# all config for srs
#############################################################################################
# RTMP sections
#############################################################################################
# the rtmp listen ports, split by space.
listen 1935;
# the pid file
# to ensure only one process can use a pid file
# and provides the current running process id, for script,
# for example, init.d script to manage the server.
# default: ./objs/srs.pid
pid ./objs/srs.pid;
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# however, most clients supports it and it can improve
# performance about 10%.
# default: 60000
chunk_size 60000;
# the logs dir.
# if enabled ffmpeg, each stracoding stream will create a log file.
# /dev/null to disable the log.
# default: ./objs
ff_log_dir ./objs;
# the log tank, console or file.
# if console, print log to console.
# if file, write log to file. requires srs_log_file if log to file.
# default: file.
srs_log_tank file;
# the log level, for all log tanks.
# can be: verbose, info, trace, warn, error
# default: trace
srs_log_level trace;
# when srs_log_tank is file, specifies the log file.
# default: ./objs/srs.log
srs_log_file ./objs/srs.log;
# the max connections.
# if exceed the max connections, server will drop the new connection.
# default: 1000
max_connections 1000;
# whether start as deamon
# @remark: donot support reload.
# default: on
daemon on;
#############################################################################################
# heartbeat/stats sections
#############################################################################################
# heartbeat to api server
# @remark, the ip report to server, is retrieve from system stat,
# which need the config item stats.network.
heartbeat {
# whether heartbeat is enalbed.
# default: off
enabled off;
# the interval seconds for heartbeat,
# recommend 0.3,0.6,0.9,1.2,1.5,1.8,2.1,2.4,2.7,3,...,6,9,12,....
# default: 9.9
interval 9.3;
# when startup, srs will heartbeat to this api.
# @remark: must be a restful http api url, where SRS will POST with following data:
# {
# "device_id": "my-srs-device",
# "ip": "192.168.1.100"
# }
# default: http://127.0.0.1:8085/api/v1/servers
url http://127.0.0.1:8085/api/v1/servers;
# the id of devide.
device_id "my-srs-device";
# whether report with summaries
# if true, put /api/v1/summaries to the request data:
# {
# "summaries": summaries object.
# }
# @remark: optional config.
# default: off
summaries off;
}
# system statistics section.
# the main cycle will retrieve the system stat,
# for example, the cpu/mem/network/disk-io data,
# the http api, for instance, /api/v1/summaries will show these data.
# @remark the heartbeat depends on the network,
# for example, the eth0 maybe the device which index is 0.
stats {
# the index of device ip.
# we may retrieve more than one network device.
# default: 0
network 0;
# the device name to stat the disk iops.
# ignore the device of /proc/diskstats if not configed.
disk sda sdb xvda xvdb;
}
#############################################################################################
# HTTP sections
#############################################################################################
# api of srs.
# the http api config, export for external program to manage srs.
# user can access http api of srs in browser directly, for instance, to access by:
# curl http://192.168.1.170:1985/api/v1/reload
# which will reload srs, like cmd killall -1 srs, but the js can also invoke the http api,
# where the cli can only be used in shell/terminate.
http_api {
# whether http api is enabled.
# default: off
enabled on;
# the http api port
# default: 1985
listen 1985;
}
# embeded http server in srs.
# the http streaming config, for HLS/HDS/DASH/HTTPProgressive
# global config for http streaming, user must config the http section for each vhost.
# the embed http server used to substitute nginx in ./objs/nginx,
# for example, srs runing in arm, can provides RTMP and HTTP service, only with srs installed.
# user can access the http server pages, generally:
# curl http://192.168.1.170:80/srs.html
# which will show srs version and welcome to srs.
# @remark, the http embeded stream need to config the vhost, for instance, the __defaultVhost__
# need to open the feature http of vhost.
http_stream {
# whether http streaming service is enabled.
# default: off
enabled on;
# the http streaming port
# @remark, if use lower port, for instance 80, user must start srs by root.
# default: 8080
listen 8080;
