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merge from @mbeacom

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winlin 2016-04-22 10:18:13 +08:00
parent 2e75c9f45f
commit 892591f9a6
1 changed files with 90 additions and 95 deletions
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trunk/conf/full.conf
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@ -7,23 +7,20 @@
# for example, 192.168.1.100:1935 10.10.10.100:1935
# where the ip is optional, default to 0.0.0.0, that is 1935 equals to 0.0.0.0:1935
listen 1935;
# change to this dir as the cwd.
# ignore if empty or not configed.
work_dir /usr/local/srs;
# the pid file
# to ensure only one process can use a pid file
# and provides the current running process id, for script,
# 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
# however, most clients support 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.
# if enabled ffmpeg, each transcoding stream will create a log file.
# /dev/null to disable the log.
# default: ./objs
ff_log_dir ./objs;
@ -43,8 +40,8 @@ srs_log_file ./objs/srs.log;
# if exceed the max connections, server will drop the new connection.
# default: 1000
max_connections 1000;
# whether start as deamon
# @remark: donot support reload.
# whether start as daemon
# @remark: do not support reload.
# default: on
daemon on;
# whether use utc_time to generate the time struct,
@ -65,7 +62,7 @@ pithy_print_ms 10000;
# @remark, the ip report to server, is retrieve from system stat,
# which need the config item stats.network.
heartbeat {
# whether heartbeat is enalbed.
# whether heartbeat is enabled.
# default: off
enabled off;
# the interval seconds for heartbeat,
@ -80,7 +77,7 @@ heartbeat {
# }
# default: http://127.0.0.1:8085/api/v1/servers
url http://127.0.0.1:8085/api/v1/servers;
# the id of devide.
# the id of device.
device_id "my-srs-device";
# whether report with summaries
# if on, put /api/v1/summaries to the request data:
@ -104,7 +101,7 @@ stats {
# default: 0
network 0;
# the device name to stat the disk iops.
# ignore the device of /proc/diskstats if not configed.
# ignore the device of /proc/diskstats if not configured.
disk sda sdb xvda xvdb;
}
@ -145,15 +142,15 @@ http_api {
allow_update off;
}
}
# embeded http server in srs.
# embedded 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.
# for example, srs running 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__
# @remark, the http embedded stream need to config the vhost, for instance, the __defaultVhost__
# need to open the feature http of vhost.
http_server {
# whether http streaming service is enabled.
@ -267,20 +264,20 @@ vhost scope.vhost.srs.com {
# if off, all request access denied.
# default: on
enabled off;
# whether enable min delay mode for vhost.
# for min latence mode:
# for min latency mode:
# 1. disable the publish.mr for vhost.
# 2. use timeout for cond wait for consumer queue.
# @see https://github.com/ossrs/srs/issues/257
# default: off
min_latency off;
# whether enable the TCP_NODELAY
# if on, set the nodelay of fd by setsockopt
# default: off
tcp_nodelay off;
# 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.
@ -314,7 +311,7 @@ vhost cluster.srs.com {
# @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.
@ -322,16 +319,16 @@ vhost cluster.srs.com {
# but if user prefer origin check(auth), the token_traverse if better solution.
# default: off
token_traverse off;
# the vhost to transform for edge,
# to fetch from the specified vhost at origin,
# if not specified, use the current vhost of edge in origin, the variable [vhost].
# default: [vhost]
vhost same.edge.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,
# when connect to upnode, it will take the debug info,
# for example, the id, source id, pid.
# please see: https://github.com/ossrs/srs/wiki/v1_CN_SrsLog
# default: on
@ -397,20 +394,20 @@ vhost play.srs.com {
# default: on
gop_cache off;
# the max live queue length in seconds.
# if the messages in the queue exceed the max length,
# if the messages in the queue exceed the max length,
# drop the old whole gop.
# default: 30
queue_length 10;
# about the stream monotonically increasing:
# 1. video timestamp is monotonically increasing,
# 1. video timestamp is monotonically increasing,
# 2. audio timestamp is monotonically increasing,
# 3. video and audio timestamp is interleaved/mixed 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.
# 2. zero, only ensure stream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# default: full
time_jitter full;
@ -427,7 +424,7 @@ vhost play.srs.com {
# 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
#
@ -439,14 +436,14 @@ vhost play.srs.com {
# always ignore the onMetaData if atc_auto is off.
# default: off
atc_auto off;
