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ZLMediaKit
Commit 7805558d by baiyfcu Committed by GitHub
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Merge pull request #13 from xiongziliang/master 

update
parents 40afa204 003cd583
正在显示 包含 920 行增加 和 1016 行删除
ZLToolKit @ 987683f1
Subproject commit 681be205ef164db08effd83f925bb750eb1fe149
Subproject commit 987683f1045613098e2bcd534bc90a13d16df8a4
media-server @ 24519a59
Subproject commit 97cf5e47a5af1ff3d4d187f3ebffd9254595df75
Subproject commit 24519a594c2c634b21fbe09fad28d54c4eba0885
README.md
......@@ -3,29 +3,33 @@
[english readme](https://github.com/xiongziliang/ZLMediaKit/blob/master/README_en.md)
# 一个基于C++11的高性能运营级流媒体服务框架
[![Build Status](https://travis-ci.org/xiongziliang/ZLMediaKit.svg?branch=master)](https://travis-ci.org/xiongziliang/ZLMediaKit)
## 项目特点
- 基于C++11开发，避免使用裸指针，代码稳定可靠；同时跨平台移植简单方便，代码清晰简洁。
- 打包多种流媒体协议(RTSP/RTMP/HLS/HTTP-FLV/Websocket-FLV），支持协议间的互相转换，提供一站式的服务。
- 使用epoll+线程池+异步网络IO模式开发，并发性能优越。
- 已实现主流的的H264/H265+AAC流媒体方案，代码精简,脉络清晰，适合学习。
- 编码格式与框架代码解耦，方便自由简洁的添加支持其他编码格式。
- 代码经过大量的稳定性、性能测试，可满足商用服务器项目。
- 支持linux、macos、ios、android、windows平台。
- 支持画面秒开(GOP缓存)、极低延时([500毫秒内，最低可达100毫秒](https://github.com/zlmediakit/ZLMediaKit/wiki/%E5%BB%B6%E6%97%B6%E6%B5%8B%E8%AF%95))。
- [ZLMediaKit高并发实现原理](https://github.com/xiongziliang/ZLMediaKit/wiki/ZLMediaKit%E9%AB%98%E5%B9%B6%E5%8F%91%E5%AE%9E%E7%8E%B0%E5%8E%9F%E7%90%86)。
- 基于C++11开发，避免使用裸指针，代码稳定可靠，性能优越。
- 支持多种协议(RTSP/RTMP/HLS/HTTP-FLV/Websocket-FLV/GB28181/MP4),支持协议互转。
- 使用多路复用/多线程/异步网络IO模式开发，并发性能优越，支持海量客户端连接。
- 代码经过长期大量的稳定性、性能测试，已经在线上商用验证已久。
- 支持linux、macos、ios、android、windows全平台。
- 支持画面秒开、极低延时([500毫秒内，最低可达100毫秒](https://github.com/zlmediakit/ZLMediaKit/wiki/%E5%BB%B6%E6%97%B6%E6%B5%8B%E8%AF%95))。
- 提供完善的标准[C API](https://github.com/xiongziliang/ZLMediaKit/tree/master/api/include),可以作SDK用，或供其他语言调用。
- 提供完整的[MediaServer](https://github.com/xiongziliang/ZLMediaKit/tree/master/server)服务器，可以免开发直接部署为商用服务器。
- 提供完善的[restful api](https://github.com/xiongziliang/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-API)以及[web hook](https://github.com/xiongziliang/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-HOOK-API)，支持丰富的业务逻辑。
- 打通了视频监控协议栈与直播协议栈，对RTSP/RTMP支持都很完善。
- 全面支持H265。
## 项目定位
- 移动嵌入式跨平台流媒体解决方案。
- 商用级流媒体服务器。
- 网络编程二次开发SDK。
## 功能清单
- RTSP
- RTSP 服务器，支持RTMP/MP4转RTSP
- RTSPS 服务器，支持亚马逊echo show这样的设备
......@@ -35,7 +39,7 @@
- 服务器/客户端完整支持Basic/Digest方式的登录鉴权，全异步可配置化的鉴权接口
- 支持H265编码
- 服务器支持RTSP推流(包括`rtp over udp` `rtp over tcp`方式)
- 支持任意编码格式的rtsp推流，只是除H264/H265+AAC外无法转协议
- 支持任意编码格式的rtsp推流，只是除H264/H265/AAC/G711外无法转协议
- RTMP
- RTMP 播放服务器，支持RTSP/MP4转RTMP
......@@ -44,7 +48,7 @@
- RTMP 推流客户端
- 支持http[s]-flv直播
- 支持websocket-flv直播
- 支持任意编码格式的rtmp推流，只是除H264/H265+AAC外无法转协议
- 支持任意编码格式的rtmp推流，只是除H264/H265/AAC/G711外无法转协议
- 支持[RTMP-H265](https://github.com/ksvc/FFmpeg/wiki)
- HLS
......@@ -63,11 +67,11 @@
- GB28181
- 支持UDP/TCP国标RTP(PS或TS)推流，可以转换成RTSP/RTMP/HLS等协议
- 点播
- 支持录制为FLV/HLS/MP4
- RTSP/RTMP/HTTP-FLV/WS-FLV支持MP4文件点播，支持seek
- 其他
- 支持丰富的restful api以及web hook事件
- 支持简单的telnet调试
......@@ -77,170 +81,43 @@
- 支持按需拉流，无人观看自动关断拉流
- 支持先拉流后推流，提高及时推流画面打开率
- 提供c api sdk
## 细节列表
- 转协议:
| 功能/编码格式 | H264 | H265 | AAC | other |
| :------------------------------: | :--: | :--: | :--: | :---: |
| RTSP[S] --> RTMP/HTTP[S]-FLV/FLV | Y | Y | Y | N |
| RTMP --> RTSP[S] | Y | Y | Y | N |
| RTSP[S] --> HLS | Y | Y | Y | N |
| RTMP --> HLS | Y | Y | Y | N |
| RTSP[S] --> MP4 | Y | Y | Y | N |
| RTMP --> MP4 | Y | Y | Y | N |
| MP4 --> RTSP[S] | Y | Y | Y | N |
| MP4 --> RTMP | Y | Y | Y | N |
- 流生成：
| 功能/编码格式 | H264 | H265 | AAC | other |
| :------------------------------: | :--: | :--: | :--: | :---: |
| RTSP[S]推流 | Y | Y | Y | Y |
| RTSP拉流代理 | Y | Y | Y | Y |
| RTMP推流 | Y | Y | Y | Y |
| RTMP拉流代理 | Y | Y | Y | Y |
- RTP传输方式:
| 功能/RTP传输方式 | tcp | udp | http | udp_multicast |
| :-----------------: | :--: | :--: | :--: | :-----------: |
| RTSP[S] Play Server | Y | Y | Y | Y |
| RTSP[S] Push Server | Y | Y | N | N |
| RTSP Player | Y | Y | N | Y |
| RTSP Pusher | Y | Y | N | N |
- 支持的服务器类型列表
## 编译以及测试
| 服务类型 | Y/N |
| :-----------------: | :--: |
| RTSP[S] Play Server | Y |
| RTSP[S] Push Server | Y |
| RTMP | Y |
| HTTP[S]/WebSocket[S] | Y |
请参考wiki:[快速开始](https://github.com/xiongziliang/ZLMediaKit/wiki/%E5%BF%AB%E9%80%9F%E5%BC%80%E5%A7%8B)
- 支持的客户端类型
## 怎么使用
| 客户端类型 | Y/N |
| :---------: | :--: |
| RTSP Player | Y |
| RTSP Pusher | Y |
| RTMP Player | Y |
| RTMP Pusher | Y |
| HTTP[S] | Y |
| WebSocket[S] | Y |
你有三种方法使用ZLMediaKit，分别是：
## 编译以及测试
请参考wiki:[快速开始](https://github.com/xiongziliang/ZLMediaKit/wiki/%E5%BF%AB%E9%80%9F%E5%BC%80%E5%A7%8B)
- 1、使用c api，作为sdk使用，请参考[这里](https://github.com/xiongziliang/ZLMediaKit/tree/master/api/include).
- 2、作为独立的流媒体服务器使用，不想做c/c++开发的，可以参考[restful api](https://github.com/xiongziliang/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-API)和[web hook](https://github.com/xiongziliang/ZLMediaKit/wiki/MediaServer%E6%94%AF%E6%8C%81%E7%9A%84HTTP-HOOK-API).
- 3、如果想做c/c++开发，添加业务逻辑增加功能，可以参考这里的[测试程序](https://github.com/xiongziliang/ZLMediaKit/tree/master/tests).
## Docker 镜像
你可以从Docker Hub下载已经编译好的镜像并启动它：
```bash
docker run -id -p 1935:1935 -p 8080:80 gemfield/zlmediakit:20.04-runtime-ubuntu18.04
```
你也可以根据Dockerfile编译镜像：
```bash
bash build_docker_images.sh
```
## 使用方法
- 作为服务器：
```cpp
TcpServer::Ptr rtspSrv(new TcpServer());
TcpServer::Ptr rtmpSrv(new TcpServer());
TcpServer::Ptr httpSrv(new TcpServer());
TcpServer::Ptr httpsSrv(new TcpServer());
rtspSrv->start<RtspSession>(mINI::Instance()[Config::Rtsp::kPort]);
rtmpSrv->start<RtmpSession>(mINI::Instance()[Config::Rtmp::kPort]);
httpSrv->start<HttpSession>(mINI::Instance()[Config::Http::kPort]);
httpsSrv->start<HttpsSession>(mINI::Instance()[Config::Http::kSSLPort]);
```
- 作为播放器：
```cpp
MediaPlayer::Ptr player(new MediaPlayer());
weak_ptr<MediaPlayer> weakPlayer = player;
player->setOnPlayResult([weakPlayer](const SockException &ex) {
InfoL << "OnPlayResult:" << ex.what();
auto strongPlayer = weakPlayer.lock();
if (ex || !strongPlayer) {
return;
}
auto viedoTrack = strongPlayer->getTrack(TrackVideo);
if (!viedoTrack) {
WarnL << "没有视频Track!";
return;
}
viedoTrack->addDelegate(std::make_shared<FrameWriterInterfaceHelper>([](const Frame::Ptr &frame) {
//此处解码并播放
}));
});
player->setOnShutdown([](const SockException &ex) {
ErrorL << "OnShutdown:" << ex.what();
});
//支持rtmp、rtsp
(*player)[Client::kRtpType] = Rtsp::RTP_TCP;
player->play("rtsp://admin:jzan123456@192.168.0.122/");
```
- 作为代理服务器：
```cpp
//support rtmp and rtsp url
//just support H264+AAC
auto urlList = {"rtmp://live.hkstv.hk.lxdns.com/live/hks",
"rtsp://184.72.239.149/vod/mp4://BigBuckBunny_175k.mov"};
map<string , PlayerProxy::Ptr> proxyMap;
int i=0;
for(auto url : urlList){
//PlayerProxy构造函数前两个参数分别为应用名（app）,流id（streamId）
//比如说应用为live，流id为0，那么直播地址为：
//http://127.0.0.1/live/0/hls.m3u8
//rtsp://127.0.0.1/live/0
//rtmp://127.0.0.1/live/0
//录像地址为：
//http://127.0.0.1/record/live/0/2017-04-11/11-09-38.mp4
//rtsp://127.0.0.1/record/live/0/2017-04-11/11-09-38.mp4
//rtmp://127.0.0.1/record/live/0/2017-04-11/11-09-38.mp4
PlayerProxy::Ptr player(new PlayerProxy("live",to_string(i++).data()));
player->play(url);
proxyMap.emplace(string(url),player);
}
```
- 作为推流客户端器：
```cpp
PlayerProxy::Ptr player(new PlayerProxy("app","stream"));
//拉一个流，生成一个RtmpMediaSource，源的名称是"app/stream"
//你也可以以其他方式生成RtmpMediaSource，比如说MP4文件（请研读MediaReader代码）
player->play("rtmp://live.hkstv.hk.lxdns.com/live/hks");
RtmpPusher::Ptr pusher;
//监听RtmpMediaSource注册事件,在PlayerProxy播放成功后触发。
NoticeCenter::Instance().addListener(nullptr,Config::Broadcast::kBroadcastRtmpSrcRegisted,
[&pusher](BroadcastRtmpSrcRegistedArgs){
//媒体源"app/stream"已经注册，这时方可新建一个RtmpPusher对象并绑定该媒体源
const_cast<RtmpPusher::Ptr &>(pusher).reset(new RtmpPusher(app,stream));
//推流地址，请改成你自己的服务器。
//这个范例地址（也是基于mediakit）是可用的，但是带宽只有1mb，访问可能很卡顿。
pusher->publish("rtmp://jizan.iok.la/live/test");
});
```
## 参考案例
- [IOS摄像头实时录制,生成rtsp/rtmp/hls/http-flv](https://gitee.com/xiahcu/IOSMedia)
- [IOS rtmp/rtsp播放器，视频推流器](https://gitee.com/xiahcu/IOSPlayer)
- [支持linux、windows、mac的rtmp/rtsp播放器](https://github.com/xiongziliang/ZLMediaPlayer)
- [配套的管理WEB网站](https://github.com/chenxiaolei/ZLMediaKit_NVR_UI)
- [DotNetCore的RESTful客户端](https://github.com/MingZhuLiu/ZLMediaKit.DotNetCore.Sdk)
- [支持GB28181信令服务器、onvif的NVS系统](https://gitee.com/qinqi/JNVS)
## 授权协议
本项目自有代码使用宽松的MIT协议，在保留版权信息的情况下可以自由应用于各自商用、非商业的项目。
......@@ -248,17 +125,21 @@ bash build_docker_images.sh
由于使用本项目而产生的商业纠纷或侵权行为一概与本项项目及开发者无关，请自行承担法律风险。
## 联系方式
- 邮箱：<771730766@qq.com>(本项目相关或流媒体相关问题请走issue流程，否则恕不邮件答复)
- QQ群：542509000
## 怎么提问？
如果要对项目有相关疑问，建议您这么做：
- 1、仔细看下readme、wiki，如果有必要可以查看下issue.
- 2、如果您的问题还没解决，可以提issue.
- 3、有些问题，如果不具备参考性的，无需在issue提的，可以在qq群提.
- 4、QQ私聊一般不接受无偿技术咨询和支持([为什么不提倡QQ私聊](https://github.com/xiongziliang/ZLMediaKit/wiki/%E4%B8%BA%E4%BB%80%E4%B9%88%E4%B8%8D%E5%BB%BA%E8%AE%AEQQ%E7%A7%81%E8%81%8A%E5%92%A8%E8%AF%A2%E9%97%AE%E9%A2%98%EF%BC%9F)).
## 致谢
感谢以下各位对本项目包括但不限于代码贡献、问题反馈、资金捐赠等各种方式的支持！以下排名不分先后：
[老陈](https://github.com/ireader)
......@@ -276,13 +157,12 @@ bash build_docker_images.sh
[linkingvision](https://www.linkingvision.com/)
[茄子](https://github.com/taotaobujue2008)
[好心情](<409257224@qq.com>)
[浮沉](https://github.com/MingZhuLiu)
## 捐赠
欢迎捐赠以便更好的推动项目的发展，谢谢您的支持!
[支付宝](https://raw.githubusercontent.com/xiongziliang/other/master/IMG_3919.JPG)
[微信](https://raw.githubusercontent.com/xiongziliang/other/master/IMG_3920.JPG)
README_en.md
......@@ -20,7 +20,7 @@
- RTSP player and pusher.
- RTP Transport : `rtp over udp` `rtp over tcp` `rtp over http` `rtp udp multicast` .
- Basic/Digest/Url Authentication.
- H265/H264/AAC codec.
- H264/H265/AAC/G711 codec.
- Recorded as mp4.
- Vod of mp4.
......@@ -28,7 +28,7 @@
- RTMP server,support player and pusher.
- RTMP player and pusher.
- Support HTTP-FLV player.
- H264/AAC codec.
- H264/H265/AAC/G711 codec.
- Recorded as flv or mp4.
- Vod of mp4.