# the default dir for http root.
# default: ./objs/nginx/html
dir ./objs/nginx/html;
}
#############################################################################################
# RTMP/HTTP VHOST sections
#############################################################################################
# vhost list, the __defaultVhost__ is the default vhost
# for example, user use ip to access the stream: rtmp://192.168.1.2/live/livestream.
# for which cannot identify the required vhost.
vhost __defaultVhost__ {
}
# the security to allow or deny clients.
vhost security.srs.com {
# security for host to allow or deny clients.
# @see https://github.com/winlinvip/simple-rtmp-server/issues/211
security {
# whether enable the security for vhost.
# default: off
enabled on;
# the security list, each item format as:
# allow|deny publish|play all|<ip>
# for example:
# allow publish all;
# deny publish all;
# allow publish 127.0.0.1;
# deny publish 127.0.0.1;
# allow play all;
# deny play all;
# allow play 127.0.0.1;
# deny play 127.0.0.1;
# SRS apply the following simple strategies one by one:
# 1. allow all if security disabled.
# 2. default to deny all when security enabled.
# 3. allow if matches allow strategy.
# 4. deny if matches deny strategy.
allow play all;
allow publish all;
}
}
# the MR(merged-read) setting for publisher.
# the MW(merged-write) settings for player.
vhost mrw.srs.com {
# whether enable min delay mode for vhost.
# for min latence mode:
# 1. disable the mr for vhost.
# 2. use timeout for cond wait for consumer queue.
# @see https://github.com/winlinvip/simple-rtmp-server/issues/257
# default: off
min_latency off;
# about MR, read https://github.com/winlinvip/simple-rtmp-server/issues/241
mr {
# whether enable the MR(merged-read)
# default: off
enabled on;
# the latency in ms for MR(merged-read),
# the performance+ when latency+, and memory+,
# memory(buffer) = latency * kbps / 8
# for example, latency=500ms, kbps=3000kbps, each publish connection will consume
# memory = 500 * 3000 / 8 = 187500B = 183KB
# when there are 2500 publisher, the total memory of SRS atleast:
# 183KB * 2500 = 446MB
# the value recomment is [300, 2000]
# default: 350
latency 350;
}
# set the MW(merged-write) latency in ms.
# SRS always set mw on, so we just set the latency value.
# the latency of stream >= mw_latency + mr_latency
# the value recomment is [300, 1800]
# default: 350
mw_latency 350;
}
# vhost for edge, edge and origin is the same vhost
vhost same.edge.srs.com {
# the mode of vhost, local or remote.
# local: vhost is origin vhost, which provides stream source.
# remote: vhost is edge vhost, which pull/push to origin.
# default: local
mode remote;
# for edge(remote mode), user must specifies the origin server
# format as: <server_name|ip>[:port]
# @remark user can specifies multiple origin for error backup, by space,
# for example, 192.168.1.100:1935 192.168.1.101:1935 192.168.1.102:1935
origin 127.0.0.1:1935 localhost:1935;
# for edge, whether open the token traverse mode,
# if token traverse on, all connections of edge will forward to origin to check(auth),
# it's very important for the edge to do the token auth.
# the better way is use http callback to do the token auth by the edge,
# but if user prefer origin check(auth), the token_traverse if better solution.
# default: off
token_traverse off;
}
# vhost for dvr
vhost dvr.srs.com {
# dvr RTMP stream to file,
# start to record to file when encoder publish,
# reap flv according by specified dvr_plan.
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/winlinvip/simple-rtmp-server/wiki/v2_CN_DVR#custom-path
# @see https://github.com/winlinvip/simple-rtmp-server/wiki/v2_EN_DVR#custom-path
# default: ./objs/nginx/html
dvr_path ./objs/nginx/html;
