v2_EN_FFMPEG
SRS can transcode RTMP streams and output to any RTMP server, typically itself.
The important use scenario of FFMPEG:
- One in N out: Publish a high resolution video with big bitrate, for intance, h.264 5Mbps 1080p. Then use FFMPEG to transcode to multiple bitrates, for example, 1080p/720p/576p, the 576p is for mobile devices.
- Support multiple screen: The stream published by flash is in h264/vp6/mp3/speex codec. Use FFMPEG to transcode to HLS(h264+aac) for IOS/Android.
- Stream filters: For example, add logo to stream. SRS supports all filters from FFMPEG.
The workflow of SRS transcoding:
- Encoder publishes RTMP to SRS.
- SRS forks a process for FFMPEG when transcoding is configured.
- The forked FFMPEG transcodes the stream and publishes it to SRS or other servers.
The SRS transcoding feature can apply on vhost, app or a specified stream.
listen 1935;
vhost __defaultVhost__ {
# 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
# @remark 0 to use source video bitrate.
# default: 0
vbitrate 1500;
# video framerate.
# @remark 0 to use source video fps.
# default: 0
vfps 25;
# video width, must be even numbers.
# @remark 0 to use source video width.
# default: 0
vwidth 768;
# video height, must be even numbers.
# @remark 0 to use source video height.
# default: 0
vheight 320;
# 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 libfdk_aac;
# 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];
}
}
}The configuration applies to all streams of this vhost, for example:
- Publish stream to: rtmp://dev:1935/live/livestream
- Play the origin stream: rtmp://dev:1935/live/livestream
- Play the transcoded stream: rtmp://dev:1935/live/livestream_ff
The output URL contains some variables:
- [vhost] The input stream vhost, for instance, dev.ossrs.net
- [port] The input stream port, for instance, 1935
- [app] The input stream app, for instance, live
- [stream] The input stream name, for instance, livestream
- [engine] The transcode engine name, which follows the keyword engine, for instance, ff
Add the app or app/stream when you need to apply transcoding to it:
listen 1935;
vhost __defaultVhost__ {
# Transcode all streams of app "live"
transcode live {
}
}Or for streams:
listen 1935;
vhost __defaultVhost__ {
# Transcode stream name is "livestream" and app is "live"
transcode live/livestream{
}
}All params of SRS transcode are for FFMPEG, and SRS renames some parameters:
| SRS | FFMPEG | Exammple | Description |
|---|---|---|---|
| vcodec | vcodec | ffmpeg ... -vcodec libx264 ... | The codec to use. |
| vbitrate | b:v | ffmpeg ... -b:v 500000 ... | The bitrate in kbps (for SRS) or bps (for FFMPEG) at which to output the transcoded stream. |
| vfps | r | ffmpeg ... -r 25 ... | The output framerate. |
| vwidth/vheight | s | ffmpeg ... -s 400x300 -aspect 400:300 ... | The output video size, the width x height and the aspect set to width:height. |
| vthreads | threads | ffmpeg ... -threads 8 ... | The number of encoding threads for x264. |
| vprofile | profile:v | ffmpeg ... -profile:v high ... | The profile for x264. |
| vpreset | preset | ffmpeg ... -preset medium ... | The preset for x264. |
| acodec | acodec | ffmpeg ... -acodec libfdk_aac ... | The codec for audio. |
| abitrate | b:a | ffmpeg ... -b:a 70000 ... | The bitrate in kbps (for SRS) and bps (for FFMPEG) for output audio. For libaacplus：16-72k. No limit for libfdk_aac. |
| asample_rate | ar | ffmpeg ... -ar 44100 ... | The audio sample rate. |
| achannels | ac | ffmpeg ... -ac 2 ... | The audio channel. |
There are more parameters for SRS:
- vfilter：Parameters added before the vcodec, for the FFMPEG filters.
- vparams：Parameters added after the vcodec, for the video transcode parameters.
- aparams：Parameters added after the acodec and before the -y, for the audio transcode parameters.
These parameters will generated by the sequence:
ffmpeg -f flv -i <input_rtmp> {vfilter} -vcodec ... {vparams} -acodec ... {aparams} -f flv -y {output}The actual parameters used to fork FFMPEG can be found in the log by the keywords start transcoder:
[2014-02-28 21:38:09.603][4][trace][start] start transcoder,
log: ./objs/logs/encoder-__defaultVhost__-live-livestream.log,
params: ./objs/ffmpeg/bin/ffmpeg -f flv -i
rtmp://127.0.0.1:1935/live?vhost=__defaultVhost__/livestream
-vcodec libx264 -b:v 500000 -r 25.00 -s 768x320 -aspect 768:320
-threads 12 -profile:v main -preset medium -acodec libfdk_aac
-b:a 70000 -ar 44100 -ac 2 -f flv
-y rtmp://127.0.0.1:1935/live?vhost=__defaultVhost__/livestream_ff When an FFMPEG process is forked, SRS will redirect the stdout and stderr to the log file, for instance, ./objs/logs/encoder-__defaultVhost__-live-livestream.log. Sometimes the log file is very large, so users can add parameters to vfilter to tell FFMPEG to generate less verbose logs:
listen 1935;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vfilter {
# -v quiet
v quiet;
}
vcodec libx264;
vbitrate 500;
vfps 25;
vwidth 768;
vheight 320;
vthreads 12;
vprofile main;
vpreset medium;
vparams {
}
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}That is, add the parameter -v quiet to FFMPEG.
Set the vcodec/acodec to copy, FFMPEG will demux and mux without transcoding, like the forward of SRS. Users can copy video and transcode audio, for example, when flash is publishing the stream with h264+speex.
listen 1935;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vcodec copy;
acodec libfdk_aac;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}Or, copy video and audio:
listen 1935;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vcodec copy;
acodec copy;
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}FFMPEG can drop video or audio streams by configuring vcodec to vn and acodec to an. For example:
listen 1935;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine vn {
enabled on;
vcodec vn;
acodec libfdk_aac;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}The configuration above will output pure audio in the aac codec.
There are lots of vhost in conf/full.conf for transcoding, or refer to FFMPEG:
- mirror.transcode.srs.com
- drawtext.transcode.srs.com
- crop.transcode.srs.com
- logo.transcode.srs.com
- audio.transcode.srs.com
- copy.transcode.srs.com
- all.transcode.srs.com
- ffempty.transcode.srs.com
- app.transcode.srs.com
- stream.transcode.srs.com
- vn.transcode.srs.com
SRS can fork FFMEPG on ARM, see: Raspberry pi Transcode
Note: To use your own tool, you can disable ffmpeg by --without-ffmpeg, but you must open a transcoder by using --with-transcode, see: Build
Flash web pages can encode and publish RTMP streams to the server, and the audio codec must be speex, nellymoser or pcma/pcmu.
Flash will disable audio when no audio is published, so FFMPEG may cannot discover the audio in the stream and will disable the audio.
FFMPEG links:
Winlin 2015.6