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v1_EN_FFMPEG

winlinvip edited this page Nov 2, 2014 · 22 revisions

Live Streaming Transcode

SRS can transcode the RTMP stream and output to any RTMP server, typically itself.

Use Scenario

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.

Workflow

The workflow of SRS transcoding:

  1. Encoder publish RTMP to SRS.
  2. SRS fork process for FFMPEG when transcode configed.
  3. The forked FFMPEG transcode stream and publish to SRS or other servers.

Transcode Config

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{
    }
}

Transcode Rulers

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.
vbitrate b:v ffmpeg ... -b:v 500000 ... The bitrate in kbps(for SRS) or bps(for FFMPEG) to output transcode 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 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 

FFMPEG Log Path

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.

Copy Without Transcode

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];
        }
    }
}

Drop Video or Audio

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.

其他转码配置

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。

ARM下转码

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

FFMPEG转码flash流

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

FFMPEG相关链接:

Winlin 2014.10

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