- support [RTMP-H265](https://github.com/ksvc/FFmpeg/wiki)
......
api/include/mk_media.h
......@@ -25,11 +25,14 @@ typedef void *mk_media;
* @param app 应用名，推荐为live
* @param stream 流id，例如camera
* @param duration 时长(单位秒)，直播则为0
* @param rtsp_enabled 是否启用rtsp协议
* @param rtmp_enabled 是否启用rtmp协议
* @param hls_enabled 是否生成hls
* @param mp4_enabled 是否生成mp4
* @return 对象指针
*/
API_EXPORT mk_media API_CALL mk_media_create(const char *vhost, const char *app, const char *stream, float duration, int hls_enabled, int mp4_enabled);
API_EXPORT mk_media API_CALL mk_media_create(const char *vhost, const char *app, const char *stream, float duration,
int rtsp_enabled, int rtmp_enabled, int hls_enabled, int mp4_enabled);
/**
* 销毁媒体源
......@@ -38,42 +41,24 @@ API_EXPORT mk_media API_CALL mk_media_create(const char *vhost, const char *app,
API_EXPORT void API_CALL mk_media_release(mk_media ctx);
/**
* 添加h264视频轨道
* 添加视频轨道
* @param ctx 对象指针
* @param track_id 0:CodecH264/1:CodecH265
* @param width 视频宽度
* @param height 视频高度
* @param fps 视频fps
*/
API_EXPORT void API_CALL mk_media_init_h264(mk_media ctx, int width, int height, int fps);
API_EXPORT void API_CALL mk_media_init_video(mk_media ctx, int track_id, int width, int height, int fps);
/**
* 添加h265视频轨道
* 添加音频轨道
* @param ctx 对象指针
* @param width 视频宽度
* @param height 视频高度
* @param fps 视频fps
*/
API_EXPORT void API_CALL mk_media_init_h265(mk_media ctx, int width, int height, int fps);
/**
* 添加aac音频轨道
* @param ctx 对象指针
* @param channel 通道数
* @param sample_bit 采样位数，只支持16
* @param sample_rate 采样率
* @param profile aac编码profile，在不输入adts头时用于生产adts头
*/
API_EXPORT void API_CALL mk_media_init_aac(mk_media ctx, int channel, int sample_bit, int sample_rate, int profile);
/**
* 添加g711音频轨道
* @param ctx 对象指针
* @param au 1.G711A 2.G711U
* @param track_id 2:CodecAAC/3:CodecG711A/4:CodecG711U
* @param channel 通道数
* @param sample_bit 采样位数，只支持16
* @param sample_rate 采样率
*/
API_EXPORT void API_CALL mk_media_init_g711(mk_media ctx, int au, int sample_bit, int sample_rate);
API_EXPORT void API_CALL mk_media_init_audio(mk_media ctx, int track_id, int sample_rate, int channels, int sample_bit);
/**
* 初始化h264/h265/aac完毕后调用此函数，
......@@ -104,16 +89,6 @@ API_EXPORT void API_CALL mk_media_input_h264(mk_media ctx, void *data, int len,
API_EXPORT void API_CALL mk_media_input_h265(mk_media ctx, void *data, int len, uint32_t dts, uint32_t pts);
/**
* 输入单帧AAC音频
* @param ctx 对象指针
* @param data 单帧AAC数据
* @param len 单帧AAC数据字节数
* @param dts 时间戳，毫秒
* @param with_adts_header data中是否包含7个字节的adts头
*/
API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, uint32_t dts, int with_adts_header);
/**
* 输入单帧AAC音频(单独指定adts头)
* @param ctx 对象指针
* @param data 不包含adts头的单帧AAC数据
......@@ -121,7 +96,7 @@ API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, u
* @param dts 时间戳，毫秒
* @param adts adts头
*/
API_EXPORT void API_CALL mk_media_input_aac1(mk_media ctx, void *data, int len, uint32_t dts, void *adts);
API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, uint32_t dts, void *adts);
/**
* 输入单帧G711音频
......
api/include/mk_player.h
......@@ -31,7 +31,7 @@ typedef void(API_CALL *on_mk_play_event)(void *user_data,int err_code,const char
* 收到音视频数据回调
* @param user_data 用户数据指针
* @param track_type 0：视频，1：音频
* @param codec_id 0：H264，1：H265，2：AAC
* @param codec_id 0：H264，1：H265，2：AAC 3.G711A 4.G711U
* @param data 数据指针
* @param len 数据长度
* @param dts 解码时间戳，单位毫秒
......@@ -98,13 +98,15 @@ API_EXPORT void API_CALL mk_player_set_on_shutdown(mk_player ctx, on_mk_play_eve
/**
* 设置音视频数据回调函数
* 该接口只能在播放成功事件触发后才能调用
* 该接口在播放成功事件触发后才有效
* @param ctx 播放器指针
* @param cb 回调函数指针,不得为null
* @param user_data 用户数据指针
*/
API_EXPORT void API_CALL mk_player_set_on_data(mk_player ctx, on_mk_play_data cb, void *user_data);
///////////////////////////获取音视频相关信息接口在播放成功回调触发后才有效///////////////////////////////
/**
* 获取视频codec_id -1：不存在 0：H264，1：H265，2：AAC 3.G711A 4.G711U
* @param ctx 播放器指针
......
api/source/mk_media.cpp
......@@ -96,9 +96,11 @@ API_EXPORT int API_CALL mk_media_total_reader_count(mk_media ctx){
return (*obj)->getChannel()->totalReaderCount();
}
API_EXPORT mk_media API_CALL mk_media_create(const char *vhost, const char *app, const char *stream, float duration, int hls_enabled, int mp4_enabled) {
API_EXPORT mk_media API_CALL mk_media_create(const char *vhost, const char *app, const char *stream, float duration,
int rtsp_enabled, int rtmp_enabled, int hls_enabled, int mp4_enabled) {
assert(vhost && app && stream);
MediaHelper::Ptr *obj(new MediaHelper::Ptr(new MediaHelper(vhost, app, stream, duration, true, true, hls_enabled, mp4_enabled)));
MediaHelper::Ptr *obj(new MediaHelper::Ptr(new MediaHelper(vhost, app, stream, duration,
rtsp_enabled, rtmp_enabled, hls_enabled, mp4_enabled)));
(*obj)->attachEvent();
return (mk_media) obj;
}
......@@ -109,48 +111,25 @@ API_EXPORT void API_CALL mk_media_release(mk_media ctx) {
delete obj;
}
API_EXPORT void API_CALL mk_media_init_h264(mk_media ctx, int width, int height, int frameRate) {
API_EXPORT void API_CALL mk_media_init_video(mk_media ctx, int track_id, int width, int height, int fps){
assert(ctx);
MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
VideoInfo info;
info.iFrameRate = frameRate;
info.codecId = (CodecId)track_id;
info.iFrameRate = fps;
info.iWidth = width;
info.iHeight = height;
(*obj)->getChannel()->initVideo(info);
}
API_EXPORT void API_CALL mk_media_init_h265(mk_media ctx, int width, int height, int frameRate) {
assert(ctx);
MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
VideoInfo info;
info.iFrameRate = frameRate;
info.iWidth = width;
info.iHeight = height;
(*obj)->getChannel()->initH265Video(info);
}
API_EXPORT void API_CALL mk_media_init_aac(mk_media ctx, int channel, int sample_bit, int sample_rate, int profile) {
API_EXPORT void API_CALL mk_media_init_audio(mk_media ctx, int track_id, int sample_rate, int channels, int sample_bit){
assert(ctx);
MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
AudioInfo info;
info.codecId = (CodecId)track_id;
info.iSampleRate = sample_rate;
info.iChannel = channel;
info.iChannel = channels;
info.iSampleBit = sample_bit;
info.iProfile = profile;
(*obj)->getChannel()->initAudio(info);
}
API_EXPORT void API_CALL mk_media_init_g711(mk_media ctx, int au, int sample_bit, int sample_rate)
{
assert(ctx);
MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
AudioInfo info;
info.iSampleRate = sample_rate;
info.iChannel = 1;
info.iSampleBit = sample_bit;
info.iProfile = 0;
info.codecId = (CodecId)au;
(*obj)->getChannel()->initAudio(info);
}
......@@ -172,24 +151,14 @@ API_EXPORT void API_CALL mk_media_input_h265(mk_media ctx, void *data, int len,
(*obj)->getChannel()->inputH265((char *) data, len, dts, pts);
}
API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, uint32_t dts, int with_adts_header) {
assert(ctx && data && len > 0);
MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
(*obj)->getChannel()->inputAAC((char *) data, len, dts, with_adts_header);
}
API_EXPORT void API_CALL mk_media_input_aac1(mk_media ctx, void *data, int len, uint32_t dts, void *adts) {
API_EXPORT void API_CALL mk_media_input_aac(mk_media ctx, void *data, int len, uint32_t dts, void *adts) {
assert(ctx && data && len > 0 && adts);
MediaHelper::Ptr *obj = (MediaHelper::Ptr *) ctx;
(*obj)->getChannel()->inputAAC((char *) data, len, dts, (char *) adts);
}
API_EXPORT void API_CALL mk_media_input_g711(mk_media ctx, void* data, int len, uint32_t dts)
{
API_EXPORT void API_CALL mk_media_input_g711(mk_media ctx, void* data, int len, uint32_t dts){
assert(ctx && data && len > 0);
MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
(*obj)->getChannel()->inputG711((char*)data, len, dts);
}
api/source/mk_player.cpp
......@@ -101,9 +101,7 @@ API_EXPORT void API_CALL mk_player_set_on_data(mk_player ctx, on_mk_play_data cb
});
}
API_EXPORT int API_CALL mk_player_video_codecId(mk_player ctx)
{
API_EXPORT int API_CALL mk_player_video_codecId(mk_player ctx){
assert(ctx);
MediaPlayer::Ptr& player = *((MediaPlayer::Ptr*)ctx);
auto track = dynamic_pointer_cast<VideoTrack>(player->getTrack(TrackVideo));
......@@ -131,9 +129,7 @@ API_EXPORT int API_CALL mk_player_video_fps(mk_player ctx) {
return track ? track->getVideoFps() : 0;
}
API_EXPORT int API_CALL mk_player_audio_codecId(mk_player ctx)
{
API_EXPORT int API_CALL mk_player_audio_codecId(mk_player ctx){
assert(ctx);
MediaPlayer::Ptr& player = *((MediaPlayer::Ptr*)ctx);
auto track = dynamic_pointer_cast<AudioTrack>(player->getTrack(TrackAudio));
......
conf/config.ini
......@@ -42,6 +42,9 @@ publishToRtxp=1
publishToHls=1
#是否默认推流时mp4录像，hook接口(on_publish)中可以覆盖该设置
publishToMP4=0
#合并写缓存大小(单位毫秒)，合并写指服务器缓存一定的数据后才会一次性写入socket，这样能提高性能，但是会提高延时
#在开启低延时模式后，该参数不起作用
mergeWriteMS=300
[hls]
#hls写文件的buf大小，调整参数可以提高文件io性能
......
server/WebApi.cpp
......@@ -38,11 +38,12 @@ using namespace mediakit;
namespace API {
typedef enum {
InvalidArgs = -300,
SqlFailed = -200,
AuthFailed = -100,
OtherFailed = -1,
Success = 0
Exception = -400,//代码抛异常
InvalidArgs = -300,//参数不合法
SqlFailed = -200,//sql执行失败
AuthFailed = -100,//鉴权失败
OtherFailed = -1,//业务代码执行失败，
Success = 0//执行成功
} ApiErr;
#define API_FIELD "api."
......@@ -154,7 +155,8 @@ static inline void addHttpListener(){
HttpSession::KeyValue headerOut;
auto allArgs = getAllArgs(parser);
HttpSession::KeyValue &headerIn = parser.getValues();
headerOut["Content-Type"] = "application/json; charset=utf-8";
GET_CONFIG(string,charSet,Http::kCharSet);
headerOut["Content-Type"] = StrPrinter << "application/json; charset=" << charSet;
if(api_debug){
auto newInvoker = [invoker,parser,allArgs](const string &codeOut,
const HttpSession::KeyValue &headerOut,
......@@ -207,7 +209,7 @@ static inline void addHttpListener(){
}
#endif// ENABLE_MYSQL
catch (std::exception &ex) {
val["code"] = API::OtherFailed;
val["code"] = API::Exception;
val["msg"] = ex.what();
invoker("200 OK", headerOut, val.toStyledString());
}
......@@ -447,9 +449,11 @@ void installWebApi() {
bool flag = src->close(allArgs["force"].as<bool>());
val["result"] = flag ? 0 : -1;
val["msg"] = flag ? "success" : "close failed";
val["code"] = API::OtherFailed;
}else{
val["result"] = -2;
val["msg"] = "can not find the stream";
val["code"] = API::OtherFailed;
}
});
......@@ -725,21 +729,27 @@ void installWebApi() {
api_regist1("/index/api/startRecord",[](API_ARGS1){
CHECK_SECRET();
CHECK_ARGS("type","vhost","app","stream");
val["result"] = Recorder::startRecord((Recorder::type) allArgs["type"].as<int>(),
auto result = Recorder::startRecord((Recorder::type) allArgs["type"].as<int>(),
allArgs["vhost"],
allArgs["app"],
allArgs["stream"],
allArgs["customized_path"]);
val["result"] = result;
val["code"] = result ? API::Success : API::OtherFailed;
val["msg"] = result ? "success" : "start record failed";
});
// 停止录制hls或MP4
api_regist1("/index/api/stopRecord",[](API_ARGS1){
CHECK_SECRET();
CHECK_ARGS("type","vhost","app","stream");
val["result"] = Recorder::stopRecord((Recorder::type) allArgs["type"].as<int>(),
auto result = Recorder::stopRecord((Recorder::type) allArgs["type"].as<int>(),
allArgs["vhost"],
allArgs["app"],
allArgs["stream"]);
val["result"] = result;
val["code"] = result ? API::Success : API::OtherFailed;
val["msg"] = result ? "success" : "stop record failed";
});
// 获取hls或MP4录制状态
......@@ -802,12 +812,10 @@ void installWebApi() {
api_regist1("/index/hook/on_play",[](API_ARGS1){
//开始播放事件
throw SuccessException();
});
api_regist1("/index/hook/on_flow_report",[](API_ARGS1){
//流量统计hook api
throw SuccessException();
});
api_regist1("/index/hook/on_rtsp_realm",[](API_ARGS1){
......@@ -827,7 +835,6 @@ void installWebApi() {
api_regist1("/index/hook/on_stream_changed",[](API_ARGS1){
//媒体注册或反注册事件
throw SuccessException();
});
......@@ -890,12 +897,10 @@ void installWebApi() {
api_regist1("/index/hook/on_record_mp4",[](API_ARGS1){
//录制mp4分片完毕事件
throw SuccessException();
});
api_regist1("/index/hook/on_shell_login",[](API_ARGS1){
//shell登录调试事件
throw SuccessException();
});
api_regist1("/index/hook/on_stream_none_reader",[](API_ARGS1){
......@@ -931,7 +936,6 @@ void installWebApi() {
api_regist1("/index/hook/on_server_started",[](API_ARGS1){
//服务器重启报告
throw SuccessException();
});
......
src/Common/Device.cpp
......@@ -8,13 +8,9 @@
* may be found in the AUTHORS file in the root of the source tree.