# set the MW(merged-write) latency in ms.
# 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;
# the minimal packets send interval in ms,
# used to control the ndiff of stream by srs_rtmp_dump,
# for example, some device can only accept some stream which
@ -487,33 +484,33 @@ vhost atc.srs.com {
vhost mrw.srs.com {
# @see scope.vhost.srs.com
min_latency off;
# @see play.srs.com
play {
mw_latency 350;
}
# @see publish.srs.com
publish {
mr on;
mr_latenct 350;
mr_latency 350;
}
}
# the vhost for min delay, donot cache any stream.
# the vhost for min delay, do not cache any stream.
vhost min.delay.com {
# @see scope.vhost.srs.com
min_latency on;
# @see scope.vhost.srs.com
tcp_nodelay on;
# @see play.srs.com
play {
mw_latency 100;
gop_cache off;
queue_length 10;
}
# @see publish.srs.com
publish {
mr off;
@ -534,7 +531,7 @@ vhost stream.control.com {
min_latency on;
# @see scope.vhost.srs.com
tcp_nodelay on;
# @see play.srs.com
play {
mw_latency 100;
@ -542,7 +539,7 @@ vhost stream.control.com {
send_min_interval 10.0;
reduce_sequence_header on;
}
# @see publish.srs.com
publish {
mr off;
@ -564,12 +561,12 @@ vhost publish.srs.com {
# 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:
# when there are 2500 publisher, the total memory of SRS at least:
# 183KB * 2500 = 446MB
# the value recomment is [300, 2000]
# the recommended value is [300, 2000]
# default: 350
mr_latency 350;
# the 1st packet timeout in ms for encoder.
# default: 20000
firstpkt_timeout 20000;
@ -584,7 +581,7 @@ vhost publish.srs.com {
}
}
# the vhost for antisuck.
# the vhost for anti-suck.
vhost refer.anti_suck.com {
# refer hotlink-denial.
refer {
@ -597,12 +594,12 @@ vhost refer.anti_suck.com {
# default: not specified.
all github.com github.io;
# refer for publish clients specified.
# the common refer is not overrided by this.
# the common refer is not overridden by this.
# if not specified this field, allow all.
# default: not specified.
publish github.com github.io;
# refer for play clients specified.
# the common refer is not overrided by this.
# the common refer is not overridden by this.
# if not specified this field, allow all.
# default: not specified.
play github.com github.io;
@ -624,7 +621,7 @@ vhost bandcheck.srs.com {
# 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.
# server do not allow new test request.
# default: 30
interval 30;
# the max available check bandwidth in kbps.
@ -637,7 +634,7 @@ vhost bandcheck.srs.com {
# the security to allow or deny clients.
vhost security.srs.com {
# security for host to allow or deny clients.
# @see https://github.com/ossrs/srs/issues/211
# @see https://github.com/ossrs/srs/issues/211
security {
# whether enable the security for vhost.
# default: off
@ -670,7 +667,7 @@ vhost http.static.srs.com {
# whether enabled the http static service for vhost.
# default: off
enabled on;
# the url to mount to,
# the url to mount to,
# typical mount to [vhost]/
# the variables:
# [vhost] current vhost for http server.
@ -710,7 +707,7 @@ vhost http.remux.srs.com {
# @remark 0 to disable fast cache for http audio stream.
# default: 0
fast_cache 30;
# the stream mout for rtmp to remux to live streaming.
# the stream mount for rtmp to remux to live streaming.
# typical mount to [vhost]/[app]/[stream].flv
# the variables:
# [vhost] current vhost for http live stream.
@ -748,7 +745,7 @@ vhost http.remux.srs.com {
# 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.
# whether the http hooks enable.
# default off.
enabled on;
# when client connect to vhost/app, call the hook,
@ -760,7 +757,7 @@ vhost hooks.callback.srs.com {
# "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
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -774,7 +771,7 @@ vhost hooks.callback.srs.com {
# "ip": "192.168.1.10", "vhost": "video.test.com", "app": "live",
# "send_bytes": 10240, "recv_bytes": 10240
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -788,7 +785,7 @@ vhost hooks.callback.srs.com {
# "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
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -802,7 +799,7 @@ vhost hooks.callback.srs.com {
# "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
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -817,7 +814,7 @@ vhost hooks.callback.srs.com {
# "stream": "livestream",
# "pageUrl": "http://www.test.com/live.html"
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -831,7 +828,7 @@ vhost hooks.callback.srs.com {
# "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
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
# support multiple api hooks, format:
@ -847,7 +844,7 @@ vhost hooks.callback.srs.com {
# "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
# if valid, the hook must return HTTP code 200(Status 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;
@ -866,7 +863,7 @@ vhost hooks.callback.srs.com {
# "m3u8_url": "live/livestream/live.m3u8",
# "seq_no": 100
# }
# if valid, the hook must return HTTP code 200(Stauts OK) and response
# if valid, the hook must return HTTP code 200(Status OK) and response
# an int value specifies the error code(0 corresponding to success):
# 0
on_hls http://127.0.0.1:8085/api/v1/hls http://localhost:8085/api/v1/hls;
@ -891,10 +888,10 @@ vhost exec.srs.com {
enabled off;
# when publish stream, exec the process with variables:
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [port] the input stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
# [engine] the transcode engine name.
# other variables for exec only:
# [url] the rtmp url which trigger the publish.
# [tcUrl] the client request tcUrl.
@ -909,7 +906,7 @@ vhost exec.srs.com {
vhost with-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# if off, do not write hls(ts and m3u8) when publish.
