SRS使用小记
SRS简介
互联网上的两种主要的分发方式:HLS和RTMP。
SRS支持HLS/RTMP两种成熟而且广泛应用的流媒体分发方式。
HLS指Apple的HLS(Http Live Streaming),本身就是Live(直播)的,不过Vod(点播)也能支持。HLS是Apple平台的标准流媒体协议,和RTMP在PC上一样支持得天衣无缝。
SRS直播将RTMP作为基本协议,以各种方式转码为RTMP后输入到SRS,输出为RTMP和HLS,支持广泛的客户端和各种应用场景。
SRS点播还在计划中,不会使用RTMP作为点播协议,点播还是文件为主,即HTTP协议为主。
SRS编译配置参数详解
https://github.com/ossrs/srs/wiki/v2_CN_Build
获取SRS
git clone https://github.com/ossrs/srs
编译SRS
./configure && make -j4
启动SRS
./objs/srs -c conf/srs.conf
cp etc/init.d/srs /etc/init.d/srs
注意修改配置文件
CONFIG="./conf/srs.conf"
配置文件说明
SRS的使用,主要是对配置文件的理解
conf/srs.conf
# main config for srs.
# @see full.conf for detail config.
listen 1935; #监听端口
max_connections 1000;
srs_log_tank file;
srs_log_file ./objs/srs.log;
http_api {
enabled on;
listen 1985;
}
http_server {
enabled on;
listen 8085;
dir ./objs/nginx/html;
}
stats {
network 0;
disk sda sdb xvda xvdb;
}
vhost __defaultVhost__ {
}
---
RTMP Cluster配置
conf/edge.conf
# the config for srs origin-edge cluster
# @see https://github.com/ossrs/srs/wiki/v1_CN_Edge
# @see full.conf for detail config.
listen 19350; #边缘节点监听端口
max_connections 1000;
pid objs/edge.pid;
daemon off;
srs_log_tank console;
vhost __defaultVhost__ {
mode remote;
origin 127.0.0.1:1935; #源节点监听端口
}
---
HTTP FLV Live Stream配置
在vhost下添加http_remux
vhost __defaultVhost__ {
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
hstrs on;
}
}
播放流地址
HTTP FLV: http://192.168.1.158:8085/live/ljd.flv
https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHttpStream#http-live-stream-config
---
HTTP FLV Live Stream Cluster配置
单独节点,区别于在vhost下添加
vhost __defaultVhost__ {
mode remote; # 边缘节点开启
origin 192.168.1.158; #源节点地址
http_remux {
enabled on;
mount [vhost]/[app]/[stream].flv;
hstrs on;
}
---
HLS分发配置
在vhost下添加hls
vhost __defaultVhost__ {
hls {
enabled on;
hls_fragment 10;
hls_window 60;
hls_path ./objs/nginx/html;
hls_m3u8_file [app]/[stream].m3u8;
hls_ts_file [app]/[stream]-[seq].ts;
}
}
HLS流地址为: http://192.168.1.158/live/ljd.m3u8
---
转码HLS后分发
HLS需要h.264+aac,若不符合这个要求则需要转码。
在vhost下添加transcode 转码
vhost __defaultVhost__ {
...
#低延时配置
gop_cache off;
queue_length 10;
min_latency on;
mr {
enabled off;
}
mw_latency 100;
tcp_nodelay on;
#转码HLS
transcode live{
# whether the transcode enabled.
# if off, donot transcode.
# default: off.
enabled on;
# the ffmpeg
ffmpeg /usr/local/ffmpeg/bin/ffmpeg;
# the transcode engine for matched stream.
# all matched stream will transcoded to the following stream.
# the transcode set name(ie. hd) is optional and not used.
engine ff {
# whether the engine is enabled
# default: off.
enabled on;
# input format, can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# default: flv
iformat flv;
# video encoder name. can be:
# libx264: use h.264(libx264) video encoder.
# copy: donot encoder the video stream, copy it.
# vn: disable video output.
vcodec libx264;
# video bitrate, in kbps
# @remark 0 to use source video bitrate.
# default: 0
vbitrate 1500;
# video framerate.
# @remark 0 to use source video fps.
# default: 0
vfps 0;
# video width, must be even numbers.
# @remark 0 to use source video width.
# default: 0
vwidth 1440;