*/
#include <stdio.h>
#include <stdio.h>
#include "Device.h"
#include "Util/logger.h"
#include "Util/util.h"
#include "Util/base64.h"
#include "Util/TimeTicker.h"
#include "Extension/AAC.h"
#include "Extension/G711.h"
#include "Extension/H264.h"
......@@ -24,15 +20,15 @@ using namespace toolkit;
namespace mediakit {
DevChannel::DevChannel(const string &strVhost,
const string &strApp,
const string &strId,
float fDuration,
bool bEanbleRtsp,
bool bEanbleRtmp,
bool bEanbleHls,
bool bEnableMp4) :
MultiMediaSourceMuxer(strVhost, strApp, strId, fDuration, bEanbleRtsp, bEanbleRtmp, bEanbleHls, bEnableMp4) {}
DevChannel::DevChannel(const string &vhost,
const string &app,
const string &stream_id,
float duration,
bool enable_rtsp,
bool enable_rtmp,
bool enable_hls,
bool enable_mp4) :
MultiMediaSourceMuxer(vhost, app, stream_id, duration, enable_rtsp, enable_rtmp, enable_hls, enable_mp4) {}
DevChannel::~DevChannel() {}
......@@ -68,14 +64,14 @@ void DevChannel::inputPCM(char* pcData, int iDataLen, uint32_t uiStamp) {
if (_pAacEnc) {
unsigned char *pucOut;
int iRet = _pAacEnc->inputData(pcData, iDataLen, &pucOut);
if (iRet > 0) {
inputAAC((char *) pucOut, iRet, uiStamp);
if (iRet > 7) {
inputAAC((char *) pucOut + 7, iRet - 7, uiStamp, (char *)pucOut);
}
}
}
#endif //ENABLE_FAAC
void DevChannel::inputH264(const char* pcData, int iDataLen, uint32_t dts,uint32_t pts) {
void DevChannel::inputH264(const char *data, int len, uint32_t dts, uint32_t pts) {
if(dts == 0){
dts = (uint32_t)_aTicker[0].elapsedTime();
}
......@@ -83,24 +79,27 @@ void DevChannel::inputH264(const char* pcData, int iDataLen, uint32_t dts,uint32
pts = dts;
}
int prefixeSize;
if (memcmp("\x00\x00\x00\x01", pcData, 4) == 0) {
if (memcmp("\x00\x00\x00\x01", data, 4) == 0) {
prefixeSize = 4;
} else if (memcmp("\x00\x00\x01", pcData, 3) == 0) {
} else if (memcmp("\x00\x00\x01", data, 3) == 0) {
prefixeSize = 3;
} else {
prefixeSize = 0;
}
//由于rtmp/hls/mp4需要缓存时间戳相同的帧，
//所以使用FrameNoCacheAble类型的帧反而会在转换成FrameCacheAble时多次内存拷贝
//在此处只拷贝一次，性能开销更低
H264Frame::Ptr frame = std::make_shared<H264Frame>();
frame->_dts = dts;
frame->_pts = pts;
frame->_buffer.assign("\x00\x00\x00\x01",4);
frame->_buffer.append(pcData + prefixeSize, iDataLen - prefixeSize);
frame->_buffer.append(data + prefixeSize, len - prefixeSize);
frame->_prefix_size = 4;
inputFrame(frame);
}
void DevChannel::inputH265(const char* pcData, int iDataLen, uint32_t dts,uint32_t pts) {
void DevChannel::inputH265(const char *data, int len, uint32_t dts, uint32_t pts) {
if(dts == 0){
dts = (uint32_t)_aTicker[0].elapsedTime();
}
......@@ -108,98 +107,81 @@ void DevChannel::inputH265(const char* pcData, int iDataLen, uint32_t dts,uint32
pts = dts;
}
int prefixeSize;
if (memcmp("\x00\x00\x00\x01", pcData, 4) == 0) {
if (memcmp("\x00\x00\x00\x01", data, 4) == 0) {
prefixeSize = 4;
} else if (memcmp("\x00\x00\x01", pcData, 3) == 0) {
} else if (memcmp("\x00\x00\x01", data, 3) == 0) {
prefixeSize = 3;
} else {
prefixeSize = 0;
}
//由于rtmp/hls/mp4需要缓存时间戳相同的帧，
//所以使用FrameNoCacheAble类型的帧反而会在转换成FrameCacheAble时多次内存拷贝
//在此处只拷贝一次，性能开销更低
H265Frame::Ptr frame = std::make_shared<H265Frame>();
frame->_dts = dts;
frame->_pts = pts;
frame->_buffer.assign("\x00\x00\x00\x01",4);
frame->_buffer.append(pcData + prefixeSize, iDataLen - prefixeSize);
frame->_buffer.append(data + prefixeSize, len - prefixeSize);
frame->_prefix_size = 4;
inputFrame(frame);
}
void DevChannel::inputAAC(const char* pcData, int iDataLen, uint32_t uiStamp,bool withAdtsHeader) {
if(withAdtsHeader){
inputAAC(pcData+7,iDataLen-7,uiStamp,pcData);
} else if(_audio) {
inputAAC(pcData,iDataLen,uiStamp,(char *)_adtsHeader);
}
}
class AACFrameCacheAble : public AACFrameNoCacheAble{
public:
template <typename ... ARGS>
AACFrameCacheAble(ARGS && ...args) : AACFrameNoCacheAble(std::forward<ARGS>(args)...){};
virtual ~AACFrameCacheAble() {
delete [] _ptr;
};
void DevChannel::inputAAC(const char *pcDataWithoutAdts,int iDataLen, uint32_t uiStamp,const char *pcAdtsHeader){
if(uiStamp == 0){
uiStamp = (uint32_t)_aTicker[1].elapsedTime();
}
if(pcAdtsHeader + 7 == pcDataWithoutAdts){
inputFrame(std::make_shared<AACFrameNoCacheAble>((char *)pcDataWithoutAdts - 7,iDataLen + 7,uiStamp,0,7));
} else {
char *dataWithAdts = new char[iDataLen + 7];
memcpy(dataWithAdts,pcAdtsHeader,7);
memcpy(dataWithAdts + 7 , pcDataWithoutAdts , iDataLen);
inputFrame(std::make_shared<AACFrameNoCacheAble>(dataWithAdts,iDataLen + 7,uiStamp,0,7));
delete [] dataWithAdts;
bool cacheAble() const override {
return true;
}
}
};
void DevChannel::inputG711(const char* pcData, int iDataLen, uint32_t uiStamp)
{
if (uiStamp == 0) {
uiStamp = (uint32_t)_aTicker[1].elapsedTime();
void DevChannel::inputAAC(const char *data_without_adts, int len, uint32_t dts, const char *adts_header){
if(dts == 0){
dts = (uint32_t)_aTicker[1].elapsedTime();
}
if(adts_header){
if(adts_header + 7 == data_without_adts){
//adts头和帧在一起
inputFrame(std::make_shared<AACFrameNoCacheAble>((char *)data_without_adts - 7, len + 7, dts, 0, 7));
}else{
//adts头和帧不在一起
char *dataWithAdts = new char[len + 7];
memcpy(dataWithAdts, adts_header, 7);
memcpy(dataWithAdts + 7 , data_without_adts , len);
inputFrame(std::make_shared<AACFrameCacheAble>(dataWithAdts, len + 7, dts, 0, 7));
}
}
inputFrame(std::make_shared<G711FrameNoCacheAble>(_audio->codecId, (char*)pcData, iDataLen, uiStamp, 0));
}
void DevChannel::initVideo(const VideoInfo& info) {
_video = std::make_shared<VideoInfo>(info);
addTrack(std::make_shared<H264Track>());
void DevChannel::inputG711(const char *data, int len, uint32_t dts){
if (dts == 0) {
dts = (uint32_t)_aTicker[1].elapsedTime();
}
inputFrame(std::make_shared<G711FrameNoCacheAble>(_audio->codecId, (char*)data, len, dts, 0));
}
void DevChannel::initH265Video(const VideoInfo &info){
void DevChannel::initVideo(const VideoInfo &info) {
_video = std::make_shared<VideoInfo>(info);
addTrack(std::make_shared<H265Track>());
switch (info.codecId){
case CodecH265 : addTrack(std::make_shared<H265Track>()); break;
case CodecH264 : addTrack(std::make_shared<H264Track>()); break;
default: WarnL << "不支持该类型的视频编码类型:" << info.codecId; break;
}
}
void DevChannel::initAudio(const AudioInfo& info) {
_audio = std::make_shared<AudioInfo>(info);
if (info.codecId == CodecAAC)
{
addTrack(std::make_shared<AACTrack>());
AACFrame adtsHeader;
adtsHeader.syncword = 0x0FFF;
adtsHeader.id = 0;
adtsHeader.layer = 0;
adtsHeader.protection_absent = 1;
adtsHeader.profile = info.iProfile;//audioObjectType - 1;
int i = 0;
for (auto rate : samplingFrequencyTable) {
if (rate == info.iSampleRate) {
adtsHeader.sf_index = i;
};
++i;
}
adtsHeader.private_bit = 0;
adtsHeader.channel_configuration = info.iChannel;
adtsHeader.original = 0;
adtsHeader.home = 0;
adtsHeader.copyright_identification_bit = 0;
adtsHeader.copyright_identification_start = 0;
adtsHeader.aac_frame_length = 7;
adtsHeader.adts_buffer_fullness = 2047;
adtsHeader.no_raw_data_blocks_in_frame = 0;
writeAdtsHeader(adtsHeader, _adtsHeader);
}
else if (info.codecId == CodecG711A || info.codecId == CodecG711U)
{
addTrack(std::make_shared<G711Track>(info.codecId, info.iSampleBit, info.iSampleRate));
void DevChannel::initAudio(const AudioInfo &info) {
_audio = std::make_shared<AudioInfo>(info);
switch (info.codecId) {
case CodecAAC : addTrack(std::make_shared<AACTrack>()); break;
case CodecG711A :
case CodecG711U : addTrack(std::make_shared<G711Track>(info.codecId, info.iSampleRate, info.iChannel, info.iSampleBit)); break;
default: WarnL << "不支持该类型的音频编码类型:" << info.codecId; break;
}
}
......
src/Common/Device.h
......@@ -30,105 +30,91 @@ using namespace toolkit;
#include "Codec/H264Encoder.h"
#endif //ENABLE_X264
namespace mediakit {
class VideoInfo {
public:
CodecId codecId = CodecH264;
int iWidth;
int iHeight;
float iFrameRate;
};
class AudioInfo {
public:
CodecId codecId;
CodecId codecId = CodecAAC;
int iChannel;
int iSampleBit;
int iSampleRate;
int iProfile;
};
/**
* 该类已经废弃，保留只为兼容旧代码，请直接使用MultiMediaSourceMuxer类！
* MultiMediaSourceMuxer类的包装，方便初学者使用
*/
class DevChannel : public MultiMediaSourceMuxer{
public:
typedef std::shared_ptr<DevChannel> Ptr;
//fDuration<=0为直播，否则为点播
DevChannel(const string &strVhost,
const string &strApp,
const string &strId,
float fDuration = 0,
bool bEanbleRtsp = true,
bool bEanbleRtmp = true,
bool bEanbleHls = true,
bool bEnableMp4 = false);
DevChannel(const string &vhost,
const string &app,
const string &stream_id,
float duration = 0,
bool enable_rtsp = true,
bool enable_rtmp = true,
bool enable_hls = true,
bool enable_mp4 = false);
virtual ~DevChannel();
/**
* 初始化h264视频Track
* 相当于MultiMediaSourceMuxer::addTrack(H264Track::Ptr );
* @param info
* 初始化视频Track
* 相当于MultiMediaSourceMuxer::addTrack(VideoTrack::Ptr );
* @param info 视频相关信息
*/
void initVideo(const VideoInfo &info);
/**
* 初始化h265视频Track
* @param info
*/
void initH265Video(const VideoInfo &info);
/**
* 初始化aac音频Track
* 相当于MultiMediaSourceMuxer::addTrack(AACTrack::Ptr );
* @param info
* 初始化音频Track
* 相当于MultiMediaSourceMuxer::addTrack(AudioTrack::Ptr );
* @param info 音频相关信息
*/
void initAudio(const AudioInfo &info);
/**
* 输入264帧
* @param pcData 264单帧数据指针
* @param iDataLen 数据指针长度
* @param data 264单帧数据指针
* @param len 数据指针长度
* @param dts 解码时间戳，单位毫秒；等于0时内部会自动生成时间戳
* @param pts 播放时间戳，单位毫秒；等于0时内部会赋值为dts
*/
void inputH264(const char *pcData, int iDataLen, uint32_t dts,uint32_t pts = 0);
void inputH264(const char *data, int len, uint32_t dts, uint32_t pts = 0);
/**
* 输入265帧
* @param pcData 265单帧数据指针
* @param iDataLen 数据指针长度
* @param data 265单帧数据指针
* @param len 数据指针长度
* @param dts 解码时间戳，单位毫秒；等于0时内部会自动生成时间戳
* @param pts 播放时间戳，单位毫秒；等于0时内部会赋值为dts
*/
void inputH265(const char *pcData, int iDataLen, uint32_t dts,uint32_t pts = 0);
void inputH265(const char *data, int len, uint32_t dts, uint32_t pts = 0);
/**
* 输入可能带adts头的aac帧
* @param pcDataWithAdts 可能带adts头的aac帧
* @param iDataLen 帧数据长度
* @param uiStamp 时间戳，单位毫秒，等于0时内部会自动生成时间戳
* @param withAdtsHeader 是否带adts头
* 输入aac帧
* @param data_without_adts 不带adts头的aac帧
* @param len 帧数据长度
* @param dts 时间戳，单位毫秒
* @param adts_header adts头
*/
void inputAAC(const char *pcDataWithAdts, int iDataLen, uint32_t uiStamp, bool withAdtsHeader = true);
/**
* 输入不带adts头的aac帧
* @param pcDataWithoutAdts 不带adts头的aac帧
* @param iDataLen 帧数据长度
* @param uiStamp 时间戳，单位毫秒
* @param pcAdtsHeader adts头
*/
void inputAAC(const char *pcDataWithoutAdts,int iDataLen, uint32_t uiStamp,const char *pcAdtsHeader);
void inputAAC(const char *data_without_adts, int len, uint32_t dts, const char *adts_header);
/**
* G711音频帧
* @param pcData 音频帧
* @param iDataLen 帧数据长度
* @param uiStamp 时间戳，单位毫秒
* @param data 音频帧
* @param len 帧数据长度
* @param dts 时间戳，单位毫秒
*/
void inputG711(const char* pcData, int iDataLen, uint32_t uiStamp);
void inputG711(const char* data, int len, uint32_t dts);
#ifdef ENABLE_X264
/**
* 输入yuv420p视频帧，内部会完成编码并调用inputH264方法
......@@ -151,6 +137,7 @@ public:
#endif //ENABLE_FAAC
private:
#ifdef ENABLE_X264
std::shared_ptr<H264Encoder> _pH264Enc;
#endif //ENABLE_X264
......@@ -160,9 +147,7 @@ private:
#endif //ENABLE_FAAC
std::shared_ptr<VideoInfo> _video;
std::shared_ptr<AudioInfo> _audio;
SmoothTicker _aTicker[2];
uint8_t _adtsHeader[7];
};
} /* namespace mediakit */
......
src/Common/MediaSink.cpp
......@@ -34,23 +34,16 @@ void MediaSink::addTrack(const Track::Ptr &track_in) {
track->addDelegate(std::make_shared<FrameWriterInterfaceHelper>([this](const Frame::Ptr &frame) {
if (_allTrackReady) {
onTrackFrame(frame);
return;
}
else
{
if (frame->getTrackType() == TrackVideo)
{
checkTrackIfReady(nullptr);
if (_allTrackReady) {
onTrackFrame(frame);
}
else
{
ErrorL << " 还有track未准备好，丢帧 codecName: " << frame->getCodecName();
}
}else
ErrorL << " 还有track未准备好，丢帧 codecName: " << frame->getCodecName();
//还有track未准备好，如果是视频的话，如果直接丢帧可能导致丢失I帧
checkTrackIfReady(nullptr);
if (_allTrackReady) {
//运行至这里说明Track状态由未就绪切换为已就绪状态,那么这帧就不应该丢弃
onTrackFrame(frame);
} else if(frame->keyFrame()){
WarnL << "some track is unready，drop key frame of: " << frame->getCodecName();
}
}));
}
......@@ -70,8 +63,8 @@ void MediaSink::inputFrame(const Frame::Ptr &frame) {
if (it == _track_map.end()) {
return;
}
it->second->inputFrame(frame);
checkTrackIfReady(it->second);
it->second->inputFrame(frame);
}
void MediaSink::checkTrackIfReady_l(const Track::Ptr &track){
......@@ -140,7 +133,7 @@ void MediaSink::emitAllTrackReady() {
//移除未准备好的Track
for (auto it = _track_map.begin(); it != _track_map.end();) {
if (!it->second->ready()) {
WarnL << "该track长时间未被初始化,已忽略:" << it->second->getCodecName();
WarnL << "track not ready for a long time, ignored: " << it->second->getCodecName();
it = _track_map.erase(it);
continue;
}
......
