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v1_EN_FFMPEG
winlinvip edited this page Nov 2, 2014
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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{
}
}SRS的转码参数全是FFMPEG的参数，有些参数SRS做了自定义，见下表。
| SRS参数 | FFMPEG参数 | 实例 | 说明 |
|---|---|---|---|
| vcodec | vcodec | ffmpeg ... -vcodec libx264 ... | 指定视频编码器 |
| vbitrate | b:v | ffmpeg ... -b:v 500000 ... | 输出的视频码率 |
| vfps | r | ffmpeg ... -r 25 ... | 输出的视频帧率 |
| vwidth/vheight | s | ffmpeg ... -s 400x300 -aspect 400:300 ... | 输出的视频宽度x高度，以及宽高比 |
| vthreads | threads | ffmpeg ... -threads 8 ... | 编码线程数 |
| vprofile | profile:v | ffmpeg ... -profile:v high ... | 编码x264的profile |
| vpreset | preset | ffmpeg ... -preset medium ... | 编码x264的preset |
| acodec | acodec | ffmpeg ... -acodec libaacplus ... | 音频编码器 |
| abitrate | b:a | ffmpeg ... -b:a 70000 ... | 音频输出码率。libaacplus：16-72k |
| asample_rate | ar | ffmpeg ... -ar 44100 ... | 音频采样率 |
| achannels | ac | ffmpeg ... -ac 2 ... | 音频声道 |
另外，还有三个是可以加其他ffmpeg参数：
- vfilter：添加在vcodec之前的滤镜参数。
- vparams：添加在vcodec之后，acodec之前的视频编码参数。
- aparams：添加在acodec之后，-y之前的音频编码参数。
这些参数应用的顺序是：
ffmpeg -f flv -i <input_rtmp> {vfilter} -vcodec ... {vparams} -acodec ... {aparams} -f flv -y {output}具体参数可以查看SRS的日志，譬如：
[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 FFMPEG启动后，SRS会将stdout和stderr都定向到日志文件，譬如./objs/logs/encoder-__defaultVhost__-live-livestream.log，有时候日志会比较大。可以配置ffmpeg输出较少日志：
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];
}
}
}对ffmpeg添加-v quiet参数即可。
可以配置vcodec/acodec copy，实现不转码。譬如，视频为h264编码，但是音频是mp3/speex，需要转码音频为aac，然后切片为HLS输出。
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];
}
}
}或者拷贝视频和音频：
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];
}
}
}可以禁用视频或者音频，只输出音频或视频。譬如，电台可以丢弃视频，对音频转码为aac后输出HLS。
可以配置vcodec为vn，acodec为an实现禁用。例如：
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];
}
}
}该配置只输出纯音频，编码为aac。
conf/full.conf中有很多FFMPEG转码配置的实例，也可以参考ffmpeg的命令行。
- mirror.transcode.srs.com 将视频流上半截，翻转到下半截，看起来像个镜子。
- drawtext.transcode.srs.com 加文字水印。
- crop.transcode.srs.com 剪裁视频。
- logo.transcode.srs.com 添加图片logo。
- audio.transcode.srs.com 只对音频转码。
- copy.transcode.srs.com 不转码只转封装，类似于SRS的Forward。
- all.transcode.srs.com 转码参数的详细说明。
- ffempty.transcode.srs.com 一个ffmpeg的mock，不转码只打印参数。
- app.transcode.srs.com 对指定的app的流转码。
- stream.transcode.srs.com 对指定的流转码。
- vn.transcode.srs.com 只输出音频，禁止视频输出。
SRS可以在ARM下调用系统的ffmpeg转码，参考：Raspberry pi 转码
注意：使用自己的工具时，需要禁用ffmpeg，但是打开transcode选项：--with-transcode --without-ffmpeg，这样就不会编译ffmpeg，但是编译了直播转码功能。参考：Build
flash可以当作编码器推流，参考演示中的编码器或者视频会议。flash只支持speex/nellymoser/pcma/pcmu，但flash会有一个特性，没有声音时就没有音频包。FFMPEG会依赖于这些音频包，如果没有会认为没有音频。
所以FFMPEG用来转码flash推上来的RTMP流时，可能会有一个问题：ffmpeg认为没有音频。
另外，FFMPEG取flash的流的时间会很长，也可能是在等待这些音频包。
FFMPEG相关链接：
Winlin 2014.10
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