# default: off
enabled on;
# the hls fragment in seconds, the duration of a piece of ts.
@ -923,14 +920,14 @@ vhost with-hls.srs.com {
hls_td_ratio 1.5;
# the audio overflow ratio.
# for pure audio, the duration to reap the segment.
# for example, the hls_fragment is 10s, hsl_aof_ratio is 2.0,
# the segemnt will reap to 20s for pure audio.
# for example, the hls_fragment is 10s, hls_aof_ratio is 2.0,
# the segment will reap to 20s for pure audio.
# default: 2.0
hls_aof_ratio 2.0;
# the hls window in seconds, the number of ts in m3u8.
# default: 60
hls_window 60;
# the error strategy. canbe:
# the error strategy. can be:
# ignore, disable the hls.
# disconnect, require encoder republish.
# continue, ignore failed try to continue output hls.
@ -939,14 +936,14 @@ vhost with-hls.srs.com {
hls_on_error continue;
# the hls storage: disk, ram or both.
# disk, to write hls m3u8/ts to disk.
# ram, serve m3u8/ts in memory, which use embeded http server to delivery.
# ram, serve m3u8/ts in memory, which use embedded http server to delivery.
# both, disk and ram.
# default: disk
hls_storage disk;
# the hls output path.
# the m3u8 file is configed by hls_path/hls_m3u8_file, the default is:
# the m3u8 file is configured by hls_path/hls_m3u8_file, the default is:
# ./objs/nginx/html/[app]/[stream].m3u8
# the ts file is configed by hls_path/hls_ts_file, the default is:
# the ts file is configured by hls_path/hls_ts_file, the default is:
# ./objs/nginx/html/[app]/[stream]-[seq].ts
# @remark the hls_path is compatible with srs v1 config.
# default: ./objs/nginx/html
@ -956,7 +953,6 @@ vhost with-hls.srs.com {
# [vhost], the vhost of stream.
# [app], the app of stream.
# [stream], the stream name of stream.
# recommend: [vhost]/[app]/[stream].m3u8
# default: [app]/[stream].m3u8
hls_m3u8_file [app]/[stream].m3u8;
# the hls ts file name.
@ -968,24 +964,23 @@ vhost with-hls.srs.com {
# [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.
# [04], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace this const to current millisecond.
# [timestamp],replace this const to current UNIX timestamp in ms.
# [seq], the sequence number of ts.
# @see https://github.com/ossrs/srs/wiki/v2_CN_DVR#custom-path
# @see https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#hls-config
# recommend: [vhost]/[app]/[stream]-[seq].ts
# default: [app]/[stream]-[seq].ts
hls_ts_file [app]/[stream]-[seq].ts;
# whether use floor for the hls_ts_file path generation.
# if on, use floor(timestamp/hls_fragment) as the variable [timestamp],
# and use enahanced algorithm to calc deviation for segment.
# and use enhanced algorithm to calc deviation for segment.
# @remark when floor on, recommend the hls_segment>=2*gop.
# default: off
hls_ts_floor off;
# the hls entry prefix, which is base url of ts url.
# for exmaple, the prefix is:
# for example, the prefix is:
# http://your-server/
# then, the ts path in m3u8 will be like:
# http://your-server/live/livestream-0.ts
@ -994,7 +989,7 @@ vhost with-hls.srs.com {
# optional, default to empty string.
hls_entry_prefix http://your-server;