src/Common/MediaSource.cpp
......@@ -450,4 +450,50 @@ MediaSource::Ptr MediaSource::createFromMP4(const string &schema, const string &
#endif //ENABLE_MP4
}
static bool isFlushAble_default(bool is_audio, uint32_t last_stamp, uint32_t new_stamp, int cache_size) {
if (new_stamp < last_stamp) {
//时间戳回退(可能seek中)
return true;
}
if (!is_audio) {
//这是视频,时间戳发送变化或者缓存超过1024个
return last_stamp != new_stamp || cache_size >= 1024;
}
//这是音频,缓存超过100ms或者缓存个数超过10个
return new_stamp > last_stamp + 100 || cache_size > 10;
}
static bool isFlushAble_merge(bool is_audio, uint32_t last_stamp, uint32_t new_stamp, int cache_size, int merge_ms) {
if (new_stamp < last_stamp) {
//时间戳回退(可能seek中)
return true;
}
if(new_stamp > last_stamp + merge_ms){
//时间戳增量超过合并写阈值
return true;
}
if (!is_audio) {
//这是视频,缓存数超过1024个,这个逻辑用于避免时间戳异常的流导致的内存暴增问题
//而且sendmsg接口一般最多只能发送1024个数据包
return cache_size >= 1024;
}
//这是音频，音频缓存超过20个
return cache_size > 20;
}
bool FlushPolicy::isFlushAble(uint32_t last_stamp, uint32_t new_stamp, int cache_size) {
GET_CONFIG(bool,ultraLowDelay, General::kUltraLowDelay);
GET_CONFIG(int,mergeWriteMS, General::kMergeWriteMS);
if(ultraLowDelay || mergeWriteMS <= 0){
//关闭了合并写或者合并写阈值小于等于0
return isFlushAble_default(_is_audio, last_stamp, new_stamp, cache_size);
}
return isFlushAble_merge(_is_audio, last_stamp, new_stamp, cache_size,mergeWriteMS);
}
} /* namespace mediakit */
\ No newline at end of file
src/Common/MediaSource.h
......@@ -21,6 +21,9 @@
#include "Util/logger.h"
#include "Util/TimeTicker.h"
#include "Util/NoticeCenter.h"
#include "Util/List.h"
#include "Rtsp/Rtsp.h"
#include "Rtmp/Rtmp.h"
#include "Extension/Track.h"
#include "Record/Recorder.h"
......@@ -153,6 +156,114 @@ private:
static recursive_mutex g_mtxMediaSrc;
};
///缓存刷新策略类
class FlushPolicy {
public:
FlushPolicy(bool is_audio) {
_is_audio = is_audio;
};
~FlushPolicy() = default;
uint32_t getStamp(const RtpPacket::Ptr &packet) {
return packet->timeStamp;
}
uint32_t getStamp(const RtmpPacket::Ptr &packet) {
return packet->timeStamp;
}
bool isFlushAble(uint32_t last_stamp, uint32_t new_stamp, int cache_size);
private:
bool _is_audio;
};
/// 视频合并写缓存模板
/// \tparam packet 包类型
/// \tparam policy 刷新缓存策略
/// \tparam packet_list 包缓存类型
template<typename packet, typename policy = FlushPolicy, typename packet_list = List<std::shared_ptr<packet> > >
class VideoPacketCache {
public:
VideoPacketCache() : _policy(true) {
_cache = std::make_shared<packet_list>();
}
virtual ~VideoPacketCache() = default;
void inputVideo(const std::shared_ptr<packet> &rtp, bool key_pos) {
auto new_stamp = _policy.getStamp(rtp);
if (_policy.isFlushAble(_last_stamp, new_stamp, _cache->size())) {
flushAll();
}
//追加数据到最后
_cache->emplace_back(rtp);
_last_stamp = new_stamp;
if (key_pos) {
_key_pos = key_pos;
}
}
virtual void onFlushVideo(std::shared_ptr<packet_list> &, bool key_pos) = 0;
private:
void flushAll() {
if (_cache->empty()) {
return;
}
onFlushVideo(_cache, _key_pos);
_cache = std::make_shared<packet_list>();
_key_pos = false;
}
private:
policy _policy;
std::shared_ptr<packet_list> _cache;
uint32_t _last_stamp = 0;
bool _key_pos = false;
};
/// 音频频合并写缓存模板
/// \tparam packet 包类型
/// \tparam policy 刷新缓存策略
/// \tparam packet_list 包缓存类型
template<typename packet, typename policy = FlushPolicy, typename packet_list = List<std::shared_ptr<packet> > >
class AudioPacketCache {
public:
AudioPacketCache() : _policy(false) {
_cache = std::make_shared<packet_list>();
}
virtual ~AudioPacketCache() = default;
void inputAudio(const std::shared_ptr<packet> &rtp) {
auto new_stamp = _policy.getStamp(rtp);
if (_policy.isFlushAble(_last_stamp, new_stamp, _cache->size())) {
flushAll();
}
//追加数据到最后
_cache->emplace_back(rtp);
_last_stamp = new_stamp;
}
virtual void onFlushAudio(std::shared_ptr<packet_list> &) = 0;
private:
void flushAll() {
if (_cache->empty()) {
return;
}
onFlushAudio(_cache);
_cache = std::make_shared<packet_list>();
}
private:
policy _policy;
std::shared_ptr<packet_list> _cache;
uint32_t _last_stamp = 0;
};
} /* namespace mediakit */
......
src/Common/config.cpp
......@@ -67,6 +67,7 @@ const string kResetWhenRePlay = GENERAL_FIELD"resetWhenRePlay";
const string kPublishToRtxp = GENERAL_FIELD"publishToRtxp";
const string kPublishToHls = GENERAL_FIELD"publishToHls";
const string kPublishToMP4 = GENERAL_FIELD"publishToMP4";
const string kMergeWriteMS = GENERAL_FIELD"mergeWriteMS";
onceToken token([](){
mINI::Instance()[kFlowThreshold] = 1024;
......@@ -79,6 +80,7 @@ onceToken token([](){
mINI::Instance()[kPublishToRtxp] = 1;
mINI::Instance()[kPublishToHls] = 1;
mINI::Instance()[kPublishToMP4] = 0;
mINI::Instance()[kMergeWriteMS] = 300;
},nullptr);
}//namespace General
......@@ -286,6 +288,8 @@ const string kTimeoutMS = "protocol_timeout_ms";
const string kMediaTimeoutMS = "media_timeout_ms";
const string kBeatIntervalMS = "beat_interval_ms";
const string kMaxAnalysisMS = "max_analysis_ms";
const string kBenchmarkMode = "benchmark_mode";
}
} // namespace mediakit
......
src/Common/config.h
......@@ -173,6 +173,9 @@ extern const string kPublishToRtxp ;
extern const string kPublishToHls ;
//是否默认推流时mp4录像，hook接口(on_publish)中可以覆盖该设置
extern const string kPublishToMP4 ;
//合并写缓存大小(单位毫秒)，合并写指服务器缓存一定的数据后才会一次性写入socket，这样能提高性能，但是会提高延时
//在开启低延时模式后，该参数不起作用
extern const string kMergeWriteMS ;
}//namespace General
......@@ -315,6 +318,8 @@ extern const string kMediaTimeoutMS;
extern const string kBeatIntervalMS;
//Track编码格式探测最大时间，单位毫秒，默认2000
extern const string kMaxAnalysisMS;
//是否为性能测试模式，性能测试模式开启后不会解析rtp或rtmp包
extern const string kBenchmarkMode;
}
} // namespace mediakit
......
src/Extension/AAC.cpp
......@@ -98,10 +98,9 @@ void getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel){
iChannel = adts.channel_configuration;
}
Sdp::Ptr AACTrack::getSdp() {
if(!ready()){
WarnL << "AAC Track未准备好";
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<AACSdp>(getAacCfg(),getAudioSampleRate());
......
src/Extension/AACRtmp.cpp
......@@ -102,10 +102,12 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
rtmpPkt->strBuf.clear();
//////////header
//header
uint8_t is_config = false;
rtmpPkt->strBuf.push_back(_ui8AudioFlags);
rtmpPkt->strBuf.push_back(_audio_flv_flags);
rtmpPkt->strBuf.push_back(!is_config);
//aac data
rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
rtmpPkt->bodySize = rtmpPkt->strBuf.size();
......@@ -115,45 +117,18 @@ void AACRtmpEncoder::inputFrame(const Frame::Ptr &frame) {
rtmpPkt->typeId = MSG_AUDIO;
RtmpCodec::inputRtmp(rtmpPkt, false);
}
}
void AACRtmpEncoder::makeAudioConfigPkt() {
makeAdtsHeader(_aac_cfg,*_adts);
int iSampleRate , iChannel , iSampleBit = 16;
getAACInfo(*_adts,iSampleRate,iChannel);
uint8_t flvStereoOrMono = (iChannel > 1);
uint8_t flvSampleRate;
switch (iSampleRate) {
case 48000:
case 44100:
flvSampleRate = 3;
break;
case 24000:
case 22050:
flvSampleRate = 2;
break;
case 12000:
case 11025:
flvSampleRate = 1;
break;
default:
flvSampleRate = 0;
break;
}
uint8_t flvSampleBit = iSampleBit == 16;
uint8_t flvAudioType = FLV_CODEC_AAC;
_ui8AudioFlags = (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
_audio_flv_flags = getAudioRtmpFlags(std::make_shared<AACTrack>(_aac_cfg));
RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
rtmpPkt->strBuf.clear();
//////////header
//header
uint8_t is_config = true;
rtmpPkt->strBuf.push_back(_ui8AudioFlags);
rtmpPkt->strBuf.push_back(_audio_flv_flags);
rtmpPkt->strBuf.push_back(!is_config);
//aac config
rtmpPkt->strBuf.append(_aac_cfg);
rtmpPkt->bodySize = rtmpPkt->strBuf.size();
......
src/Extension/AACRtmp.h
......@@ -79,7 +79,7 @@ public:
private:
void makeAudioConfigPkt();
private:
uint8_t _ui8AudioFlags;
uint8_t _audio_flv_flags;
AACTrack::Ptr _track;
};
......
src/Extension/Factory.cpp
......@@ -48,11 +48,11 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
}
if (strcasecmp(track->_codec.data(), "PCMA") == 0) {
return std::make_shared<G711Track>(CodecG711A);
return std::make_shared<G711Track>(CodecG711A, track->_samplerate, track->_channel, 16);
}
if (strcasecmp(track->_codec.data(), "PCMU") == 0) {
return std::make_shared<G711Track>(CodecG711U);
return std::make_shared<G711Track>(CodecG711U, track->_samplerate, track->_channel, 16);
}
if (strcasecmp(track->_codec.data(), "h264") == 0) {
......@@ -84,34 +84,18 @@ Track::Ptr Factory::getTrackBySdp(const SdpTrack::Ptr &track) {
return std::make_shared<H265Track>(vps,sps,pps,0,0,0);
}
//可以根据传统的payload type 获取编码类型以及采样率等信息
CodecId codec_id = RtpPayload::getCodecId(track->_pt);
switch (codec_id){
case CodecG711A :
case CodecG711U : return std::make_shared<G711Track>(codec_id, track->_samplerate, track->_channel, 16);
default : break;
}
WarnL << "暂不支持该sdp:" << track->getName();
return nullptr;
}
Track::Ptr Factory::getTrackByCodecId(CodecId codecId) {
switch (codecId){
case CodecH264:{
return std::make_shared<H264Track>();
}
case CodecH265:{
return std::make_shared<H265Track>();
}
case CodecAAC:{
return std::make_shared<AACTrack>();
}
case CodecG711A: {
return std::make_shared<G711Track>(CodecG711A);
}
case CodecG711U: {
return std::make_shared<G711Track>(CodecG711U);
}
default:
WarnL << "暂不支持该CodecId:" << codecId;
return nullptr;
}
}
RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
GET_CONFIG(uint32_t,audio_mtu,Rtp::kAudioMtuSize);
GET_CONFIG(uint32_t,video_mtu,Rtp::kVideoMtuSize);
......@@ -135,59 +119,29 @@ RtpCodec::Ptr Factory::getRtpEncoderBySdp(const Sdp::Ptr &sdp) {
auto interleaved = sdp->getTrackType() * 2;
auto codec_id = sdp->getCodecId();
switch (codec_id){
case CodecH264:
return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecH265:
return std::make_shared<H265RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecAAC:
return std::make_shared<AACRtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
default:
WarnL << "暂不支持该CodecId:" << codec_id;
return nullptr;
case CodecH264 : return std::make_shared<H264RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecH265 : return std::make_shared<H265RtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecAAC : return std::make_shared<AACRtpEncoder>(ssrc,mtu,sample_rate,pt,interleaved);
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtpEncoder>(ssrc, mtu, sample_rate, pt, interleaved);
default : WarnL << "暂不支持该CodecId:" << codec_id; return nullptr;
}
}
RtpCodec::Ptr Factory::getRtpDecoderByTrack(const Track::Ptr &track) {
switch (track->getCodecId()){
case CodecH264:
return std::make_shared<H264RtpDecoder>();
case CodecH265:
return std::make_shared<H265RtpDecoder>();
case CodecAAC:
return std::make_shared<AACRtpDecoder>(track->clone());
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtpDecoder>(track->clone());
default:
WarnL << "暂不支持该CodecId:" << track->getCodecName();
return nullptr;
case CodecH264 : return std::make_shared<H264RtpDecoder>();
case CodecH265 : return std::make_shared<H265RtpDecoder>();
case CodecAAC : return std::make_shared<AACRtpDecoder>(track->clone());
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtpDecoder>(track->clone());
default : WarnL << "暂不支持该CodecId:" << track->getCodecName(); return nullptr;
}
}
/////////////////////////////rtmp相关///////////////////////////////////////////
Track::Ptr Factory::getVideoTrackByAmf(const AMFValue &amf) {
CodecId codecId = getCodecIdByAmf(amf);
if(codecId == CodecInvalid){
return nullptr;
}
return getTrackByCodecId(codecId);
}
mediakit::Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf)
{
CodecId codecId = getAudioCodecIdByAmf(amf);
if (codecId == CodecInvalid) {
return nullptr;
}
return getTrackByCodecId(codecId);
}
CodecId Factory::getCodecIdByAmf(const AMFValue &val){
static CodecId getVideoCodecIdByAmf(const AMFValue &val){
if (val.type() == AMF_STRING){
auto str = val.as_string();
if(str == "avc1"){
......@@ -209,20 +163,34 @@ CodecId Factory::getCodecIdByAmf(const AMFValue &val){
case FLV_CODEC_H264: return CodecH264;
case FLV_CODEC_AAC: return CodecAAC;
case FLV_CODEC_H265: return CodecH265;
default:
WarnL << "暂不支持该Amf:" << type_id;
return CodecInvalid;
default : WarnL << "暂不支持该Amf:" << type_id; return CodecInvalid;
}
}else{
WarnL << "Metadata不存在相应的Track";
}
return CodecInvalid;
}
CodecId Factory::getAudioCodecIdByAmf(const AMFValue& val)
{
Track::Ptr getTrackByCodecId(CodecId codecId, int sample_rate = 0, int channels = 0, int sample_bit = 0) {
switch (codecId){
case CodecH264 : return std::make_shared<H264Track>();
case CodecH265 : return std::make_shared<H265Track>();
case CodecAAC : return std::make_shared<AACTrack>();
case CodecG711A :
