mirror of
https://github.com/ZLMediaKit/ZLMediaKit.git
synced 2026-07-05 02:38:10 +08:00
feat: 增加webrtc代理拉流 (#4389)
- 增加客户端模式，支持主动拉流、推流： - addStreamProxy接口新增支持whep主动拉流，拉流地址目前只兼容zlm的whep url。 - addStreamPusherProxy接口新增支持whip主动推流，推流地址目前只兼容zlm的whip url。 - 以上推流url格式为webrtc[s]://server_host:server_port/app/stream_id?key=value, 内部会自动转换为http[s]://server_host:server_port/index/api/[whip/whep]?app=app&stream=stream_id&key=value。 - 增加WebRtc p2p 模式： - 增加 ICE FULL模式。 - 增加STUN/TURN 服务器。 - 增加websocket 信令。 - 增加P2P代理拉流。 --------- Co-authored-by: xia-chu <771730766@qq.com> Co-authored-by: mtdxc <mtdxc@126.com> Co-authored-by: cqm <cqm@97kid.com>
This commit is contained in:
baigao-X
@@ -1,164 +1,165 @@
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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 ZLMEDIAKIT_WEBRTCPLAYER_H
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#define ZLMEDIAKIT_WEBRTCPLAYER_H
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#include "Rtsp/RtspMediaSource.h"
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#include "WebRtcTransport.h"
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namespace mediakit {
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/**
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* @brief H.264 B 帧过滤器
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* 用于从 H.264 RTP 流中移除 B 帧
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*/
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class H264BFrameFilter {
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public:
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/**
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* ISO_IEC_14496-10-AVC-2012
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* Table 7-6 – Name association to slice_type
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*/
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enum H264SliceType {
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H264SliceTypeP = 0,
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H264SliceTypeB = 1,
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H264SliceTypeI = 2,
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H264SliceTypeSP = 3,
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H264SliceTypeSI = 4,
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H264SliceTypeP1 = 5,
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H264SliceTypeB1 = 6,
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H264SliceTypeI1 = 7,
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H264SliceTypeSP1 = 8,
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H264SliceTypeSI1 = 9,
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};
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enum H264NALUType {
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NAL_NIDR = 1,
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NAL_PARTITION_A = 2,
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NAL_PARTITION_B = 3,
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NAL_PARTITION_C = 4,
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NAL_IDR = 5,
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};
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H264BFrameFilter();
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~H264BFrameFilter() = default;
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/**
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* @brief 处理单个 RTP 包，移除 B 帧
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* @param packet 输入的 RTP 包
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* @return 如果不是 B 帧则返回原包，否则返回 nullptr
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*/
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RtpPacket::Ptr processPacket(const RtpPacket::Ptr &packet);
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private:
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/**
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* @brief 判断 RTP 包是否包含 H.264 的 B 帧
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* @param packet RTP 包
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool isH264BFrame(const RtpPacket::Ptr &packet) const;
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/**
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* @brief 根据 NAL 类型和数据判断是否是 B 帧
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* @param nal_type NAL 单元类型
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* @param data NAL 单元数据（不含 NAL 头）
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* @param size 数据大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool isBFrameByNalType(uint8_t nal_type, const uint8_t *data, size_t size) const;
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/**
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* @brief 解析指数哥伦布编码
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @param bits_offset 位偏移量
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* @return 解析出的数值
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*/
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int decodeExpGolomb(const uint8_t *data, size_t size, size_t &bitPos) const;
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/**
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* @brief 从比特流中读取位
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @return 读取的位值（0 或 1）
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*/
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int getBit(const uint8_t *data, size_t size) const;
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/**
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* @brief 提取切片类型值
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @return 切片类型值
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*/
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uint8_t extractSliceType(const uint8_t *data, size_t size) const;
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/**
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* @brief 处理FU-A分片
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* @param payload 数据缓冲区
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* @param payload_size 缓冲区大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool handleFua(const uint8_t *payload, size_t payload_size) const;
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/**
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* @brief 处理 STAP-A 组合包
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* @param payload 数据缓冲区
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* @param payload_size 缓冲区大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool handleStapA(const uint8_t *payload, size_t payload_size) const;
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private:
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uint16_t _last_seq; // 维护输出流的序列号
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uint32_t _last_stamp; // 维护输出流的时间戳
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bool _first_packet; // 是否是第一个包的标记
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};
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class WebRtcPlayer : public WebRtcTransportImp {
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public:
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using Ptr = std::shared_ptr<WebRtcPlayer>;
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static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
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MediaInfo getMediaInfo() { return _media_info; }
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protected:
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///////WebRtcTransportImp override///////
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void onStartWebRTC() override;
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void onDestory() override;
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void onRtcConfigure(RtcConfigure &configure) const override;
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private:
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WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
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void sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp);
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private:
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// 媒体相关元数据 [AUTO-TRANSLATED:f4cf8045]
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// Media related metadata
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MediaInfo _media_info;
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// 播放的rtsp源 [AUTO-TRANSLATED:9963eed1]
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// Playing rtsp source
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std::weak_ptr<RtspMediaSource> _play_src;
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// rtp 直接转发情况下通常会缺少 sps/pps, 在转发 rtp 前, 先发送一次相关帧信息, 部分情况下是可以播放的 [AUTO-TRANSLATED:65fdf16a]
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// In the case of direct RTP forwarding, sps/pps is usually missing. Before forwarding RTP, send the relevant frame information once. In some cases, it can be played.