# the hls mount for hls_storage ram,
# which use srs embeded http server to delivery HLS,
# which use srs embedded http server to delivery HLS,
# where the mount specifies the HTTP url to mount.
# @see the mount of http_remux.
# @remark the hls_mount must endswith .m3u8.
@ -1003,7 +998,7 @@ vhost with-hls.srs.com {
# the default audio codec of hls.
# when codec changed, write the PAT/PMT table, but maybe ok util next ts.
# so user can set the default codec for mp3.
# the available audio codec:
# the available audio codec:
# aac, mp3, an
# default: aac
hls_acodec aac;
@ -1039,7 +1034,7 @@ vhost with-hls.srs.com {
# for the hls http callback, @see http_hooks.on_hls of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DeliveryHLS#http-callback
# @read https://github.com/ossrs/srs/wiki/v2_EN_DeliveryHLS#http-callback
# on_hls_notify, never config in here, should config in http_hooks.
# we support the variables to generate the notify url:
# [app], replace with the app.
@ -1054,7 +1049,7 @@ vhost with-hls.srs.com {
vhost no-hls.srs.com {
hls {
# whether the hls is enabled.
# if off, donot write hls(ts and m3u8) when publish.
# if off, do not write hls(ts and m3u8) when publish.
# default: off
enabled off;
}
@ -1087,7 +1082,7 @@ vhost dvr.srs.com {
# whether enabled dvr features
# default: off
enabled on;
# the filter for dvr to aplly to.
# the filter for dvr to apply to.
# all, dvr all streams of all apps.
# <app>/<stream>, apply to specified stream of app.
# for example, to dvr the following two streams:
@ -1109,9 +1104,9 @@ vhost dvr.srs.com {
# [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.
# [04], replace this const to current minute.
# [05], replace this const to current second.
# [999], replace 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" as "[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
@ -1136,7 +1131,7 @@ vhost dvr.srs.com {
# segment,session apply it.
# default: ./objs/nginx/html/[app]/[stream].[timestamp].flv
dvr_path ./objs/nginx/html/[app]/[stream].[timestamp].flv;
# the duration for dvr file, reap if exeed, in seconds.
# the duration for dvr file, reap if exceed, in seconds.
# segment apply it.
# session,append ignore.
# default: 30
@ -1149,19 +1144,19 @@ vhost dvr.srs.com {
# default: on
dvr_wait_keyframe on;
# about the stream monotonically increasing:
# 1. video timestamp is 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.
# 2. zero, only ensure stream start at zero, ignore timestamp jitter.
# 3. off, disable the time jitter algorithm, like atc.
# apply for all dvr plan.
# default: full
time_jitter full;
# on_dvr, never config in here, should config in http_hooks.
# for the dvr http callback, @see http_hooks.on_dvr of vhost hooks.callback.srs.com
# @read https://github.com/ossrs/srs/wiki/v2_CN_DVR#http-callback
@ -1184,15 +1179,15 @@ vhost ingest.srs.com {
input {
# the type of input.
# can be file/stream/device, that is,
# file: ingest file specifies by url.
# stream: ingest stream specifeis by url.
# file: ingest file specified by url.
# stream: ingest stream specified by url.
# device: not support yet.
# default: file
type file;
# the url of file/stream.
url ./doc/source.200kbps.768x320.flv;
}
# the ffmpeg
# the ffmpeg
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
# the transcode engine, @see all.transcode.srs.com
# @remark, the output is specified following.
@ -1209,7 +1204,7 @@ vhost ingest.srs.com {
}
}
# the vhost for intest with transcode engine.
# the vhost for ingest with transcode engine.
vhost transcode.ingest.srs.com {
ingest livestream {
enabled on;
@ -1262,7 +1257,7 @@ vhost example.transcode.srs.com {
# if off, donot transcode.
# default: off.
enabled on;
# the ffmpeg
# the ffmpeg
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
# the transcode engine for matched stream.
# all matched stream will transcoded to the following stream.
@ -1317,7 +1312,7 @@ vhost example.transcode.srs.com {
# high,main,baseline
vprofile main;
# x264 preset, "ffmpeg -preset"
# @see x264 -help, can be:
# @see x264 -help, can be:
# ultrafast,superfast,veryfast,faster,fast
# medium,slow,slower,veryslow,placebo
vpreset medium;
@ -1369,10 +1364,10 @@ vhost example.transcode.srs.com {
oformat flv;
# output stream, "ffmpeg -y", variables:
# [vhost] the input stream vhost.
# [port] the intput stream port.
# [port] the input stream port.
# [app] the input stream app.
# [stream] the input stream name.
# [engine] the tanscode engine name.
# [engine] the transcode engine name.
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
@ -1472,7 +1467,7 @@ vhost logo.transcode.srs.com {
}
}
# audio transcode only.
# for example, FMLE publish audio codec in mp3, and donot support HLS output,
# for example, FMLE publish audio codec in mp3, and do not 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 {
@ -1656,7 +1651,7 @@ vhost all.transcode.srs.com {
}
}
}
# transcode all stream using the empty ffmpeg demo, donothing.
# transcode all stream using the empty ffmpeg demo, do nothing.
vhost ffempty.transcode.srs.com {
transcode {
enabled on;