case CodecG711U : return (sample_rate && channels && sample_bit) ? std::make_shared<G711Track>(codecId, sample_rate, channels, sample_bit) : nullptr;
default : WarnL << "暂不支持该CodecId:" << codecId; return nullptr;
}
}
Track::Ptr Factory::getVideoTrackByAmf(const AMFValue &amf) {
CodecId codecId = getVideoCodecIdByAmf(amf);
if(codecId == CodecInvalid){
return nullptr;
}
return getTrackByCodecId(codecId);
}
static CodecId getAudioCodecIdByAmf(const AMFValue &val) {
if (val.type() == AMF_STRING) {
auto str = val.as_string();
if (str == "mp4a") {
......@@ -235,41 +203,38 @@ CodecId Factory::getAudioCodecIdByAmf(const AMFValue& val)
if (val.type() != AMF_NULL) {
auto type_id = val.as_integer();
switch (type_id) {
case FLV_CODEC_AAC: return CodecAAC;
case FLV_CODEC_G711A: return CodecG711A;
case FLV_CODEC_G711U: return CodecG711U;
default:
WarnL << "暂不支持该Amf:" << type_id;
return CodecInvalid;
case FLV_CODEC_AAC : return CodecAAC;
case FLV_CODEC_G711A : return CodecG711A;
case FLV_CODEC_G711U : return CodecG711U;
default : WarnL << "暂不支持该Amf:" << type_id; return CodecInvalid;
}
}
else {
WarnL << "Metadata不存在相应的Track";
}
return CodecInvalid;
}
Track::Ptr Factory::getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit){
CodecId codecId = getAudioCodecIdByAmf(amf);
if (codecId == CodecInvalid) {
return nullptr;
}
return getTrackByCodecId(codecId, sample_rate, channels, sample_bit);
}
RtmpCodec::Ptr Factory::getRtmpCodecByTrack(const Track::Ptr &track) {
switch (track->getCodecId()){
case CodecH264:
return std::make_shared<H264RtmpEncoder>(track);
case CodecAAC:
return std::make_shared<AACRtmpEncoder>(track);
case CodecH265:
return std::make_shared<H265RtmpEncoder>(track);
case CodecG711A:
case CodecG711U:
return std::make_shared<G711RtmpEncoder>(track);
default:
WarnL << "暂不支持该CodecId:" << track->getCodecName();
return nullptr;
case CodecH264 : return std::make_shared<H264RtmpEncoder>(track);
case CodecAAC : return std::make_shared<AACRtmpEncoder>(track);
case CodecH265 : return std::make_shared<H265RtmpEncoder>(track);
case CodecG711A :
case CodecG711U : return std::make_shared<G711RtmpEncoder>(track);
default : WarnL << "暂不支持该CodecId:" << track->getCodecName(); return nullptr;
}
}
AMFValue Factory::getAmfByCodecId(CodecId codecId) {
switch (codecId){
//此处用string标明rtmp编码类型目的是为了兼容某些android系统
case CodecAAC: return AMFValue("mp4a");
case CodecH264: return AMFValue("avc1");
case CodecH265: return AMFValue(FLV_CODEC_H265);
......@@ -279,6 +244,5 @@ AMFValue Factory::getAmfByCodecId(CodecId codecId) {
}
}
}//namespace mediakit
src/Extension/Factory.h
......@@ -24,14 +24,6 @@ namespace mediakit{
class Factory {
public:
/**
* 根据CodecId获取Track，该Track的ready()状态一般都为false
* @param codecId 编解码器id
* @return
*/
static Track::Ptr getTrackByCodecId(CodecId codecId);
////////////////////////////////rtsp相关//////////////////////////////////
/**
* 根据sdp生成Track对象
......@@ -41,14 +33,11 @@ public:
/**
* 根据sdp生成rtp编码器
* @param sdp sdp对象
* @return
*/
static RtpCodec::Ptr getRtpEncoderBySdp(const Sdp::Ptr &sdp);
/**
* 根据Track生成Rtp解包器
* @param track
* @return
*/
static RtpCodec::Ptr getRtpDecoderByTrack(const Track::Ptr &track);
......@@ -58,43 +47,23 @@ public:
/**
* 根据amf对象获取视频相应的Track
* @param amf rtmp metadata中的videocodecid的值
* @return
*/
static Track::Ptr getVideoTrackByAmf(const AMFValue &amf);
/**
* 根据amf对象获取音频相应的Track
* @param amf rtmp metadata中的audiocodecid的值
* @return
*/
static Track::Ptr getAudioTrackByAmf(const AMFValue& amf);
/**
* 根据amf对象获取相应的CodecId
* @param val rtmp metadata中的videocodecid或audiocodecid的值
* @return
*/
static CodecId getCodecIdByAmf(const AMFValue &val);
/**
* 根据amf对象获取音频相应的CodecId
* @param val rtmp metadata中的audiocodecid的值
* @return
*/
static CodecId getAudioCodecIdByAmf(const AMFValue& val);
static Track::Ptr getAudioTrackByAmf(const AMFValue& amf, int sample_rate, int channels, int sample_bit);
/**
* 根据Track获取Rtmp的编解码器
* @param track 媒体描述对象
* @return
*/
static RtmpCodec::Ptr getRtmpCodecByTrack(const Track::Ptr &track);
/**
* 根据codecId获取rtmp的codec描述
* @param codecId
* @return
*/
static AMFValue getAmfByCodecId(CodecId codecId);
};
......
src/Extension/G711.cpp
......@@ -12,13 +12,12 @@
namespace mediakit{
Sdp::Ptr G711Track::getSdp() {
if(!ready()){
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<G711Sdp>(getCodecId(), getAudioSampleRate(), getCodecId() == CodecG711A ? 8 : 0, getAudioSampleBit());
return std::make_shared<G711Sdp>(getCodecId(), getAudioSampleRate(), getAudioChannel());
}
}//namespace mediakit
......
src/Extension/G711.h
......@@ -16,40 +16,27 @@
namespace mediakit{
class G711Frame;
unsigned const samplingFrequencyTableG711[16] = { 96000, 88200,
64000, 48000,
44100, 32000,
24000, 22050,
16000, 12000,
11025, 8000,
7350, 0, 0, 0 };
void makeAdtsHeader(const string &strAudioCfg,G711Frame &adts);
void writeAdtsHeader(const G711Frame &adts, uint8_t *pcAdts) ;
string makeG711AdtsConfig(const uint8_t *pcAdts);
void getAACInfo(const G711Frame &adts,int &iSampleRate,int &iChannel);
/**
* aac帧，包含adts头
* G711帧
*/
class G711Frame : public Frame {
public:
typedef std::shared_ptr<G711Frame> Ptr;
char *data() const override{
return (char *)buffer;
return (char *)buffer.data();
}
uint32_t size() const override {
return frameLength;
return buffer.size();
}
uint32_t dts() const override {
return timeStamp;
}
uint32_t prefixSize() const override{
return iPrefixSize;
return 0;
}
TrackType getTrackType() const override{
......@@ -69,17 +56,15 @@ public:
}
public:
CodecId _codecId = CodecG711A;
unsigned int frameLength; // 一个帧的长度包括 raw data block
unsigned char buffer[2 * 1024 + 7];
string buffer;
uint32_t timeStamp;
uint32_t iPrefixSize = 0;
} ;
class G711FrameNoCacheAble : public FrameNoCacheAble {
public:
typedef std::shared_ptr<G711FrameNoCacheAble> Ptr;
G711FrameNoCacheAble(CodecId codecId, char *ptr,uint32_t size,uint32_t dts,int prefixeSize = 7){
G711FrameNoCacheAble(CodecId codecId, char *ptr,uint32_t size,uint32_t dts,int prefixeSize = 0){
_codecId = codecId;
_ptr = ptr;
_size = size;
......@@ -105,91 +90,61 @@ public:
private:
CodecId _codecId;
} ;
};
/**
* g711音频通道
* G711音频通道
*/
class G711Track : public AudioTrack{
public:
typedef std::shared_ptr<G711Track> Ptr;
/**
* 延后获取adts头信息
* 在随后的inputFrame中获取adts头信息
*/
G711Track(){}
/**
* G711A G711U
*/
G711Track(CodecId codecId, int sampleBit = 16, int sampleRate = 8000){
G711Track(CodecId codecId,int sample_rate, int channels, int sample_bit){
_codecid = codecId;
_sampleBit = sampleBit;
_sampleRate = sampleRate;
onReady();
_sample_rate = sample_rate;
_channels = channels;
_sample_bit = sample_bit;
}
/**
* 返回编码类型
* @return
*/
CodecId getCodecId() const override{
return _codecid;
}
/**
* 在获取aac_cfg前是无效的Track
* @return
* 是否已经初始化
*/
bool ready() override {
return true;
}
/**
* 返回音频采样率
* @return
*/
* 返回音频采样率
*/
int getAudioSampleRate() const override{
return _sampleRate;
return _sample_rate;
}
/**
* 返回音频采样位数，一般为16或8
* @return
*/
int getAudioSampleBit() const override{
return _sampleBit;
return _sample_bit;
}
/**
* 返回音频通道数
* @return
*/
int getAudioChannel() const override{
return _channel;
return _channels;
}
/**
* 输入数据帧,并获取aac_cfg
* @param frame 数据帧
*/
void inputFrame(const Frame::Ptr &frame) override{
AudioTrack::inputFrame(frame);
}
private:
/**
*
*/
void onReady(){
/*
if(_cfg.size() < 2){
return;
}
G711Frame aacFrame;
makeAdtsHeader(_cfg,aacFrame);
getAACInfo(aacFrame,_sampleRate,_channel);*/
}
Track::Ptr clone() override {
return std::make_shared<std::remove_reference<decltype(*this)>::type >(*this);
}
......@@ -197,34 +152,31 @@ private:
//生成sdp
Sdp::Ptr getSdp() override ;
private:
string _cfg;
CodecId _codecid = CodecG711A;
int _sampleRate = 8000;
int _sampleBit = 16;
int _channel = 1;
CodecId _codecid;
int _sample_rate;
int _channels;
int _sample_bit;
};
/**
* aac类型SDP
*/
* G711类型SDP
*/
class G711Sdp : public Sdp {
public:
/**
*
* @param aac_codecId G711A G711U
* G711采样率固定为8000
* @param codecId G711A G711U
* @param sample_rate 音频采样率
* @param playload_type rtp playload type 默认0为G711U， 8为G711A
* @param playload_type rtp playload
* @param bitrate 比特率
*/
G711Sdp(CodecId codecId,
int sample_rate,
int playload_type = 0,
int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
int sample_rate,
int channels,
int playload_type = 98,
int bitrate = 128) : Sdp(sample_rate,playload_type), _codecId(codecId){
_printer << "m=audio 0 RTP/AVP " << playload_type << "\r\n";
//_printer << "b=AS:" << bitrate << "\r\n";
_printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "\r\n";
_printer << "a=rtpmap:" << playload_type << (codecId == CodecG711A ? " PCMA/" : " PCMU/") << sample_rate << "/" << channels << "\r\n";
_printer << "a=control:trackID=" << getTrackType() << "\r\n";
}
......@@ -235,12 +187,14 @@ public:
TrackType getTrackType() const override {
return TrackAudio;
}
CodecId getCodecId() const override {
return _codecId;
}
private:
_StrPrinter _printer;
CodecId _codecId = CodecG711A;
CodecId _codecId;
};
}//namespace mediakit
......
src/Extension/G711Rtmp.cpp
......@@ -12,57 +12,52 @@
namespace mediakit{
G711RtmpDecoder::G711RtmpDecoder() {
_adts = obtainFrame();
G711RtmpDecoder::G711RtmpDecoder(CodecId codecId) {
_frame = obtainFrame();
_codecId = codecId;
}
G711Frame::Ptr G711RtmpDecoder::obtainFrame() {
//从缓存池重新申请对象，防止覆盖已经写入环形缓存的对象
auto frame = ResourcePoolHelper<G711Frame>::obtainObj();
frame->frameLength = 0;
frame->iPrefixSize = 0;
frame->buffer.clear();
frame->_codecId = _codecId;
return frame;
}
bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool key_pos) {
onGetG711(pkt->strBuf.data() + 2, pkt->strBuf.size() - 2, pkt->timeStamp);
bool G711RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt, bool) {
//拷贝G711负载
_frame->buffer.assign(pkt->strBuf.data() + 1, pkt->strBuf.size() - 1);
_frame->timeStamp = pkt->timeStamp;
//写入环形缓存
RtmpCodec::inputFrame(_frame);
_frame = obtainFrame();
return false;
}
void G711RtmpDecoder::onGetG711(const char* pcData, int iLen, uint32_t ui32TimeStamp) {
if(iLen + 7 > sizeof(_adts->buffer)){
WarnL << "Illegal adts data, exceeding the length limit.";
return;
}
//拷贝aac负载
memcpy(_adts->buffer, pcData, iLen);
_adts->frameLength = iLen;
_adts->timeStamp = ui32TimeStamp;
//写入环形缓存
RtmpCodec::inputFrame(_adts);
_adts = obtainFrame();
}
/////////////////////////////////////////////////////////////////////////////////////
G711RtmpEncoder::G711RtmpEncoder(const Track::Ptr &track) {
_track = dynamic_pointer_cast<G711Track>(track);
G711RtmpEncoder::G711RtmpEncoder(const Track::Ptr &track) : G711RtmpDecoder(track->getCodecId()) {
_audio_flv_flags = getAudioRtmpFlags(track);
}
void G711RtmpEncoder::inputFrame(const Frame::Ptr& frame) {
void G711RtmpEncoder::inputFrame(const Frame::Ptr &frame) {
if(!_audio_flv_flags){
return;
}
RtmpPacket::Ptr rtmpPkt = ResourcePoolHelper<RtmpPacket>::obtainObj();
rtmpPkt->strBuf.clear();
rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
//header
rtmpPkt->strBuf.push_back(_audio_flv_flags);
//g711 data
rtmpPkt->strBuf.append(frame->data() + frame->prefixSize(), frame->size() - frame->prefixSize());
rtmpPkt->bodySize = rtmpPkt->strBuf.size();
rtmpPkt->chunkId = CHUNK_AUDIO;
rtmpPkt->streamId = STREAM_MEDIA;
rtmpPkt->timeStamp = frame->dts();
rtmpPkt->typeId = MSG_AUDIO;
RtmpCodec::inputRtmp(rtmpPkt, false);
}
}//namespace mediakit
\ No newline at end of file
src/Extension/G711Rtmp.h
......@@ -17,13 +17,13 @@
namespace mediakit{
/**
* G711 Rtmp转adts类
* G711 Rtmp转G711 Frame类
*/
class G711RtmpDecoder : public RtmpCodec , public ResourcePoolHelper<G711Frame> {
public:
typedef std::shared_ptr<G711RtmpDecoder> Ptr;
G711RtmpDecoder();
G711RtmpDecoder(CodecId codecId);
~G711RtmpDecoder() {}
/**
......@@ -37,48 +37,32 @@ public:
return TrackAudio;
}
void setCodecId(CodecId codecId)
{
_codecid = codecId;
}
CodecId getCodecId() const override{
return _codecid;
return _codecId;
}
protected:
void onGetG711(const char* pcData, int iLen, uint32_t ui32TimeStamp);
private:
G711Frame::Ptr obtainFrame();
protected:
G711Frame::Ptr _adts;
CodecId _codecid = CodecInvalid;
private:
G711Frame::Ptr _frame;
CodecId _codecId;
};
/**
* aac adts转Rtmp类
* G711 RTMP打包类
*/
class G711RtmpEncoder : public G711RtmpDecoder , public ResourcePoolHelper<RtmpPacket> {
public:
typedef std::shared_ptr<G711RtmpEncoder> Ptr;
/**
* 构造函数，track可以为空，此时则在inputFrame时输入adts头
* 如果track不为空且包含adts头相关信息，
* 那么inputFrame时可以不输入adts头
* @param track
*/
G711RtmpEncoder(const Track::Ptr &track);
~G711RtmpEncoder() {}
/**
* 输入aac 数据，可以不带adts头
* @param frame aac数据
* 输入G711 数据
*/
void inputFrame(const Frame::Ptr &frame) override;
private:
G711Track::Ptr _track;
uint8_t _audio_flv_flags = 0;
};
}//namespace mediakit
......