# 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;
# on_dvr
# for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
# @read https://github.com/winlinvip/simple-rtmp-server/wiki/v2_CN_DVR#http-callback
# @read https://github.com/winlinvip/simple-rtmp-server/wiki/v2_EN_DVR#http-callback
}
}
# vhost for ingest
vhost ingest.srs.com {
# ingest file/stream/device then push to SRS over RTMP.
# the name/id used to identify the ingest, must be unique in global.
# ingest id is used in reload or http api management.
ingest livestream {
# whether enabled ingest features
# default: off
enabled on;
# input file/stream/device
# @remark only support one input.
input {
# the type of input.
# can be file/stream/device, that is,
# file: ingest file specifies by url.
# stream: ingest stream specifeis by url.
# device: not support yet.
# default: file
type file;
# the url of file/stream.
url ./doc/source.200kbps.768x320.flv;
}
# the ffmpeg
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
# the transcode engine, @see all.transcode.srs.com
# @remark, the output is specified following.
engine {
# @see enabled of transcode engine.
# if disabled or vcodec/acodec not specified, use copy.
# default: off.
enabled off;
# output stream. variables:
# [vhost] current vhost which start the ingest.
# [port] system RTMP stream port.
output rtmp://127.0.0.1:[port]/live?vhost=[vhost]/livestream;
}
}
}
# vhost for http static and flv vod stream for each vhost.
vhost http.static.srs.com {
# http static vhost specified config
http {
# whether enabled the http static service for vhost.
# default: off
enabled on;
# the url to mount to,
# typical mount to [vhost]/
# the variables:
# [vhost] current vhost for http server.
# @remark the [vhost] is optional, used to mount at specified vhost.
# @remark the http of __defaultVhost__ will override the http_stream section.
# for example:
# mount to [vhost]/
# access by http://ossrs.net:8080/xxx.html
# mount to [vhost]/hls
# access by http://ossrs.net:8080/hls/xxx.html
# mount to /
# access by http://ossrs.net:8080/xxx.html
# or by http://192.168.1.173:8080/xxx.html
# mount to /hls
# access by http://ossrs.net:8080/hls/xxx.html
# or by http://192.168.1.173:8080/hls/xxx.html
# @remark the port of http is specified by http_stream section.
# default: [vhost]/
mount [vhost]/hls;
# main dir of vhost,
# to delivery HTTP stream of this vhost.
# default: ./objs/nginx/html
dir ./objs/nginx/html/hls;
}
}
# vhost for http flv/aac/mp3 live stream for each vhost.
vhost http.flv.srs.com {
# http flv/mp3/aac stream vhost specified config
http_flv {
# whether enable the http flv live streaming service for vhost.
# default: off
enabled on;
# the fast cache for audio stream(mp3/aac),
# to cache more audio and send to client in a time to make android(weixin) happy.
# @remark the flv stream ignore it
# default: 30
fast_cache 30;
# the stream mout for rtmp to remux to flv live streaming.
# typical mount to [vhost]/[app]/[stream].flv
# the variables:
# [vhost] current vhost for http flv live stream.
# [app] current app for http flv live stream.
# [stream] current stream for http flv live stream.
# @remark the [vhost] is optional, used to mount at specified vhost.
# the extension:
# .flv mount http live flv stream, use default gop cache.
# .mp3 mount http live mp3 stream, ignore video and audio mp3 codec required.
# .aac mount http live aac stream, ignore video and audio aac codec required.
# for example:
# mount to [vhost]/[app]/[stream].flv
# access by http://ossrs.net:8080/live/livestream.flv
# mount to /[app]/[stream].flv
# access by http://ossrs.net:8080/live/livestream.flv
# or by http://192.168.1.173:8080/live/livestream.flv
# mount to [vhost]/[app]/[stream].mp3
# access by http://ossrs.net:8080/live/livestream.mp3
# mount to [vhost]/[app]/[stream].aac
# access by http://ossrs.net:8080/live/livestream.aac
# @remark the port of http is specified by http_stream section.
# default: [vhost]/[app]/[stream].flv
mount [vhost]/[app]/[stream].flv;
}
}
# the vhost with hls specified.
vhost with-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled on;
