213 lines
6.2 KiB
C
213 lines
6.2 KiB
C
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/*
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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 SRC_RTSP_RTSPMEDIASOURCE_H_
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#define SRC_RTSP_RTSPMEDIASOURCE_H_
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#include <mutex>
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#include <string>
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#include <memory>
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#include <functional>
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#include "Common/MediaSource.h"
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#include "Common/PacketCache.h"
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#include "Util/RingBuffer.h"
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#define RTP_GOP_SIZE 512
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namespace mediakit {
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/**
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* rtsp媒体源的数据抽象
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* rtsp有关键的两要素,分别是sdp、rtp包
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* 只要生成了这两要素,那么要实现rtsp推流、rtsp服务器就很简单了
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* rtsp推拉流协议中,先传递sdp,然后再协商传输方式(tcp/udp/组播),最后一直传递rtp
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* Data abstraction of rtsp media source
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* Rtsp has two key elements, sdp and rtp packets
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* As long as these two elements are generated, it is very simple to implement rtsp push stream and rtsp server
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* In the rtsp push and pull stream protocol, sdp is transmitted first, then the transmission method (tcp/udp/multicast) is negotiated, and finally rtp is continuously transmitted
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* [AUTO-TRANSLATED:e04eee56]
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*/
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class RtspMediaSource : public MediaSource, public toolkit::RingDelegate<RtpPacket::Ptr>, private PacketCache<RtpPacket> {
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public:
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using Ptr = std::shared_ptr<RtspMediaSource>;
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using RingDataType = std::shared_ptr<toolkit::List<RtpPacket::Ptr> >;
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using RingType = toolkit::RingBuffer<RingDataType>;
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/**
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* 构造函数
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* @param vhost 虚拟主机名
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* @param app 应用名
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* @param stream_id 流id
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* @param ring_size 可以设置固定的环形缓冲大小,0则自适应
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* Constructor
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* @param vhost Virtual host name
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* @param app Application name
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* @param stream_id Stream id
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* @param ring_size You can set a fixed ring buffer size, 0 is adaptive
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* [AUTO-TRANSLATED:5dd23423]
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*/
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RtspMediaSource(const MediaTuple& tuple, int ring_size = RTP_GOP_SIZE): MediaSource(RTSP_SCHEMA, tuple), _ring_size(ring_size) {}
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~RtspMediaSource() override {
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try {
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flush();
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} catch (std::exception &ex) {
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WarnL << ex.what();
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}
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}
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/**
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* 获取媒体源的环形缓冲
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* Get the ring buffer of the media source
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* [AUTO-TRANSLATED:91a762bc]
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*/
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const RingType::Ptr &getRing() const {
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return _ring;
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}
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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) override {
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assert(_ring);
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_ring->getInfoList(cb, on_change);
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}
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bool broadcastMessage(const toolkit::Any &data) override {
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assert(_ring);
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_ring->sendMessage(data);
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return true;
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}
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/**
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* 获取播放器个数
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* Get the number of players
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* [AUTO-TRANSLATED:a451c846]
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*/
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int readerCount() override {
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return _ring ? _ring->readerCount() : 0;
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}
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/**
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* 获取该源的sdp
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* Get the sdp of this source
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* [AUTO-TRANSLATED:ebc43430]
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*/
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const std::string &getSdp() const {
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return _sdp;
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}
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virtual RtspMediaSource::Ptr clone(const std::string& stream) {
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return nullptr;
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}
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/**
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* 获取相应轨道的ssrc
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* Get the ssrc of the corresponding track
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* [AUTO-TRANSLATED:d26d7f76]
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*/
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virtual uint32_t getSsrc(TrackType trackType) {
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assert(trackType >= 0 && trackType < TrackMax);
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auto &track = _tracks[trackType];
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if (!track) {
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return 0;
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}
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return track->_ssrc;
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}
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/**
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* 获取相应轨道的seqence
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* Get the sequence of the corresponding track
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* [AUTO-TRANSLATED:24b0ee74]
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*/
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virtual uint16_t getSeqence(TrackType trackType) {
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assert(trackType >= 0 && trackType < TrackMax);
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auto &track = _tracks[trackType];
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if (!track) {
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return 0;
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}
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return track->_seq;
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}
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/**
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* 获取相应轨道的时间戳,单位毫秒
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* Get the timestamp of the corresponding track, in milliseconds
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* [AUTO-TRANSLATED:564a0794]
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*/
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uint32_t getTimeStamp(TrackType trackType) override;
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/**
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* 更新时间戳
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* Update timestamp
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* [AUTO-TRANSLATED:8defe253]
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*/
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void setTimeStamp(uint32_t stamp) override;
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/**
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* 设置sdp
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* Set sdp
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* [AUTO-TRANSLATED:76a533c4]
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*/
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virtual void setSdp(const std::string &sdp);
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/**
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* 输入rtp
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* @param rtp rtp包
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* @param keyPos 该包是否为关键帧的第一个包
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* Input rtp
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* @param rtp rtp packet
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* @param keyPos Whether this packet is the first packet of a key frame
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* [AUTO-TRANSLATED:fe55afe8]
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*/
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void onWrite(RtpPacket::Ptr rtp, bool keyPos) override;
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void clearCache() override{
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PacketCache<RtpPacket>::clearCache();
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_ring->clearCache();
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}
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private:
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/**
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* 批量flush rtp包时触发该函数
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* @param rtp_list rtp包列表
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* @param key_pos 是否包含关键帧
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* Trigger this function when flushing rtp packets in batches
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* @param rtp_list rtp packet list
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* @param key_pos Whether it contains a key frame
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* [AUTO-TRANSLATED:612c574b]
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*/
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void onFlush(std::shared_ptr<toolkit::List<RtpPacket::Ptr> > rtp_list, bool key_pos) override {
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// 如果不存在视频,那么就没有存在GOP缓存的意义,所以is_key一直为true确保一直清空GOP缓存 [AUTO-TRANSLATED:5818a8d8]
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// If there is no video, then there is no point in having a GOP cache, so is_key is always true to ensure that the GOP cache is always cleared
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_ring->write(std::move(rtp_list), _have_video ? key_pos : true);
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}
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private:
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bool _have_video = false;
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int _ring_size;
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std::string _sdp;
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RingType::Ptr _ring;
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SdpTrack::Ptr _tracks[TrackMax];
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};
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} /* namespace mediakit */
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#endif /* SRC_RTSP_RTSPMEDIASOURCE_H_ */
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