src/Extension/G711Rtp.cpp
......@@ -12,16 +12,63 @@
namespace mediakit{
G711RtpDecoder::G711RtpDecoder(const Track::Ptr &track){
_codecid = track->getCodecId();
_frame = obtainFrame();
}
G711Frame::Ptr G711RtpDecoder::obtainFrame() {
//从缓存池重新申请对象，防止覆盖已经写入环形缓存的对象
auto frame = ResourcePoolHelper<G711Frame>::obtainObj();
frame->buffer.clear();
frame->_codecId = _codecid;
frame->timeStamp = 0;
return frame;
}
bool G711RtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool) {
// 获取rtp数据长度
int length = rtppack->size() - rtppack->offset;
// 获取rtp数据
const char *rtp_packet_buf = rtppack->data() + rtppack->offset;
if (rtppack->timeStamp != _frame->timeStamp) {
//时间戳变更，清空上一帧
onGetG711(_frame);
}
//追加数据
_frame->buffer.append(rtp_packet_buf, length);
//赋值时间戳
_frame->timeStamp = rtppack->timeStamp;
if (rtppack->mark || _frame->buffer.size() > 10 * 1024) {
//标记为mark时，或者内存快溢出时，我们认为这是该帧最后一个包
onGetG711(_frame);
}
return false;
}
void G711RtpDecoder::onGetG711(const G711Frame::Ptr &frame) {
if(!frame->buffer.empty()){
//写入环形缓存
RtpCodec::inputFrame(frame);
_frame = obtainFrame();
}
}
/////////////////////////////////////////////////////////////////////////////////////
G711RtpEncoder::G711RtpEncoder(uint32_t ui32Ssrc,
uint32_t ui32MtuSize,
uint32_t ui32SampleRate,
uint8_t ui8PlayloadType,
uint8_t ui8Interleaved) :
uint32_t ui32MtuSize,
uint32_t ui32SampleRate,
uint8_t ui8PlayloadType,
uint8_t ui8Interleaved) :
RtpInfo(ui32Ssrc,
ui32MtuSize,
ui32SampleRate,
ui8PlayloadType,
ui8Interleaved){
ui8Interleaved) {
}
void G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
......@@ -45,56 +92,9 @@ void G711RtpEncoder::inputFrame(const Frame::Ptr &frame) {
}
void G711RtpEncoder::makeG711Rtp(const void *data, unsigned int len, bool mark, uint32_t uiStamp) {
RtpCodec::inputRtp(makeRtp(getTrackType(),data,len,mark,uiStamp), false);
RtpCodec::inputRtp(makeRtp(getTrackType(), data, len, mark, uiStamp), false);
}
/////////////////////////////////////////////////////////////////////////////////////
G711RtpDecoder::G711RtpDecoder(const Track::Ptr &track){
auto g711Track = dynamic_pointer_cast<G711Track>(track);
_codecid = g711Track->getCodecId();
if(!g711Track || !g711Track->ready()){
WarnL << "该g711 track无效!";
}else{
}
_adts = obtainFrame();
}
G711RtpDecoder::G711RtpDecoder() {
_adts = obtainFrame();
}
G711Frame::Ptr G711RtpDecoder::obtainFrame() {
//从缓存池重新申请对象，防止覆盖已经写入环形缓存的对象
auto frame = ResourcePoolHelper<G711Frame>::obtainObj();
frame->frameLength = 0;
frame->iPrefixSize = 0;
return frame;
}
bool G711RtpDecoder::inputRtp(const RtpPacket::Ptr &rtppack, bool key_pos) {
// 获取rtp数据长度
int length = rtppack->size() - rtppack->offset;
// 获取rtp数据
const uint8_t *rtp_packet_buf = (uint8_t *)rtppack->data() + rtppack->offset;
_adts->frameLength = length;
memcpy(_adts->buffer, rtp_packet_buf, length);
_adts->_codecId = _codecid;
if (rtppack->mark == true) {
_adts->timeStamp = rtppack->timeStamp;
onGetG711(_adts);
}
return false;
}
void G711RtpDecoder::onGetG711(const G711Frame::Ptr &frame) {
//写入环形缓存
RtpCodec::inputFrame(frame);
_adts = obtainFrame();
}
}//namespace mediakit
......
src/Extension/G711Rtp.h
......@@ -10,12 +10,12 @@
#ifndef ZLMEDIAKIT_G711RTPCODEC_H
#define ZLMEDIAKIT_G711RTPCODEC_H
#include "Rtsp/RtpCodec.h"
#include "Extension/G711.h"
namespace mediakit{
/**
* G711 rtp转adts类
* rtp转G711类
*/
class G711RtpDecoder : public RtpCodec , public ResourcePoolHelper<G711Frame> {
public:
......@@ -34,20 +34,23 @@ public:
TrackType getTrackType() const override{
return TrackAudio;
}
CodecId getCodecId() const override{
return _codecid;
}
protected:
G711RtpDecoder();
G711RtpDecoder() {}
private:
void onGetG711(const G711Frame::Ptr &frame);
G711Frame::Ptr obtainFrame();
private:
G711Frame::Ptr _adts;
CodecId _codecid = CodecInvalid;
G711Frame::Ptr _frame;
CodecId _codecid;
};
/**
* g711 转rtp类
*/
......@@ -63,10 +66,10 @@ public:
* @param ui8Interleaved rtsp interleaved 值
*/
G711RtpEncoder(uint32_t ui32Ssrc,
uint32_t ui32MtuSize,
uint32_t ui32SampleRate,
uint8_t ui8PlayloadType = 0,
uint8_t ui8Interleaved = TrackAudio * 2);
uint32_t ui32MtuSize,
uint32_t ui32SampleRate,
uint8_t ui8PlayloadType = 0,
uint8_t ui8Interleaved = TrackAudio * 2);
~G711RtpEncoder() {}
/**
......@@ -75,8 +78,6 @@ public:
void inputFrame(const Frame::Ptr &frame) override;
private:
void makeG711Rtp(const void *pData, unsigned int uiLen, bool bMark, uint32_t uiStamp);
private:
unsigned char _aucSectionBuf[1600];
};
}//namespace mediakit
......
src/Extension/H264.cpp
......@@ -62,7 +62,7 @@ void splitH264(const char *ptr, int len, const std::function<void(const char *,
Sdp::Ptr H264Track::getSdp() {
if(!ready()){
WarnL << "H264 Track未准备好";
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<H264Sdp>(getSps(),getPps());
......
src/Extension/H265.cpp
......@@ -52,7 +52,7 @@ bool getHEVCInfo(const string &strVps, const string &strSps, int &iVideoWidth, i
Sdp::Ptr H265Track::getSdp() {
if(!ready()){
WarnL << "H265 Track未准备好";
WarnL << getCodecName() << " Track未准备好";
return nullptr;
}
return std::make_shared<H265Sdp>(getVps(),getSps(),getPps());
......
src/Http/HttpSession.cpp
......@@ -615,20 +615,29 @@ void HttpSession::setSocketFlags(){
}
}
void HttpSession::onWrite(const Buffer::Ptr &buffer) {
void HttpSession::onWrite(const Buffer::Ptr &buffer, bool flush) {
if(flush){
//需要flush那么一次刷新缓存
HttpSession::setSendFlushFlag(true);
}
_ticker.resetTime();
if(!_flv_over_websocket){
_ui64TotalBytes += buffer->size();
send(buffer);
return;
}else{
WebSocketHeader header;
header._fin = true;
header._reserved = 0;
header._opcode = WebSocketHeader::BINARY;
header._mask_flag = false;
WebSocketSplitter::encode(header,buffer);
}
WebSocketHeader header;
header._fin = true;
header._reserved = 0;
header._opcode = WebSocketHeader::BINARY;
header._mask_flag = false;
WebSocketSplitter::encode(header,buffer);
if(flush){
//本次刷新缓存后，下次不用刷新缓存
HttpSession::setSendFlushFlag(false);
}
}
void HttpSession::onWebSocketEncodeData(const Buffer::Ptr &buffer){
......
src/Http/HttpSession.h
......@@ -49,7 +49,7 @@ public:
static string urlDecode(const string &str);
protected:
//FlvMuxer override
void onWrite(const Buffer::Ptr &data) override ;
void onWrite(const Buffer::Ptr &data, bool flush) override ;
void onDetach() override;
std::shared_ptr<FlvMuxer> getSharedPtr() override;
......
src/Record/MP4Demuxer.cpp
......@@ -142,7 +142,9 @@ struct Context{
BufferRaw::Ptr buffer;
};
Frame::Ptr MP4Demuxer::readFrame(bool &keyFrame) {
Frame::Ptr MP4Demuxer::readFrame(bool &keyFrame, bool &eof) {
keyFrame = false;
eof = false;
static mov_reader_onread mov_reader_onread = [](void *param, uint32_t track_id, const void *buffer, size_t bytes, int64_t pts, int64_t dts, int flags) {
Context *ctx = (Context *) param;
ctx->pts = pts;
......@@ -162,17 +164,21 @@ Frame::Ptr MP4Demuxer::readFrame(bool &keyFrame) {
Context ctx = {this, 0};
auto ret = mov_reader_read2(_mov_reader.get(), mov_onalloc, mov_reader_onread, &ctx);
switch (ret) {
case 0 :
case 0 : {
eof = true;
return nullptr;
}
case 1 : {
keyFrame = ctx.flags & MOV_AV_FLAG_KEYFREAME;
return makeFrame(ctx.track_id, ctx.buffer, ctx.pts, ctx.dts);
}
default:
default : {
eof = true;
WarnL << "读取mp4文件数据失败:" << ret;
return nullptr;
}
}
}
......
src/Record/MP4Demuxer.h
......@@ -22,7 +22,7 @@ public:
MP4Demuxer(const char *file);
~MP4Demuxer() override;
int64_t seekTo(int64_t stamp_ms);
Frame::Ptr readFrame(bool &keyFrame);
Frame::Ptr readFrame(bool &keyFrame, bool &eof);
vector<Track::Ptr> getTracks(bool trackReady) const override ;
uint64_t getDurationMS() const;
private:
......
src/Record/MP4Reader.cpp
......@@ -48,11 +48,10 @@ MP4Reader::MP4Reader(const string &strVhost,const string &strApp, const string &
bool MP4Reader::readSample() {
bool keyFrame = false;
bool eof = false;
while (true) {
auto frame = _demuxer->readFrame(keyFrame);
while (!eof) {
auto frame = _demuxer->readFrame(keyFrame, eof);
if (!frame) {
eof = true;
break;
continue;
}
_mediaMuxer->inputFrame(frame);
if (frame->dts() > getCurrentStamp()) {
......@@ -122,11 +121,12 @@ bool MP4Reader::seekTo(uint32_t ui32Stamp){
}
//搜索到下一帧关键帧
bool keyFrame = false;
while (true) {
auto frame = _demuxer->readFrame(keyFrame);
bool eof = false;
while (!eof) {
auto frame = _demuxer->readFrame(keyFrame, eof);
if(!frame){
//文件读完了都未找到下一帧关键帧
return false;
continue;
}
if(keyFrame || frame->keyFrame() || frame->configFrame()){
//定位到key帧
......@@ -136,6 +136,7 @@ bool MP4Reader::seekTo(uint32_t ui32Stamp){
return true;
}
}
return false;
}
bool MP4Reader::close(MediaSource &sender,bool force){
......
src/Rtmp/FlvMuxer.cpp
......@@ -50,12 +50,17 @@ void FlvMuxer::start(const EventPoller::Ptr &poller,const RtmpMediaSource::Ptr &
}
strongSelf->onDetach();
});
_ring_reader->setReadCB([weakSelf](const RtmpPacket::Ptr &pkt){
_ring_reader->setReadCB([weakSelf](const RtmpMediaSource::RingDataType &pkt){
auto strongSelf = weakSelf.lock();
if(!strongSelf){
return;
}
strongSelf->onWriteRtmp(pkt);
int i = 0;
int size = pkt->size();
pkt->for_each([&](const RtmpPacket::Ptr &rtmp){
strongSelf->onWriteRtmp(rtmp, ++i == size);
});
});
}
......@@ -84,11 +89,11 @@ void FlvMuxer::onWriteFlvHeader(const RtmpMediaSource::Ptr &mediaSrc) {
}
//flv header
onWrite(std::make_shared<BufferRaw>(flv_file_header, sizeof(flv_file_header) - 1));
onWrite(std::make_shared<BufferRaw>(flv_file_header, sizeof(flv_file_header) - 1), false);
auto size = htonl(0);
//PreviousTagSize0 Always 0
onWrite(std::make_shared<BufferRaw>((char *)&size,4));
onWrite(std::make_shared<BufferRaw>((char *)&size,4), false);
auto &metadata = mediaSrc->getMetaData();
......@@ -97,12 +102,12 @@ void FlvMuxer::onWriteFlvHeader(const RtmpMediaSource::Ptr &mediaSrc) {
//其实metadata没什么用，有些推流器不产生metadata
AMFEncoder invoke;
invoke << "onMetaData" << metadata;
onWriteFlvTag(MSG_DATA, std::make_shared<BufferString>(invoke.data()), 0);
onWriteFlvTag(MSG_DATA, std::make_shared<BufferString>(invoke.data()), 0, false);
}
//config frame
mediaSrc->getConfigFrame([&](const RtmpPacket::Ptr &pkt){
onWriteRtmp(pkt);
onWriteRtmp(pkt, true);
});
}
......@@ -125,29 +130,29 @@ public:
#pragma pack(pop)
#endif // defined(_WIN32)
void FlvMuxer::onWriteFlvTag(const RtmpPacket::Ptr &pkt, uint32_t ui32TimeStamp) {
onWriteFlvTag(pkt->typeId,pkt,ui32TimeStamp);
void FlvMuxer::onWriteFlvTag(const RtmpPacket::Ptr &pkt, uint32_t ui32TimeStamp , bool flush) {
onWriteFlvTag(pkt->typeId,pkt,ui32TimeStamp, flush);
}
void FlvMuxer::onWriteFlvTag(uint8_t ui8Type, const Buffer::Ptr &buffer, uint32_t ui32TimeStamp) {
void FlvMuxer::onWriteFlvTag(uint8_t ui8Type, const Buffer::Ptr &buffer, uint32_t ui32TimeStamp, bool flush) {
RtmpTagHeader header;
header.type = ui8Type;
set_be24(header.data_size, buffer->size());
header.timestamp_ex = (uint8_t) ((ui32TimeStamp >> 24) & 0xff);
set_be24(header.timestamp,ui32TimeStamp & 0xFFFFFF);
//tag header
onWrite(std::make_shared<BufferRaw>((char *)&header, sizeof(header)));
onWrite(std::make_shared<BufferRaw>((char *)&header, sizeof(header)), false);
//tag data
onWrite(buffer);
onWrite(buffer, false);
auto size = htonl((buffer->size() + sizeof(header)));
//PreviousTagSize
onWrite(std::make_shared<BufferRaw>((char *)&size,4));
onWrite(std::make_shared<BufferRaw>((char *)&size,4), flush);
}
void FlvMuxer::onWriteRtmp(const RtmpPacket::Ptr &pkt) {
void FlvMuxer::onWriteRtmp(const RtmpPacket::Ptr &pkt,bool flush) {
int64_t dts_out;
_stamp[pkt->typeId % 2].revise(pkt->timeStamp, 0, dts_out, dts_out);
onWriteFlvTag(pkt, dts_out);
onWriteFlvTag(pkt, dts_out,flush);
}
void FlvMuxer::stop() {
......@@ -187,7 +192,7 @@ void FlvRecorder::startRecord(const EventPoller::Ptr &poller,const RtmpMediaSour
start(poller,media);
}
void FlvRecorder::onWrite(const Buffer::Ptr &data) {
void FlvRecorder::onWrite(const Buffer::Ptr &data, bool flush) {
lock_guard<recursive_mutex> lck(_file_mtx);
if(_file){
fwrite(data->data(),data->size(),1,_file.get());
......