# the hls output path.
# the app dir is auto created under the hls_path.
# for example, for rtmp stream:
# rtmp://127.0.0.1/live/livestream
# http://127.0.0.1/live/livestream.m3u8
# where hls_path is /hls, srs will create the following files:
# /hls/live the app dir for all streams.
# /hls/live/livestream.m3u8 the HLS m3u8 file.
# /hls/live/livestream-1.ts the HLS media/ts file.
# in a word, the hls_path is for vhost.
# default: ./objs/nginx/html
hls_path ./objs/nginx/html;
# the hls fragment in seconds, the duration of a piece of ts.
# default: 10
hls_fragment 10;
# the hls window in seconds, the number of ts in m3u8.
# default: 60
hls_window 60;
# the error strategy. canbe:
# ignore, when error ignore and disable hls.
# disconnect, when error disconnect the publish connection.
# continue, when error ignore and continue output hls.
# @see https://github.com/winlinvip/simple-rtmp-server/issues/264
# default: ignore
hls_on_error ignore;
}
}
# the vhost with hls disabled.
vhost no-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# default: off
enabled off;
}
}
# the http hook callback vhost, srs will invoke the hooks for specified events.
vhost hooks.callback.srs.com {
http_hooks {
# whether the http hooks enalbe.
# default off.
enabled on;
# when client connect to vhost/app, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_connect",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "tcUrl": "rtmp://video.test.com/live?key=d2fa801d08e3f90ed1e1670e6e52651a",
# "pageUrl": "http://www.test.com/live.html"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_connect http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_connect http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client close/disconnect to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_close",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_close http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_close http://127.0.0.1:8085/api/v1/clients http://localhost:8085/api/v1/clients;
# when client(encoder) publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_publish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_publish http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_publish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client(encoder) stop publish to vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_unpublish",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_unpublish http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_unpublish http://127.0.0.1:8085/api/v1/streams http://localhost:8085/api/v1/streams;
# when client start to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_play",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_play http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_play http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
# when client stop to play vhost/app/stream, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_stop",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
# on_stop http://xxx/api0 http://xxx/api1 http://xxx/apiN
on_stop http://127.0.0.1:8085/api/v1/sessions http://localhost:8085/api/v1/sessions;
# when srs reap a dvr file, call the hook,
# the request in the POST data string is a object encode by json:
# {
# "action": "on_dvr",
# "client_id": 1985,
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "stream": "livestream",
# "cwd": "/usr/local/srs",
# "file": "./objs/nginx/html/live/livestream.1420254068776.flv"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
on_dvr http://127.0.0.1:8085/api/v1/dvrs http://localhost:8085/api/v1/dvrs;
}
}
# the vhost for srs debug info, whether send args in connect(tcUrl).
vhost debug.srs.com {
# when upnode(forward to, edge push to, edge pull from) is srs,
# it's strongly recommend to open the debug_srs_upnode,
# when connect to upnode, it will take the debug info,
# for example, the id, source id, pid.
# please see: https://github.com/winlinvip/simple-rtmp-server/wiki/v1_CN_SrsLog
# default: on
debug_srs_upnode on;
}
# the vhost for min delay, donot cache any stream.
vhost min.delay.com {
# @see vhost mrw.srs.com for detail.
min_latency on;
mr {
enabled off;
}
mw_latency 100;
# whether cache the last gop.
# if on, cache the last gop and dispatch to client,
# to enabled fast startup for client, client play immediately.
# if off, send the latest media data to client,
# client need to wait for the next Iframe to decode and show the video.
# set to off if requires min delay;
# set to on if requires client fast startup.
# default: on
gop_cache off;
# the max live queue length in seconds.
# if the messages in the queue exceed the max length,
# drop the old whole gop.
# default: 30
queue_length 10;
}
# the vhost for antisuck.
vhost refer.anti_suck.com {
# the common refer for play and publish.
# if the page url of client not in the refer, access denied.
# if not specified this field, allow all.
# default: not specified.
refer github.com github.io;
