtp(MediaSource &sender, const std::string &ssrc) override;
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float getLossRate(MediaSource &sender, TrackType type) override;
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toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
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std::shared_ptr<MultiMediaSourceMuxer> getMuxer(MediaSource &sender) const override;
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std::shared_ptr<RtpProcess> getRtpProcess(MediaSource &sender) const override;
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private:
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std::weak_ptr<MediaSourceEvent> _listener;
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};
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/**
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* 解析url获取媒体相关信息
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*/
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class MediaInfo: public MediaTuple {
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public:
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MediaInfo() = default;
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MediaInfo(const std::string &url) { parse(url); }
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void parse(const std::string &url);
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std::string getUrl() const { return schema + "://" + shortUrl(); }
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public:
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uint16_t port = 0;
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std::string full_url;
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std::string schema;
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std::string host;
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};
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bool equalMediaTuple(const MediaTuple& a, const MediaTuple& b);
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/**
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* 媒体源，任何rtsp/rtmp的直播流都源自该对象
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*/
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class MediaSource: public TrackSource, public std::enable_shared_from_this<MediaSource> {
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public:
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static MediaSource& NullMediaSource();
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using Ptr = std::shared_ptr<MediaSource>;
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MediaSource(const std::string &schema, const MediaTuple& tuple);
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virtual ~MediaSource();
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////////////////获取MediaSource相关信息////////////////
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// 获取协议类型
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const std::string& getSchema() const {
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return _schema;
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}
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const MediaTuple& getMediaTuple() const {
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return _tuple;
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}
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std::string getUrl() const { return _schema + "://" + _tuple.shortUrl(); }
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//获取对象所有权
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std::shared_ptr<void> getOwnership();
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// 获取所有Track
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std::vector<Track::Ptr> getTracks(bool ready = true) const override;
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// 获取流当前时间戳
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virtual uint32_t getTimeStamp(TrackType type) { return 0; };
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// 设置时间戳
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virtual void setTimeStamp(uint32_t stamp) {};
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// 获取数据速率，单位bytes/s
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int getBytesSpeed(TrackType type = TrackInvalid);
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// 获取流创建GMT unix时间戳，单位秒
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uint64_t getCreateStamp() const { return _create_stamp; }
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// 获取流上线时间，单位秒
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uint64_t getAliveSecond() const;
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////////////////MediaSourceEvent相关接口实现////////////////
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// 设置监听者
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virtual void setListener(const std::weak_ptr<MediaSourceEvent> &listener);
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// 获取监听者
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std::weak_ptr<MediaSourceEvent> getListener() const;
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// 本协议获取观看者个数，可能返回本协议的观看人数，也可能返回总人数
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virtual int readerCount() = 0;
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// 观看者个数，包括(hls/rtsp/rtmp)
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virtual int totalReaderCount();
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// 获取播放器列表
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virtual void getPlayerList(const std::function<void(const std::list<toolkit::Any> &info_list)> &cb,
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const std::function<toolkit::Any(toolkit::Any &&info)> &on_change) {
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assert(cb);
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cb(std::list<toolkit::Any>());
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}
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virtual bool broadcastMessage(const toolkit::Any &data) { return false; }
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// 获取媒体源类型
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MediaOriginType getOriginType() const;
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// 获取媒体源url或者文件路径
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std::string getOriginUrl() const;
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// 获取媒体源客户端相关信息
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std::shared_ptr<toolkit::SockInfo> getOriginSock() const;
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// 拖动进度条
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bool seekTo(uint32_t stamp);
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// 暂停
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bool pause(bool pause);
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// 倍数播放
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bool speed(float speed);
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// 关闭该流
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bool close(bool force);
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// 该流观看人数变化
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void onReaderChanged(int size);
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// 开启或关闭录制
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bool setupRecord(Recorder::type type, bool start, const std::string &custom_path, size_t max_second);
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// 获取录制状态
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bool isRecording(Recorder::type type);
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// 开始发送ps-rtp
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void startSendRtp(const MediaSourceEvent::SendRtpArgs &args, const std::function<void(uint16_t, const toolkit::SockException &)> cb);
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// 停止发送ps-rtp
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bool stopSendRtp(const std::string &ssrc);
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// 获取丢包率
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float getLossRate(mediakit::TrackType type);
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// 获取所在线程
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toolkit::EventPoller::Ptr getOwnerPoller();
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// 获取MultiMediaSourceMuxer对象
|
||
std::shared_ptr<MultiMediaSourceMuxer> getMuxer() const;
|
||
// 获取RtpProcess对象
|
||
std::shared_ptr<RtpProcess> getRtpProcess() const;
|
||
|
||
////////////////static方法，查找或生成MediaSource////////////////
|
||
|
||
// 同步查找流
|
||
static Ptr find(const std::string &schema, const std::string &vhost, const std::string &app, const std::string &id, bool from_mp4 = false);
|
||
static Ptr find(const MediaInfo &info, bool from_mp4 = false) {
|
||
return find(info.schema, info.vhost, info.app, info.stream, from_mp4);
|
||
}
|
||
|
||
// 忽略schema，同步查找流，可能返回rtmp/rtsp/hls类型
|
||
static Ptr find(const std::string &vhost, const std::string &app, const std::string &stream_id, bool from_mp4 = false);
|
||
|
||
// 异步查找流
|
||
static void findAsync(const MediaInfo &info, const std::shared_ptr<toolkit::Session> &session, const std::function<void(const Ptr &src)> &cb);
|
||
// 遍历所有流
|
||
static void for_each_media(const std::function<void(const Ptr &src)> &cb, const std::string &schema = "", const std::string &vhost = "", const std::string &app = "", const std::string &stream = "");
|
||
// 从mp4文件生成MediaSource
|
||
static MediaSource::Ptr createFromMP4(const std::string &schema, const std::string &vhost, const std::string &app, const std::string &stream, const std::string &file_path = "", bool check_app = true);
|
||
|
||
protected:
|
||
//媒体注册
|
||
void regist();
|
||
|
||
private:
|
||
// 媒体注销
|
||
bool unregist();
|
||
// 触发媒体事件
|
||
void emitEvent(bool regist);
|
||
|
||
protected:
|
||
toolkit::BytesSpeed _speed[TrackMax];
|
||
MediaTuple _tuple;
|
||
|
||
private:
|
||
std::atomic_flag _owned { false };
|
||
time_t _create_stamp;
|
||
toolkit::Ticker _ticker;
|
||
std::string _schema;
|
||
std::weak_ptr<MediaSourceEvent> _listener;
|
||
// 对象个数统计
|
||
toolkit::ObjectStatistic<MediaSource> _statistic;
|
||
};
|
||
|
||
} /* namespace mediakit */
|
||
#endif //ZLMEDIAKIT_MEDIASOURCE_H
|