# video height, must be even numbers.
# @remark 0 to use source video height.
# default: 0
vheight 720;
# the max threads for ffmpeg to used.
# default: 1
vthreads 12;
# x264 profile, @see x264 -help, can be:
# high,main,baseline
vprofile main;
# x264 preset, @see x264 -help, can be:
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
vpreset medium;
# other x264 or ffmpeg video params
vparams {
# ffmpeg options, @see: http://ffmpeg.org/ffmpeg.html
t 100;
# 264 params, @see: http://ffmpeg.org/ffmpeg-codecs.html#libx264
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
# audio encoder name. can be:
# libfdk_aac: use aac(libfdk_aac) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
acodec copy;
# audio bitrate, in kbps. [16, 72] for libfdk_aac.
# @remark 0 to use source audio bitrate.
# default: 0
abitrate 70;
# audio sample rate. for flv/rtmp, it must be:
# 44100,22050,11025,5512
# @remark 0 to use source audio sample rate.
# default: 0
asample_rate 44100;
# audio channel, 1 for mono, 2 for stereo.
# @remark 0 to use source audio channels.
# default: 0
achannels 2;
# other ffmpeg audio params
aparams {
# audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
# @remark SRS supported aac profile for HLS is: aac_low, aac_he, aac_he_v2
profile:a aac_low;
bsf:a aac_adtstoasc;
}
# output format, can be:
# off, do not specifies the format, ffmpeg will guess it.
# flv, for flv or RTMP stream.
# other format, for example, mp4/aac whatever.
# default: flv
oformat flv;
# output stream. variables:
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
转码后的RTMP流地址为:rtmp://192.168.1.158/live/ljd_ff
转码后的HLS流地址为: http://192.168.1.158/live/ljd_ff.m3u8
---
SRS支持将RTMP流录制成flv文件
vhost __defaultVhost__ {
dvr {
# whether enabled dvr features
# default: off
enabled on;
# the dvr output path.
# we supports some variables to generate the filename.
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# [2006], replace this const to current year.
# [01], replace this const to current month.
# [02], replace this const to current date.
# [15], replace this const to current hour.
# [04], repleace this const to current minute.
# [05], repleace this const to current second.
# [999], repleace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# @remark we use golang time format "2006-01-02 15:04:05.999"
# for example, for url rtmp://ossrs.net/live/livestream and time 2015-01-03 10:57:30.776
# 1. No variables, the rule of SRS1.0(auto add [stream].[timestamp].flv as filename):
# dvr_path ./objs/nginx/html;
# =>
# dvr_path ./objs/nginx/html/live/livestream.1420254068776.flv;
# 2. Use stream and date as dir name, time as filename:
# dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]/[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/livestream/2015/01/03/10.57.30.776.flv;
# 3. Use stream and year/month as dir name, date and time as filename:
# dvr_path /data/[vhost]/[app]/[stream]/[2006]/[01]/[02]-[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/livestream/2015/01/03-10.57.30.776.flv;
# 4. Use vhost/app and year/month as dir name, stream/date/time as filename:
# dvr_path /data/[vhost]/[app]/[2006]/[01]/[stream]-[02]-[15].[04].[05].[999].flv;
# =>
# dvr_path /data/ossrs.net/live/2015/01/livestream-03-10.57.30.776.flv;
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
# the dvr plan. canbe:
# session reap flv when session end(unpublish).
# segment reap flv when flv duration exceed the specified dvr_duration.
# default: session
dvr_plan session;
# the param for plan(segment), in seconds.
# default: 30
dvr_duration 30;
# the param for plan(segment),
# whether wait keyframe to reap segment,
# if off, reap segment when duration exceed the dvr_duration,
# if on, reap segment when duration exceed and got keyframe.
# default: on
dvr_wait_keyframe on;
# about the stream monotonically increasing:
# 1. video timestamp is monotonically increasing,
# 2. audio timestamp is monotonically increasing,
# 3. video and audio timestamp is interleaved monotonically increasing.
# it's specified by RTMP specification, @see 3. Byte Order, Alignment, and Time Format
# however, some encoder cannot provides this feature, please set this to off to ignore time jitter.
# the time jitter algorithm:
# 1. full, to ensure stream start at zero, and ensure stream monotonically increasing.
# 2. zero, only ensure sttream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# default: full
time_jitter full;
}
}
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