v1_EN_FFMPEG
SRS can transcode the RTMP stream and output to any RTMP server, typically itself.
The important use scenaio 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 support all filters of FFMPEG.
The workflow of SRS transcoding:
- Encoder publish RTMP to SRS.
- SRS fork process for FFMPEG when transcode configed.
- The forked FFMPEG transcode stream and publish to SRS or other servers.
The SRS transcoding feature can apply on vhost, app or specified stream.
listen 1935;
vhost __defaultVhost__ {
# apply on vhost, for all streams.
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
# whether the engine is enabled
# default: off.
enabled on;
# ffmpeg filters, follows the main input.
vfilter {
}
# 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 {
}
# 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 {
}
# 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 config apply 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 intput stream name, for instance, livestream
- [engine] The transcode engine name, which follow the keyword engine, for instance, ff
Add the app or app/stream when need to apply transcode to app or stream:
listen 1935;
vhost __defaultVhost__ {
# Transcode all streams of app "live"
transcode live {
}
}Or for stream:
listen 1935;
vhost __defaultVhost__ {
# Transcode stream name "livestream" and app is "live"
transcode live/livestream{
}
}All params of SRS transcode is for FFMPEG, and SRS rename some parameters:
| SRS | FFMPEG | Exammple | Description |
|---|---|---|---|
| vcodec | vcodec | ffmpeg ... -vcodec libx264 ... | The codec to use. |
| 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 encode thread 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 libaacplus ... | 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 |
| asample_rate | ar | ffmpeg ... -ar 44100 ... | The audio sample rate. |
| achannels | ac | ffmpeg ... -ac 2 ... | THe audio channel. |
There are more parameter of 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 generate by the sequence:
ffmpeg -f flv -i <input_rtmp> {vfilter} -vcodec ... {vparams} -acodec ... {aparams} -f flv -y {output}The actual parameters to fork FFMPEG can find the log by keyword 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 libaacplus
-b:a 70000 -ar 44100 -ac 2 -f flv
-y rtmp://127.0.0.1:1935/live?vhost=__defaultVhost__/livestream_ff When FFMPEG process 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. User can add parameter to vfilter to tell FFMPEG to generate less log:
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 libaacplus;
abitrate 70;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}That is, add parameter -v quiet to FFMPEG.
Set the vcodec/acodec to copy, FFMPEG will demux and mux without transcode, like the forward of SRS. User can copy video and transcode audio, for example, when the flash publish stream with h264+speex.
listen 1935;
vhost __defaultVhost__ {
transcode {
enabled on;
ffmpeg ./objs/ffmpeg/bin/ffmpeg;
engine ff {
enabled on;
vcodec copy;
acodec libaacplus;
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 stream by config the 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 libaacplus;
abitrate 45;
asample_rate 44100;
achannels 2;
aparams {
}
output rtmp://127.0.0.1:[port]/[app]?vhost=[vhost]/[stream]_[engine];
}
}
}The config above will output pure audio in aac codec.
There are lots of vhost in conf/full.conf for transcode, or refer to the 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 the FFMEPG on ARM, see: Raspberry pi Transcode
Note: To use your own tool, you can disable the ffmpeg by --without-ffmpeg, but must open the transcode --with-transcode, see: Build
The flash web page can encode and publish RTMP stream to server, and the audio codec must be speex, nellymoser or pcma/pcmu.
The flash will disable audio when no audio when publish, so FFMPEG may cannot discovery the audio in stream and disable the audio.
FFMPEG links:
Winlin 2014.10