src/Rtmp/FlvMuxer.h
......@@ -27,14 +27,14 @@ public:
void stop();
protected:
void start(const EventPoller::Ptr &poller,const RtmpMediaSource::Ptr &media);
virtual void onWrite(const Buffer::Ptr &data) = 0;
virtual void onWrite(const Buffer::Ptr &data, bool flush) = 0;
virtual void onDetach() = 0;
virtual std::shared_ptr<FlvMuxer> getSharedPtr() = 0;
private:
void onWriteFlvHeader(const RtmpMediaSource::Ptr &media);
void onWriteRtmp(const RtmpPacket::Ptr &pkt);
void onWriteFlvTag(const RtmpPacket::Ptr &pkt, uint32_t ui32TimeStamp);
void onWriteFlvTag(uint8_t ui8Type, const Buffer::Ptr &buffer, uint32_t ui32TimeStamp);
void onWriteRtmp(const RtmpPacket::Ptr &pkt,bool flush);
void onWriteFlvTag(const RtmpPacket::Ptr &pkt, uint32_t ui32TimeStamp, bool flush);
void onWriteFlvTag(uint8_t ui8Type, const Buffer::Ptr &buffer, uint32_t ui32TimeStamp, bool flush);
private:
RtmpMediaSource::RingType::RingReader::Ptr _ring_reader;
//时间戳修整器
......@@ -50,7 +50,7 @@ public:
void startRecord(const EventPoller::Ptr &poller,const string &vhost,const string &app,const string &stream,const string &file_path);
void startRecord(const EventPoller::Ptr &poller,const RtmpMediaSource::Ptr &media,const string &file_path);
private:
virtual void onWrite(const Buffer::Ptr &data) override ;
virtual void onWrite(const Buffer::Ptr &data, bool flush) override ;
virtual void onDetach() override;
virtual std::shared_ptr<FlvMuxer> getSharedPtr() override;
private:
......
src/Rtmp/Rtmp.cpp
......@@ -36,11 +36,68 @@ AudioMeta::AudioMeta(const AudioTrack::Ptr &audio,int datarate){
_metadata.set("audiosamplesize", audio->getAudioSampleBit());
}
if(audio->getAudioChannel() > 0){
_metadata.set("audiochannels", audio->getAudioChannel());
_metadata.set("stereo", audio->getAudioChannel() > 1);
}
_codecId = audio->getCodecId();
_metadata.set("audiocodecid", Factory::getAmfByCodecId(_codecId));
}
uint8_t getAudioRtmpFlags(const Track::Ptr &track){
switch (track->getTrackType()){
case TrackAudio : {
auto audioTrack = dynamic_pointer_cast<AudioTrack>(track);
if (!audioTrack) {
WarnL << "获取AudioTrack失败";
return 0;
}
auto iSampleRate = audioTrack->getAudioSampleRate();
auto iChannel = audioTrack->getAudioChannel();
auto iSampleBit = audioTrack->getAudioSampleBit();
uint8_t flvAudioType ;
switch (track->getCodecId()){
case CodecG711A : flvAudioType = FLV_CODEC_G711A; break;
case CodecG711U : flvAudioType = FLV_CODEC_G711U; break;
case CodecAAC : {
flvAudioType = FLV_CODEC_AAC;
//aac不通过flags获取音频相关信息
iSampleRate = 44100;
iSampleBit = 16;
iChannel = 2;
break;
}
default: WarnL << "该编码格式不支持转换为RTMP: " << track->getCodecName(); return 0;
}
uint8_t flvSampleRate;
switch (iSampleRate) {
case 44100:
flvSampleRate = 3;
break;
case 22050:
flvSampleRate = 2;
break;
case 11025:
flvSampleRate = 1;
break;
case 16000: // nellymoser only
case 8000: // nellymoser only
case 5512: // not MP3
flvSampleRate = 0;
break;
default:
WarnL << "FLV does not support sample rate " << iSampleRate << " ,choose from (44100, 22050, 11025)";
return 0;
}
uint8_t flvStereoOrMono = (iChannel > 1);
uint8_t flvSampleBit = iSampleBit == 16;
return (flvAudioType << 4) | (flvSampleRate << 2) | (flvSampleBit << 1) | flvStereoOrMono;
}
default : return 0;
}
}
}//namespace mediakit
\ No newline at end of file
src/Rtmp/Rtmp.h
......@@ -161,27 +161,26 @@ public:
strBuf = std::move(that.strBuf);
}
bool isVideoKeyFrame() const {
return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME
&& (uint8_t) strBuf[1] == 1;
return typeId == MSG_VIDEO && (uint8_t) strBuf[0] >> 4 == FLV_KEY_FRAME && (uint8_t) strBuf[1] == 1;
}
bool isCfgFrame() const {
return (typeId == MSG_VIDEO || typeId == MSG_AUDIO)
&& (uint8_t) strBuf[1] == 0;
switch (typeId){
case MSG_VIDEO : return strBuf[1] == 0;
case MSG_AUDIO : {
switch (getMediaType()){
case FLV_CODEC_AAC : return strBuf[1] == 0;
default : return false;
}
}
default : return false;
}
}
int getMediaType() const {
switch (typeId) {
case MSG_VIDEO: {
return (uint8_t) strBuf[0] & 0x0F;
}
break;
case MSG_AUDIO: {
return (uint8_t) strBuf[0] >> 4;
}
break;
default:
break;
case MSG_VIDEO : return (uint8_t) strBuf[0] & 0x0F;
case MSG_AUDIO : return (uint8_t) strBuf[0] >> 4;
default : return 0;
}
return 0;
}
int getAudioSampleRate() const {
if (typeId != MSG_AUDIO) {
......@@ -209,8 +208,6 @@ public:
}
};
/**
* rtmp metadata基类，用于描述rtmp格式信息
*/
......@@ -316,6 +313,8 @@ private:
CodecId _codecId;
};
//根据音频track获取flags
uint8_t getAudioRtmpFlags(const Track::Ptr &track);
}//namespace mediakit
......
src/Rtmp/RtmpDemuxer.cpp
......@@ -15,14 +15,55 @@ namespace mediakit {
void RtmpDemuxer::loadMetaData(const AMFValue &val){
try {
makeVideoTrack(val["videocodecid"]);
makeAudioTrack(val["audiocodecid"]);
int audiosamplerate = 0;
int audiochannels = 0;
int audiosamplesize = 0;
const AMFValue *audiocodecid = nullptr;
const AMFValue *videocodecid = nullptr;
val.object_for_each([&](const string &key, const AMFValue &val) {
if (key == "duration") {
_fDuration = val.as_number();
return;
}
if(key == "audiosamplerate"){
audiosamplerate = val.as_integer();
return;
}
if(key == "audiosamplesize"){
audiosamplesize = val.as_integer();
return;
}
if(key == "stereo"){
audiochannels = val.as_boolean() ? 2 : 1;
return;
}
if(key == "videocodecid"){
//找到视频
videocodecid = &val;
return;
}
if(key == "audiocodecid"){
//找到音频
audiocodecid = &val;
return;
}
});
if(videocodecid){
//有视频
makeVideoTrack(*videocodecid);
}
if(audiocodecid){
//有音频
makeAudioTrack(*audiocodecid, audiosamplerate, audiochannels, audiosamplesize);
}
}catch (std::exception &ex){
WarnL << ex.what();
}
......@@ -46,7 +87,7 @@ bool RtmpDemuxer::inputRtmp(const RtmpPacket::Ptr &pkt) {
if(!_tryedGetAudioTrack) {
_tryedGetAudioTrack = true;
auto codec = AMFValue(pkt->getMediaType());
makeAudioTrack(codec);
makeAudioTrack(codec, pkt->getAudioSampleRate(), pkt->getAudioChannel(), pkt->getAudioSampleBit());
}
if(_audioRtmpDecoder){
_audioRtmpDecoder->inputRtmp(pkt, false);
......@@ -69,6 +110,7 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
//设置rtmp解码器代理，生成的frame写入该Track
_videoRtmpDecoder->addDelegate(_videoTrack);
onAddTrack(_videoTrack);
_tryedGetVideoTrack = true;
} else {
//找不到相应的rtmp解码器，该track无效
_videoTrack.reset();
......@@ -76,9 +118,9 @@ void RtmpDemuxer::makeVideoTrack(const AMFValue &videoCodec) {
}
}
void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec,int sample_rate, int channels, int sample_bit) {
//生成Track对象
_audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec));
_audioTrack = dynamic_pointer_cast<AudioTrack>(Factory::getAudioTrackByAmf(audioCodec, sample_rate, channels, sample_bit));
if (_audioTrack) {
//生成rtmpCodec对象以便解码rtmp
_audioRtmpDecoder = Factory::getRtmpCodecByTrack(_audioTrack);
......@@ -86,6 +128,7 @@ void RtmpDemuxer::makeAudioTrack(const AMFValue &audioCodec) {
//设置rtmp解码器代理，生成的frame写入该Track
_audioRtmpDecoder->addDelegate(_audioTrack);
onAddTrack(_audioTrack);
_tryedGetAudioTrack = true;
} else {
//找不到相应的rtmp解码器，该track无效
_audioTrack.reset();
......
src/Rtmp/RtmpDemuxer.h
......@@ -40,7 +40,7 @@ public:
bool inputRtmp(const RtmpPacket::Ptr &pkt);
private:
void makeVideoTrack(const AMFValue &val);
void makeAudioTrack(const AMFValue &val);
void makeAudioTrack(const AMFValue &val, int sample_rate, int channels, int sample_bit);
private:
bool _tryedGetVideoTrack = false;
bool _tryedGetAudioTrack = false;
......
src/Rtmp/RtmpMediaSource.h
......@@ -33,6 +33,9 @@ using namespace toolkit;
#define RTMP_GOP_SIZE 512
namespace mediakit {
typedef VideoPacketCache<RtmpPacket> RtmpVideoCache;
typedef AudioPacketCache<RtmpPacket> RtmpAudioCache;
/**
* rtmp媒体源的数据抽象
* rtmp有关键的三要素，分别是metadata、config帧，普通帧
......@@ -40,10 +43,11 @@ namespace mediakit {
* 只要生成了这三要素，那么要实现rtmp推流、rtmp服务器就很简单了
* rtmp推拉流协议中，先传递metadata，然后传递config帧，然后一直传递普通帧
*/
class RtmpMediaSource : public MediaSource, public RingDelegate<RtmpPacket::Ptr> {
class RtmpMediaSource : public MediaSource, public RingDelegate<RtmpPacket::Ptr>, public RtmpVideoCache, public RtmpAudioCache{
public:
typedef std::shared_ptr<RtmpMediaSource> Ptr;
typedef RingBuffer<RtmpPacket::Ptr> RingType;
typedef std::shared_ptr<List<RtmpPacket::Ptr> > RingDataType;
typedef RingBuffer<RingDataType> RingType;
/**
* 构造函数
......@@ -122,6 +126,9 @@ public:
return;
}
//保存当前时间戳
_track_stamps_map[pkt->typeId] = pkt->timeStamp;
if (!_ring) {
weak_ptr<RtmpMediaSource> weakSelf = dynamic_pointer_cast<RtmpMediaSource>(shared_from_this());
auto lam = [weakSelf](const EventPoller::Ptr &, int size, bool) {
......@@ -142,9 +149,12 @@ public:
regist();
}
}
_track_stamps_map[pkt->typeId] = pkt->timeStamp;
//不存在视频，为了减少缓存延时，那么关闭GOP缓存
_ring->write(pkt, _have_video ? pkt->isVideoKeyFrame() : true);
if(pkt->typeId == MSG_VIDEO){
RtmpVideoCache::inputVideo(pkt, key);
}else{
RtmpAudioCache::inputAudio(pkt);
}
}
/**
......@@ -163,6 +173,25 @@ public:
}
private:
/**
* 批量flush时间戳相同的视频rtmp包时触发该函数
* @param rtmp_list 时间戳相同的rtmp包列表
* @param key_pos 是否包含关键帧
*/
void onFlushVideo(std::shared_ptr<List<RtmpPacket::Ptr> > &rtmp_list, bool key_pos) override {
_ring->write(rtmp_list, key_pos);
}
/**
* 批量flush一定数量的音频rtmp包时触发该函数
* @param rtmp_list rtmp包列表
*/
void onFlushAudio(std::shared_ptr<List<RtmpPacket::Ptr> > &rtmp_list) override{
//只有音频的话，就不存在gop缓存的意义
_ring->write(rtmp_list, !_have_video);
}
/**
* 每次增减消费者都会触发该函数
*/
......@@ -177,7 +206,7 @@ private:
bool _have_video = false;
mutable recursive_mutex _mtx;
AMFValue _metadata;
RingBuffer<RtmpPacket::Ptr>::Ptr _ring;
RingType::Ptr _ring;
unordered_map<int, uint32_t> _track_stamps_map;
unordered_map<int, RtmpPacket::Ptr> _config_frame_map;
};
......
src/Rtmp/RtmpPlayer.cpp
......@@ -118,6 +118,8 @@ void RtmpPlayer::onPlayResult_l(const SockException &ex , bool handshakeComplete
//开始播放阶段
_pPlayTimer.reset();
onPlayResult(ex);
//是否为性能测试模式
_benchmark_mode = (*this)[Client::kBenchmarkMode].as<int>();
} else if (ex) {
//播放成功后异常断开回调
onShutdown(ex);
......@@ -146,6 +148,11 @@ void RtmpPlayer::onConnect(const SockException &err){
}
void RtmpPlayer::onRecv(const Buffer::Ptr &pBuf){
try {
if(_benchmark_mode && !_pPlayTimer){
//在性能测试模式下，如果rtmp握手完毕后，不再解析rtmp包
_mediaTicker.resetTime();
return;
}
onParseRtmp(pBuf->data(), pBuf->size());
} catch (exception &e) {
SockException ex(Err_other, e.what());
......
src/Rtmp/RtmpPlayer.h
......@@ -102,6 +102,8 @@ private:
uint32_t _aiNowStamp[2] = { 0, 0 };
Ticker _aNowStampTicker[2];
bool _metadata_got = false;
//是否为性能测试模式
bool _benchmark_mode = false;
};
} /* namespace mediakit */
......
src/Rtmp/RtmpPusher.cpp
......@@ -200,12 +200,21 @@ inline void RtmpPusher::send_metaData(){
_pRtmpReader = src->getRing()->attach(getPoller());
weak_ptr<RtmpPusher> weakSelf = dynamic_pointer_cast<RtmpPusher>(shared_from_this());
_pRtmpReader->setReadCB([weakSelf](const RtmpPacket::Ptr &pkt){
_pRtmpReader->setReadCB([weakSelf](const RtmpMediaSource::RingDataType &pkt){
auto strongSelf = weakSelf.lock();
if(!strongSelf) {
return;
}
strongSelf->sendRtmp(pkt->typeId, strongSelf->_ui32StreamId, pkt, pkt->timeStamp, pkt->chunkId);
int i = 0;
int size = pkt->size();
strongSelf->setSendFlushFlag(false);
pkt->for_each([&](const RtmpPacket::Ptr &rtmp){
if(++i == size){
strongSelf->setSendFlushFlag(true);
}
strongSelf->sendRtmp(rtmp->typeId, strongSelf->_ui32StreamId, rtmp, rtmp->timeStamp, rtmp->chunkId);
});
});
_pRtmpReader->setDetachCB([weakSelf](){
auto strongSelf = weakSelf.lock();
......