# refer for publish clients specified.
# the common refer is not overrided by this.
# if not specified this field, allow all.
# default: not specified.
refer_publish github.com github.io;
# refer for play clients specified.
# the common refer is not overrided by this.
# if not specified this field, allow all.
# default: not specified.
refer_play github.com github.io;
}
# the vhost which forward publish streams.
vhost same.vhost.forward.srs.com {
# forward all publish stream to the specified server.
# this used to split/forward the current stream for cluster active-standby,
# active-active for cdn to build high available fault tolerance system.
# format: {ip}:{port} {ip_N}:{port_N}
forward 127.0.0.1:1936 127.0.0.1:1937;
}
# the main comments for transcode
vhost example.transcode.srs.com {
# the streaming transcode configs.
transcode {
# whether the transcode enabled.
# if off, donot transcode.
# default: off.
enabled on;
# the ffmpeg
ffmpeg ./objs/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 example {
# 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;
# ffmpeg filters, follows the main input.
vfilter {
# the logo input file.
i ./doc/ffmpeg-logo.png;
# the ffmpeg complex filter.
# for filters, @see: http://ffmpeg.org/ffmpeg-filters.html
filter_complex 'overlay=10:10';
}
# 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
vbitrate 1500;
# video framerate.
vfps 25;
# video width, must be even numbers.
vwidth 768;
# video height, must be even numbers.
vheight 320;
# the max threads for ffmpeg to used.
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:
# libaacplus: use aac(libaacplus) audio encoder.
# copy: donot encoder the audio stream, copy it.
# an: disable audio output.
acodec libaacplus;
# audio bitrate, in kbps. [16, 72] for libaacplus.
abitrate 70;
# audio sample rate. for flv/rtmp, it must be:
# 44100,22050,11025,5512
asample_rate 44100;
# audio channel, 1 for mono, 2 for stereo.
achannels 2;
# other ffmpeg audio params
aparams {
# audio params, @see: http://ffmpeg.org/ffmpeg-codecs.html#Audio-Encoders
profile:a aac_low;
}
# 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];
}
}
}
# the mirror filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#Filtering-Introduction
vhost mirror.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine mirror {
enabled on;
vfilter {
vf 'split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the drawtext filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#drawtext-1
# remark: we remove the libfreetype which always cause build failed, you must add it manual if needed.
#######################################################################################################
# the crop filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#crop
vhost crop.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine crop {
enabled on;
vfilter {
vf 'crop=in_w-20:in_h-160:10:80';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# the logo filter of ffmpeg, @see: http://ffmpeg.org/ffmpeg-filters.html#overlay
vhost logo.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine logo {
enabled on;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# audio transcode only.
# for example, FMLE publish audio codec in mp3, and donot support HLS output,
# we can transcode the audio to aac and copy video to the new stream with HLS.
vhost audio.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine acodec {
enabled on;
vcodec copy;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# disable video, transcode/copy audio.
# for example, publish pure audio stream.
vhost vn.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine vn {
enabled on;
vcodec vn;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# ffmpeg-copy(forward implements by ffmpeg).
# copy the video and audio to a new stream.
vhost copy.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of vhost
# the comments, read example.transcode.srs.com
vhost all.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ffsuper {
enabled on;
iformat flv;
vfilter {
i ./doc/ffmpeg-logo.png;
filter_complex 'overlay=10:10';
}
vcodec libx264;
vbitrate 1500;
vfps 25;
vwidth 768;
vheight 320;
vthreads 12;
vprofile main;
vpreset medium;
vparams {
t 100;
coder 1;
b_strategy 2;
bf 3;
refs 10;
}
acodec libaacplus;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
profile:a aac_low;
}
oformat flv;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffhd {
enabled on;
vcodec libx264;
vbitrate 1200;
vfps 25;
vwidth 1382;
vheight 576;
vthreads 6;
vprofile main;
vpreset medium;
vparams {
}
acodec libaacplus;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine ffsd {
enabled on;
vcodec libx264;
vbitrate 800;
vfps 25;
vwidth 1152;
vheight 480;
vthreads 4;
vprofile main;
vpreset fast;
vparams {
}
acodec libaacplus;
abitrate 60;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine fffast {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine vcopy {
enabled on;
vcodec copy;
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine acopy {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
engine copy {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all stream using the empty ffmpeg demo, donothing.