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bool _send_config_frames_once { false };
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// 播放rtsp源的reader对象 [AUTO-TRANSLATED:7b305055]
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// Reader object for playing rtsp source
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RtspMediaSource::RingType::RingReader::Ptr _reader;
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bool _is_h264 { false };
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bool _bfliter_flag { false };
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std::shared_ptr<H264BFrameFilter> _bfilter;
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};
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}// namespace mediakit
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#endif // ZLMEDIAKIT_WEBRTCPLAYER_H
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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 ZLMEDIAKIT_WEBRTCPLAYER_H
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#define ZLMEDIAKIT_WEBRTCPLAYER_H
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#include "WebRtcTransport.h"
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#include "Rtsp/RtspMediaSource.h"
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namespace mediakit {
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/**
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* @brief H.264 B 帧过滤器
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* 用于从 H.264 RTP 流中移除 B 帧
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*/
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class H264BFrameFilter {
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public:
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/**
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* ISO_IEC_14496-10-AVC-2012
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* Table 7-6 – Name association to slice_type
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*/
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enum H264SliceType {
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H264SliceTypeP = 0,
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H264SliceTypeB = 1,
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H264SliceTypeI = 2,
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H264SliceTypeSP = 3,
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H264SliceTypeSI = 4,
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H264SliceTypeP1 = 5,
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H264SliceTypeB1 = 6,
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H264SliceTypeI1 = 7,
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H264SliceTypeSP1 = 8,
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H264SliceTypeSI1 = 9,
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};
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enum H264NALUType {
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NAL_NIDR = 1,
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NAL_PARTITION_A = 2,
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NAL_PARTITION_B = 3,
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NAL_PARTITION_C = 4,
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NAL_IDR = 5,
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};
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H264BFrameFilter();
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~H264BFrameFilter() = default;
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/**
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* @brief 处理单个 RTP 包，移除 B 帧
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* @param packet 输入的 RTP 包
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* @return 如果不是 B 帧则返回原包，否则返回 nullptr
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*/
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RtpPacket::Ptr processPacket(const RtpPacket::Ptr &packet);
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private:
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/**
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* @brief 判断 RTP 包是否包含 H.264 的 B 帧
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* @param packet RTP 包
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool isH264BFrame(const RtpPacket::Ptr &packet) const;
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/**
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* @brief 根据 NAL 类型和数据判断是否是 B 帧
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* @param nal_type NAL 单元类型
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* @param data NAL 单元数据（不含 NAL 头）
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* @param size 数据大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool isBFrameByNalType(uint8_t nal_type, const uint8_t *data, size_t size) const;
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/**
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* @brief 解析指数哥伦布编码
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @param bits_offset 位偏移量
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* @return 解析出的数值
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*/
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int decodeExpGolomb(const uint8_t *data, size_t size, size_t &bitPos) const;
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/**
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* @brief 从比特流中读取位
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @return 读取的位值（0 或 1）
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*/
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int getBit(const uint8_t *data, size_t size) const;
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/**
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* @brief 提取切片类型值
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* @param data 数据缓冲区
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* @param size 缓冲区大小
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* @return 切片类型值
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*/
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uint8_t extractSliceType(const uint8_t *data, size_t size) const;
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/**
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* @brief 处理FU-A分片
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* @param payload 数据缓冲区
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* @param payload_size 缓冲区大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool handleFua(const uint8_t *payload, size_t payload_size) const;
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/**
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* @brief 处理 STAP-A 组合包
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* @param payload 数据缓冲区
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* @param payload_size 缓冲区大小
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* @return 如果是 B 帧返回 true，否则返回 false
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*/
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bool handleStapA(const uint8_t *payload, size_t payload_size) const;
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private:
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uint16_t _last_seq; // 维护输出流的序列号
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uint32_t _last_stamp; // 维护输出流的时间戳
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bool _first_packet; // 是否是第一个包的标记
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};
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class WebRtcPlayer : public WebRtcTransportImp {
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public:
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using Ptr = std::shared_ptr<WebRtcPlayer>;
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static Ptr create(const toolkit::EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info,
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WebRtcTransport::Role role, WebRtcTransport::SignalingProtocols signaling_protocols);
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MediaInfo getMediaInfo() { return _media_info; }
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protected:
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///////WebRtcTransportImp override///////
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void onStartWebRTC() override;
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void onDestory() override;
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void onRtcConfigure(RtcConfigure &configure) const override;
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private:
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WebRtcPlayer(const toolkit::EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
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void sendConfigFrames(uint32_t before_seq, uint32_t sample_rate, uint32_t timestamp, uint64_t ntp_timestamp);
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private:
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// 媒体相关元数据 [AUTO-TRANSLATED:f4cf8045]
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// Media related metadata
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MediaInfo _media_info;
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// 播放的rtsp源 [AUTO-TRANSLATED:9963eed1]
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// Playing rtsp source
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std::weak_ptr<RtspMediaSource> _play_src;
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// rtp 直接转发情况下通常会缺少 sps/pps, 在转发 rtp 前, 先发送一次相关帧信息, 部分情况下是可以播放的 [AUTO-TRANSLATED:65fdf16a]
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// In the case of direct RTP forwarding, sps/pps is usually missing. Before forwarding RTP, send the relevant frame information once. In some cases, it can be played.
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bool _send_config_frames_once { false };
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// 播放rtsp源的reader对象 [AUTO-TRANSLATED:7b305055]
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// Reader object for playing rtsp source
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RtspMediaSource::RingType::RingReader::Ptr _reader;
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bool _is_h264 { false };
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bool _bfliter_flag { false };
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std::shared_ptr<H264BFrameFilter> _bfilter;
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};
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}// namespace mediakit
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#endif // ZLMEDIAKIT_WEBRTCPLAYER_H
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