src/Rtmp/RtmpSession.cpp
......@@ -193,7 +193,6 @@ void RtmpSession::onCmd_deleteStream(AMFDecoder &dec) {
throw std::runtime_error(StrPrinter << "Stop publishing" << endl);
}
void RtmpSession::sendPlayResponse(const string &err,const RtmpMediaSource::Ptr &src){
bool authSuccess = err.empty();
bool ok = (src.operator bool() && authSuccess);
......@@ -272,12 +271,23 @@ void RtmpSession::sendPlayResponse(const string &err,const RtmpMediaSource::Ptr
_pRingReader = src->getRing()->attach(getPoller());
weak_ptr<RtmpSession> weakSelf = dynamic_pointer_cast<RtmpSession>(shared_from_this());
_pRingReader->setReadCB([weakSelf](const RtmpPacket::Ptr &pkt) {
_pRingReader->setReadCB([weakSelf](const RtmpMediaSource::RingDataType &pkt) {
auto strongSelf = weakSelf.lock();
if (!strongSelf) {
return;
}
strongSelf->onSendMedia(pkt);
if(strongSelf->_paused){
return;
}
int i = 0;
int size = pkt->size();
strongSelf->setSendFlushFlag(false);
pkt->for_each([&](const RtmpPacket::Ptr &rtmp){
if(++i == size){
strongSelf->setSendFlushFlag(true);
}
strongSelf->onSendMedia(rtmp);
});
});
_pRingReader->setDetachCB([weakSelf]() {
auto strongSelf = weakSelf.lock();
......@@ -393,24 +403,9 @@ void RtmpSession::onCmd_pause(AMFDecoder &dec) {
status.set("code", paused ? "NetStream.Pause.Notify" : "NetStream.Unpause.Notify");
status.set("description", paused ? "Paused stream." : "Unpaused stream.");
sendReply("onStatus", nullptr, status);
//streamBegin
sendUserControl(paused ? CONTROL_STREAM_EOF : CONTROL_STREAM_BEGIN,
STREAM_MEDIA);
if (!_pRingReader) {
throw std::runtime_error("Rtmp not started yet!");
}
if (paused) {
_pRingReader->setReadCB(nullptr);
} else {
weak_ptr<RtmpSession> weakSelf = dynamic_pointer_cast<RtmpSession>(shared_from_this());
_pRingReader->setReadCB([weakSelf](const RtmpPacket::Ptr &pkt) {
auto strongSelf = weakSelf.lock();
if(!strongSelf) {
return;
}
strongSelf->onSendMedia(pkt);
});
}
//streamBegin
sendUserControl(paused ? CONTROL_STREAM_EOF : CONTROL_STREAM_BEGIN, STREAM_MEDIA);
_paused = paused;
}
void RtmpSession::setMetaData(AMFDecoder &dec) {
......
src/Rtmp/RtmpSession.h
......@@ -80,13 +80,14 @@ private:
double _dNowReqID = 0;
bool _set_meta_data = false;
Ticker _ticker;//数据接收时间
RingBuffer<RtmpPacket::Ptr>::RingReader::Ptr _pRingReader;
RtmpMediaSource::RingType::RingReader::Ptr _pRingReader;
std::shared_ptr<RtmpMediaSourceImp> _pPublisherSrc;
std::weak_ptr<RtmpMediaSource> _pPlayerSrc;
//时间戳修整器
Stamp _stamp[2];
//消耗的总流量
uint64_t _ui64TotalBytes = 0;
bool _paused = false;
};
......
src/Rtp/RtpProcess.cpp
......@@ -204,10 +204,10 @@ void RtpProcess::onDecode(int stream,int codecid,int flags,int64_t pts,int64_t d
pts /= 90;
dts /= 90;
_stamps[codecid].revise(dts,pts,dts,pts,false);
_dts = dts;
switch (codecid) {
case STREAM_VIDEO_H264: {
_dts = dts;
if (!_codecid_video) {
//获取到视频
_codecid_video = codecid;
......@@ -232,6 +232,7 @@ void RtpProcess::onDecode(int stream,int codecid,int flags,int64_t pts,int64_t d
}
case STREAM_VIDEO_H265: {
_dts = dts;
if (!_codecid_video) {
//获取到视频
_codecid_video = codecid;
......@@ -254,6 +255,7 @@ void RtpProcess::onDecode(int stream,int codecid,int flags,int64_t pts,int64_t d
}
case STREAM_AUDIO_AAC: {
_dts = dts;
if (!_codecid_audio) {
//获取到音频
_codecid_audio = codecid;
......
src/Rtsp/Rtsp.cpp
......@@ -16,7 +16,7 @@ namespace mediakit{
int RtpPayload::getClockRate(int pt){
switch (pt){
#define SWITCH_CASE(name, type, value, clock_rate, channel) case value : return clock_rate;
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return clock_rate;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default: return 90000;
......@@ -25,7 +25,7 @@ int RtpPayload::getClockRate(int pt){
TrackType RtpPayload::getTrackType(int pt){
switch (pt){
#define SWITCH_CASE(name, type, value, clock_rate, channel) case value : return type;
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return type;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default: return TrackInvalid;
......@@ -34,7 +34,7 @@ TrackType RtpPayload::getTrackType(int pt){
int RtpPayload::getAudioChannel(int pt){
switch (pt){
#define SWITCH_CASE(name, type, value, clock_rate, channel) case value : return channel;
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return channel;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default: return 1;
......@@ -43,13 +43,22 @@ int RtpPayload::getAudioChannel(int pt){
const char * RtpPayload::getName(int pt){
switch (pt){
#define SWITCH_CASE(name, type, value, clock_rate, channel) case value : return #name;
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return #name;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default: return "unknown payload type";
}
}
CodecId RtpPayload::getCodecId(int pt) {
switch (pt) {
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return codec_id;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default : return CodecInvalid;
}
}
static void getAttrSdp(const map<string, string> &attr, _StrPrinter &printer){
const map<string, string>::value_type *ptr = nullptr;
for(auto &pr : attr){
......@@ -70,7 +79,7 @@ static void getAttrSdp(const map<string, string> &attr, _StrPrinter &printer){
string SdpTrack::getName() const{
switch (_pt){
#define SWITCH_CASE(name, type, value, clock_rate, channel) case value : return #name;
#define SWITCH_CASE(name, type, value, clock_rate, channel, codec_id) case value : return #name;
RTP_PT_MAP(SWITCH_CASE)
#undef SWITCH_CASE
default: return _codec;
......@@ -172,6 +181,7 @@ void SdpParser::load(const string &sdp) {
if (4 == sscanf(opt_val.data(), " %15[^ ] %d %15[^ ] %d", type, &port, rtp, &pt)) {
track->_pt = pt;
track->_samplerate = RtpPayload::getClockRate(pt) ;
track->_channel = RtpPayload::getAudioChannel(pt);
track->_type = toTrackType(type);
track->_m = opt_val;
track->_port = port;
......@@ -214,9 +224,14 @@ void SdpParser::load(const string &sdp) {
it = track._attr.find("rtpmap");
if(it != track._attr.end()){
auto rtpmap = it->second;
int pt, samplerate;
int pt, samplerate, channel;
char codec[16] = {0};
if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
if (4 == sscanf(rtpmap.data(), "%d %15[^/]/%d/%d", &pt, codec, &samplerate, &channel)) {
track._pt = pt;
track._codec = codec;
track._samplerate = samplerate;
track._channel = channel;
}else if (3 == sscanf(rtpmap.data(), "%d %15[^/]/%d", &pt, codec, &samplerate)) {
track._pt = pt;
track._codec = codec;
track._samplerate = samplerate;
......
src/Rtsp/Rtsp.h
......@@ -34,33 +34,33 @@ typedef enum {
} eRtpType;
#define RTP_PT_MAP(XX) \
XX(PCMU, TrackAudio, 0, 8000, 1) \
XX(GSM, TrackAudio , 3, 8000, 1) \
XX(G723, TrackAudio, 4, 8000, 1) \
XX(DVI4_8000, TrackAudio, 5, 8000, 1) \
XX(DVI4_16000, TrackAudio, 6, 16000, 1) \
XX(LPC, TrackAudio, 7, 8000, 1) \
XX(PCMA, TrackAudio, 8, 8000, 1) \
XX(G722, TrackAudio, 9, 8000, 1) \
XX(L16_Stereo, TrackAudio, 10, 44100, 2) \
XX(L16_Mono, TrackAudio, 11, 44100, 1) \
XX(QCELP, TrackAudio, 12, 8000, 1) \
XX(CN, TrackAudio, 13, 8000, 1) \
XX(MPA, TrackAudio, 14, 90000, 1) \
XX(G728, TrackAudio, 15, 8000, 1) \
XX(DVI4_11025, TrackAudio, 16, 11025, 1) \
XX(DVI4_22050, TrackAudio, 17, 22050, 1) \
XX(G729, TrackAudio, 18, 8000, 1) \
XX(CelB, TrackVideo, 25, 90000, 1) \
XX(JPEG, TrackVideo, 26, 90000, 1) \
XX(nv, TrackVideo, 28, 90000, 1) \
XX(H261, TrackVideo, 31, 90000, 1) \
XX(MPV, TrackVideo, 32, 90000, 1) \
XX(MP2T, TrackVideo, 33, 90000, 1) \
XX(H263, TrackVideo, 34, 90000, 1) \
XX(PCMU, TrackAudio, 0, 8000, 1, CodecG711U) \
XX(GSM, TrackAudio , 3, 8000, 1, CodecInvalid) \
XX(G723, TrackAudio, 4, 8000, 1, CodecInvalid) \
XX(DVI4_8000, TrackAudio, 5, 8000, 1, CodecInvalid) \
XX(DVI4_16000, TrackAudio, 6, 16000, 1, CodecInvalid) \
XX(LPC, TrackAudio, 7, 8000, 1, CodecInvalid) \
XX(PCMA, TrackAudio, 8, 8000, 1, CodecG711A) \
XX(G722, TrackAudio, 9, 8000, 1, CodecInvalid) \
XX(L16_Stereo, TrackAudio, 10, 44100, 2, CodecInvalid) \
XX(L16_Mono, TrackAudio, 11, 44100, 1, CodecInvalid) \
XX(QCELP, TrackAudio, 12, 8000, 1, CodecInvalid) \
XX(CN, TrackAudio, 13, 8000, 1, CodecInvalid) \
XX(MPA, TrackAudio, 14, 90000, 1, CodecInvalid) \
XX(G728, TrackAudio, 15, 8000, 1, CodecInvalid) \
XX(DVI4_11025, TrackAudio, 16, 11025, 1, CodecInvalid) \
XX(DVI4_22050, TrackAudio, 17, 22050, 1, CodecInvalid) \
XX(G729, TrackAudio, 18, 8000, 1, CodecInvalid) \
XX(CelB, TrackVideo, 25, 90000, 1, CodecInvalid) \
XX(JPEG, TrackVideo, 26, 90000, 1, CodecInvalid) \
XX(nv, TrackVideo, 28, 90000, 1, CodecInvalid) \
XX(H261, TrackVideo, 31, 90000, 1, CodecInvalid) \
XX(MPV, TrackVideo, 32, 90000, 1, CodecInvalid) \
XX(MP2T, TrackVideo, 33, 90000, 1, CodecInvalid) \
XX(H263, TrackVideo, 34, 90000, 1, CodecInvalid) \
typedef enum {
#define ENUM_DEF(name, type, value, clock_rate, channel) PT_ ## name = value,
#define ENUM_DEF(name, type, value, clock_rate, channel, codec_id) PT_ ## name = value,
RTP_PT_MAP(ENUM_DEF)
#undef ENUM_DEF
PT_MAX = 128
......@@ -88,6 +88,7 @@ public:
static TrackType getTrackType(int pt);
static int getAudioChannel(int pt);
static const char *getName(int pt);
static CodecId getCodecId(int pt);
private:
RtpPayload() = delete;
~RtpPayload() = delete;
......@@ -128,6 +129,7 @@ public:
int _pt;
string _codec;
int _samplerate;
int _channel;
string _fmtp;
string _control;
string _control_surffix;
......
src/Rtsp/RtspMediaSource.h
......@@ -30,94 +30,10 @@ using namespace toolkit;
#define RTP_GOP_SIZE 512
namespace mediakit {
class RtpVideoCache {
public:
RtpVideoCache() {
_cache = std::make_shared<List<RtpPacket::Ptr> >();
}
virtual ~RtpVideoCache() = default;
void inputVideoRtp(const RtpPacket::Ptr &rtp, bool key_pos) {
if (_last_rtp_stamp != rtp->timeStamp) {
//时间戳发生变化了
flushAll();
} else if (_cache->size() > RTP_GOP_SIZE) {
//这个逻辑用于避免时间戳异常的流导致的内存暴增问题
flushAll();
}
//追加数据到最后
_cache->emplace_back(rtp);
_last_rtp_stamp = rtp->timeStamp;
if (key_pos) {
_key_pos = key_pos;
}
}
virtual void onFlushVideoRtp(std::shared_ptr<List<RtpPacket::Ptr> > &, bool key_pos) = 0;
typedef VideoPacketCache<RtpPacket> RtpVideoCache;
typedef AudioPacketCache<RtpPacket> RtpAudioCache;
private:
void flushAll() {
if (_cache->empty()) {
return;
}
onFlushVideoRtp(_cache, _key_pos);
_cache = std::make_shared<List<RtpPacket::Ptr> >();
_key_pos = false;
}
private:
std::shared_ptr<List<RtpPacket::Ptr> > _cache;
uint32_t _last_rtp_stamp = 0;
bool _key_pos = false;
};
class RtpAudioCache {
public:
RtpAudioCache() {
_cache = std::make_shared<List<RtpPacket::Ptr> >();
}
virtual ~RtpAudioCache() = default;
void inputAudioRtp(const RtpPacket::Ptr &rtp) {
if (rtp->timeStamp > _last_rtp_stamp + 100) {
//累积了100ms的音频数据
flushAll();
} else if (_cache->size() > 10) {
//或者audio rtp缓存超过10个
flushAll();
}
//追加数据到最后
_cache->emplace_back(rtp);
_last_rtp_stamp = rtp->timeStamp;
}
virtual void onFlushAudioRtp(std::shared_ptr<List<RtpPacket::Ptr> > &) = 0;
private:
void flushAll() {
if (_cache->empty()) {
return;
}
onFlushAudioRtp(_cache);
_cache = std::make_shared<List<RtpPacket::Ptr> >();
}
private:
std::shared_ptr<List<RtpPacket::Ptr> > _cache;
uint32_t _last_rtp_stamp = 0;
};
/**
/**
* rtsp媒体源的数据抽象
* rtsp有关键的两要素，分别是sdp、rtp包
* 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
......@@ -261,9 +177,9 @@ public:
}
if(rtp->type == TrackVideo){
RtpVideoCache::inputVideoRtp(rtp, keyPos);
RtpVideoCache::inputVideo(rtp, keyPos);
}else{
RtpAudioCache::inputAudioRtp(rtp);
RtpAudioCache::inputAudio(rtp);
}
}
......@@ -274,7 +190,7 @@ private:
* @param rtp_list 时间戳相同的rtp包列表
* @param key_pos 是否包含关键帧
*/
void onFlushVideoRtp(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list, bool key_pos) override {
void onFlushVideo(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list, bool key_pos) override {
_ring->write(rtp_list, key_pos);
}
......@@ -282,7 +198,7 @@ private:
* 批量flush一定数量的音频rtp包时触发该函数
* @param rtp_list rtp包列表
*/
void onFlushAudioRtp(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list) override{
void onFlushAudio(std::shared_ptr<List<RtpPacket::Ptr> > &rtp_list) override{
//只有音频的话，就不存在gop缓存的意义
_ring->write(rtp_list, !_have_video);
}
......
src/Rtsp/RtspPlayer.cpp
......@@ -112,6 +112,11 @@ void RtspPlayer::onConnect(const SockException &err){
}
void RtspPlayer::onRecv(const Buffer::Ptr& pBuf) {
if(_benchmark_mode && !_pPlayTimer){
//在性能测试模式下，如果rtsp握手完毕后，不再解析rtp包
_rtpTicker.resetTime();
return;
}
input(pBuf->data(),pBuf->size());
}
void RtspPlayer::onErr(const SockException &ex) {
......@@ -750,6 +755,8 @@ void RtspPlayer::onPlayResult_l(const SockException &ex , bool handshakeComplete
//开始播放阶段
_pPlayTimer.reset();
onPlayResult(ex);
//是否为性能测试模式
_benchmark_mode = (*this)[Client::kBenchmarkMode].as<int>();
} else if (ex) {
//播放成功后异常断开回调
onShutdown(ex);
......
src/Rtsp/RtspPlayer.h
......@@ -139,6 +139,8 @@ private:
//是否为rtsp点播
bool _is_play_back;
//是否为性能测试模式
bool _benchmark_mode = false;
};
} /* namespace mediakit */
......
tests/test_benchmark.cpp
......@@ -55,6 +55,7 @@ int main(int argc, char *argv[]) {
player->setOnShutdown([&](const SockException &ex) {
--alivePlayerCnt;
});
(*player)[kBenchmarkMode] = true;
(*player)[kRtpType] = atoi(argv[4]);
player->play(argv[3]);
playerList.push_back(player);
......
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