vhost ffempty.transcode.srs.com {
transcode {
enabled on;
ffmpeg ./objs/research/ffempty;
engine empty {
enabled on;
vcodec libx264;
vbitrate 300;
vfps 20;
vwidth 768;
vheight 320;
vthreads 2;
vprofile baseline;
vpreset superfast;
vparams {
}
acodec libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}
# transcode all app and stream of app
vhost app.transcode.srs.com {
# the streaming transcode configs.
# if app specified, transcode all streams of app.
transcode live {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
# transcode specified stream.
vhost stream.transcode.srs.com {
# the streaming transcode configs.
# if stream specified, transcode the matched stream.
transcode live/livestream {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine {
enabled off;
}
}
}
# vhost for bandwidth check
# generally, the bandcheck vhost must be: bandcheck.srs.com,
# or need to modify the vhost of client.
vhost bandcheck.srs.com {
enabled on;
chunk_size 65000;
# bandwidth check config.
bandcheck {
# whether support bandwidth check,
# default: off.
enabled on;
# the key for server to valid,
# if invalid key, server disconnect and abort the bandwidth check.
key "35c9b402c12a7246868752e2878f7e0e";
# the interval in seconds for bandwidth check,
# server donot allow new test request.
# default: 30
interval 30;
# the max available check bandwidth in kbps.
# to avoid attack of bandwidth check.
# default: 1000
limit_kbps 4000;
}
}
# set the chunk size of vhost.
vhost chunksize.srs.com {
# the default chunk size is 128, max is 65536,
# some client does not support chunk size change,
# vhost chunk size will override the global value.
# default: global chunk size.
chunk_size 128;
}
# vhost for time jitter
vhost jitter.srs.com {
# 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;
}
# vhost for atc.
vhost atc.srs.com {
# vhost for atc for hls/hds/rtmp backup.
# generally, atc default to off, server delivery rtmp stream to client(flash) timestamp from 0.
# when atc is on, server delivery rtmp stream by absolute time.
# atc is used, for instance, encoder will copy stream to master and slave server,
# server use atc to delivery stream to edge/client, where stream time from master/slave server
# is always the same, client/tools can slice RTMP stream to HLS according to the same time,
# if the time not the same, the HLS stream cannot slice to support system backup.
#
# @see http://www.adobe.com/cn/devnet/adobe-media-server/articles/varnish-sample-for-failover.html
# @see http://www.baidu.com/#wd=hds%20hls%20atc
#
# default: off
atc on;
# whether enable the auto atc,
# if enabled, detect the bravo_atc="true" in onMetaData packet,
# set atc to on if matched.
# always ignore the onMetaData if atc_auto is off.
# default: on
atc_auto on;
}
# the vhost disabled.
vhost removed.srs.com {
# whether the vhost is enabled.
# if off, all request access denied.
# default: on
enabled off;
}
# config for the pithy print,
# which always print constant message specified by interval,
# whatever the clients in concurrency.
pithy_print {
# shared print interval for all publish clients, in milliseconds.
# default: 10000
publish 10000;
# shared print interval for all play clients, in milliseconds.
# default: 10000
play 10000;
# shared print interval for all forwarders, in milliseconds.
# default: 10000
forwarder 10000;
# shared print interval for all encoders, in milliseconds.
# default: 10000
encoder 10000;
# shared print interval for all ingesters, in milliseconds.
# default: 10000
ingester 10000;
# shared print interval for all hls, in milliseconds.
# default: 10000
hls 10000;
# shared print interval for all edge, in milliseconds.
# default: 10000
edge 10000;
}