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v1_EN_SrsLibrtmp
librtmp is a client side library, seems from rtmpdump.
The use scenarios of librtmp:
- Play or suck RTMP stream: For example rtmpdump, dvr RTMP stream to flv file.
- Publish RTMP stream: Publish RTMP stream to server.
- Use sync block socket: It's ok for client.
- ARM: Can used for linux arm, for some embed device, to publish stream to server.
Note: About the openssl, complex and simple handshake, read RTMP protocol
Note: To cross build srs-librtmp for ARM cpu, read srs-arm
librtmp or srs-librtmp only for client side application, impossible for server side.
SRS provides different librtmp:
- The code of librtmp is hard to maintain.
- The interface of librtmp is hard to use.
- No example, while srs-librtmp provides lots of examples at trunk/research/librtmp.
- Min depends, SRS extract core/kernel/rtmp modules for srs-librtmp.
- Min library requires, srs-librtmp only depends on stdc++.
- NO ST, srs-librtmp does not depends on st.
- Provides bandwidth api, to get the bandwidth data to server, read Bandwidth Test
- Provides tracable log, to get the information on server of client, read Tracable log
In a word, SRS provides more efficient and simple client library srs-librtmp.
编译SRS时,会自动编译srs-librtmp,譬如:
./configure --with-librtmp --without-ssl
编译会生成srs-librtmp和对应的实例。
备注:支持librtmp只需要打开--with-librtmp,但推荐打开--without-ssl,不依赖于ssl,对于一般客户端(不需要模拟flash)足够了。这样srs-librtmp不依赖于任何其他库,在x86/x64/arm等平台都可以编译和运行
备注:就算打开了--with-ssl,srslibrtmp也只提供simple_handshake函数,不提供complex_handshake函数。所以推荐关闭ssl,不依赖于ssl,没有实际的用处。
SRS编译成功后,用户就可以使用这些库开发
srs-librtmp可以只依赖于c++和socket,可以在windows下编译。不过srs没有提供直接编译的方法,可行的方法是:
- 先在linux下编译通过:
./configure --disable-all --with-librtmp && make
- 头文件就是
src/libs/srs_librtmp.hpp
,将以下文件拷贝到windows下编译:
objs/srs_auto_headers.hpp
src/core/*
src/kernel/*
src/rtmp/*
src/libs/*
注意:srs-librtmp客户端推流和抓流,不需要ssl库。代码都是c++/stl,网络部分用的是同步socket。 备注:SRS2.0提供将srs-librtmp导出为一个project或者文件,参考导出srs-librtmp。SRS1.0不支持导出,可以自己合并2.0的修改到1.0。
srs-librtmp提供了一系列接口函数,就数据按照一定格式发送到服务器,或者从服务器读取音视频数据。
数据接口包括:
- 读取数据包:int srs_read_packet(int* type, u_int32_t* timestamp, char** data, int* size)
- 发送数据包:int srs_write_packet(int type, u_int32_t timestamp, char* data, int size)
接口接受的的数据(char* data),音视频数据,格式为flv的Video/Audio数据。参考srs的doc目录的规范文件video_file_format_spec_v10_1.pdf
- 音频数据格式参考:
E.4.2.1 AUDIODATA
,p76,譬如,aac编码的音频数据。 - 视频数据格式参考:
E.4.3.1 VIDEODATA
,p78,譬如,h.264编码的视频数据。 - 脚本数据格式参考:
E.4.4.1 SCRIPTDATA
,p80,譬如,onMetadata,流的信息(宽高,码率,分辨率等)
数据类型(int type)定义如下(E.4.1 FLV Tag
,page 75):
- 音频:8 = audio,宏定义:SRS_RTMP_TYPE_AUDIO
- 视频:9 = video,宏定义:SRS_RTMP_TYPE_VIDEO
- 脚本数据:18 = script data,宏定义:SRS_RTMP_TYPE_SCRIPT
其他的数据,譬如时间戳,都是通过参数接受和发送。
另外,文档其他重要信息:
- flv文件头格式:
E.2 The FLV header
,p74。 - flv文件主体格式:
E.3 The FLV File Body
,p74。 - tag头格式:
E.4.1 FLV Tag
,p75。
使用flv格式的原因:
- flv的格式足够简单。
- ffmpeg也是用的这种格式
- 收到流后加上flv tag header,就可以直接保存为flv文件
- 从flv文件解封装数据后,只要将tag的内容给接口就可以,flv的tag头很简单。
备注:SRS2.0支持直接发送h264裸码流,参考publish h.264 raw data
SRS提供了实例sample,也会在编译srs-librtmp时自动编译:
- research/librtmp/srs_play.c:播放RTMP流实例。
- research/librtmp/srs_publish.c:推送RTMP流实例。
- research/librtmp/srs_ingest_flv.c:读取本地文件并推送RTMP流实例。
- research/librtmp/srs_ingest_rtmp.c:读取RTMP流并推送RTMP流实例。
- research/librtmp/srs_bandwidth_check.c:带宽测试工具。
- research/librtmp/srs_flv_injecter.c:点播FLV关键帧注入文件。
- research/librtmp/srs_flv_parser.c:FLV文件查看工具。
- research/librtmp/srs_ingest_rtmp.c:采集RTMP流,推送RTMP流给SRS。
- research/librtmp/srs_ingest_flv.c:采集FLV文件,推送RTMP流给SRS。
- research/librtmp/srs_detect_rtmp.c:RTMP流检测工具。
启动SRS:
[winlin@dev6 srs]$ make && ./objs/srs -c srs.conf
[2014-03-02 18:28:03.857][trace][1][16] server started, listen at port=1935, fd=4
[2014-03-02 18:28:03.857][trace][2][16] thread cycle start
推流实例:
[winlin@dev6 srs]$ make && ./research/librtmp/srs_publish_ssl
publish rtmp stream to server like FMLE/FFMPEG/Encoder
srs(simple-rtmp-server) client librtmp library.
version: 0.9.9
simple handshake success
connect vhost/app success
publish stream success
sent packet: type=Video, time=40, size=4096
sent packet: type=Video, time=80, size=4096
sent packet: type=Video, time=120, size=4096
sent packet: type=Video, time=160, size=4096
sent packet: type=Video, time=200, size=4096
sent packet: type=Video, time=240, size=4096
备注:推流实例发送的视频数据不是真正的视频数据,实际使用时,譬如从摄像头取出h.264裸码流,需要封装成接口要求的数据,然后调用接口发送出去。
播放实例:
[winlin@dev6 trunk]$ make && ./research/librtmp/srs_play_ssl
suck rtmp stream like rtmpdump
srs(simple-rtmp-server) client librtmp library.
version: 0.9.9
simple handshake success
connect vhost/app success
play stream success
got packet: type=Data, time=0, size=24
got packet: type=Data, time=0, size=44
got packet: type=Video, time=40, size=4096
got packet: type=Video, time=80, size=4096
got packet: type=Video, time=120, size=4096
got packet: type=Video, time=160, size=4096
got packet: type=Video, time=200, size=4096
got packet: type=Video, time=240, size=4096
got packet: type=Video, time=280, size=4096
got packet: type=Video, time=320, size=4096
got packet: type=Video, time=360, size=4096
got packet: type=Video, time=400, size=4096
Winlin 2014.11
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