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• ffmpeg and ffserver, rc buffer underflow ?

  25 février 2018, par Dove Devic

  I am attempting to write a simple streaming server for a project. I have an AWS Linux machine that will be running ffserver. Curently, as it stands, my config file looks like the following :

  #Server Configs
  
  HTTPPort 8090
  
  HTTPBindAddress 0.0.0.0
  
  MaxHTTPConnections 2000
  
  MaxClients 1000
  
  MaxBandwidth 1000
  
  CustomLog -
  
  
  
  #Create a Status Page
  
  <stream>
  
  Format status
  
  ACL allow localhost
  
  ACL allow 255.255.255.255 #Allow everyone to view status, for now
  
  </stream>
  
  
  
  #Creates feed, only allow from self
  
  <feed>
  
  File /tmp/feed1.ffm
  
  FileMaxSize 50M
  
  ACL allow 127.0.0.1
  
  ACL allow 
  
  </feed>
  
  
  
  #Creates stream, allow everyone
  
  <stream>
  
  Format mpeg
  
  Feed feed1.ffm
  
  VideoFrameRate 30
  
  VideoSize 640x480
  
  AudioSampleRate 44100
  
  </stream>

  I then am capturing my Webcam and sending it up to the server using the following command :

  ffmpeg -f dshow 
  
         -i video="Webcam C170":audio="Microphone (Webcam C170)" 
  
         -b:v 1400k  
  
         -maxrate 2400k  
  
         -bufsize 1200k  
  
         -ab 64k  
  
         -s 640x480  
  
         -ac 1  
  
         -ar 44100  
  
         -y http://:8090/feed1.ffm

  When I run this however, I get the following output from my console :

  Guessed Channel Layout for  Input Stream #0.1 : stereo
  
  Input #0, dshow, from 'video=Webcam C170:audio=Microphone (Webcam C170)':
  
    Duration: N/A, start: 12547.408000, bitrate: N/A
  
      Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 640x480, 30 tbr, 10000k tbn, 30 tbc
  
      Stream #0:1: Audio: pcm_s16le, 44100 Hz, 2 channels, s16, 1411 kb/s
  
  Output #0, ffm, to ':8090/feed1.ffm':
  
    Metadata:
  
      creation_time   : 2017-04-26 14:55:27
  
      encoder         : Lavf57.25.100
  
      Stream #0:0: Audio: mp2, 44100 Hz, mono, s16, 64 kb/s
  
      Metadata:
  
        encoder         : Lavc57.24.102 mp2
  
      Stream #0:1: Video: mpeg1video, yuv420p, 640x480, q=2-31, 64 kb/s, 30 fps, 1000k tbn, 30 tbc
  
      Metadata:
  
        encoder         : Lavc57.24.102 mpeg1video
  
      Side data:
  
        unknown side data type 10 (24 bytes)
  
  Stream mapping:
  
    Stream #0:1 -> #0:0 (pcm_s16le (native) -> mp2 (native))
  
    Stream #0:0 -> #0:1 (rawvideo (native) -> mpeg1video (native))
  
  Press [q] to stop, [?] for help
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:01.13 bitrate= 404.8kbits/s dup=13 drop=0 speed=2.22x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:01.63 bitrate= 361.1kbits/s dup=13 drop=0 speed=1.61x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:02.13 bitrate= 368.6kbits/s dup=13 drop=0 speed= 1.4x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:02.66 bitrate= 344.1kbits/s dup=13 drop=0 speed=1.32x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflowtime=00:00:03.16 bitrate= 331.1kbits/s dup=13 drop=0 speed=1.25x
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  [mpeg1video @ 02e95180] rc buffer underflow
  
  [mpeg1video @ 02e95180] max bitrate possibly too small or try trellis with large lmax or increase qmax
  
  frame=  117 fps= 36 q=31.0 Lsize=     156kB time=00:00:03.86 bitrate= 330.5kbits/s dup=13 drop=0 speed= 1.2x
  
  video:118kB audio:27kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 7.659440%
  
  Exiting normally, received signal 2.

  And on my viewer, I just get a black screen.

  Is there something I’m missing ? Searching lead to nothing on "increasing qmax" or anything similar to what ffmpeg complained about. There have been questions asked here, but nothing has been done/answered.

  Thanks in advance

• Duration of wav file saved in S3 using AWS Lambda

  3 juin 2021, par Salim Shamim

  Objective

  


  To calculate the duration of a wav file which is saved in S3 by AWS Lambda using node.js. I had to add ffmpeg and ffprobe executable inside a lambda layer (Downloaded linux-64 version from here). These files could be found in /opt folder on lambda file system.

  


  What I have tried

  


  I have been trying using ffprobe in numerous ways, but I get Invalid Data as error.
Here's one example

  


  const AWS = require('aws-sdk');
const s3 = new AWS.S3();
const fs = require('fs');
const ffmpeg = require('fluent-ffmpeg');

exports.handler = async function(event) {
    let path = await load();
    console.log(`Saved Path ${path}`);

    ffmpeg.setFfmpegPath('/opt/ffmpeg');
    ffmpeg.setFfprobePath("/opt/ffprobe");

    let dur = await duration(path).catch(err => {
        console.log(err);
    })
    console.log(dur);
}


function duration(path) {
    return new Promise((resolve, reject) => {
        ffmpeg(path).ffprobe(path, function(err, metadata) {
            //console.dir(metadata); // all metadata
            if (err) {
                reject(err);
            }
            else {
                resolve(metadata.format.duration);

            }
        });
    })
}

async function listFiles(path) {
    console.log('list files');
    return new Promise((resolve, reject) => {
        fs.readdir(path, (err, files) => {
            if (err) {
                console.error('Error in readdir');
                reject(err);
            }
            else {
                console.log('recieved files');
                resolve(files);
            }

        });

    });

}

async function load() {
    return new Promise((resolve, reject) => {
        let params = {
            Key: 'Fanfare60.wav',
            Bucket: 'samplevideosshamim'
        };
        console.log(`Getting s3 object : ${JSON.stringify(params)}`);
        s3.getObject(params, (err, data) => {
            if (err) {
                console.error(err);
                reject(err);
            }
            else if (data) {
                console.log('Recieved Data');
                let path = `/tmp/${params.Key}`;
                console.log('Path: ' + path);
                fs.writeFileSync(path, data.body);
                resolve(path);
            }
        });
    });

}


  


  Error :

  


  Error: ffprobe exited with code 1&#xA;ffprobe version 4.2.1-static https://johnvansickle.com/ffmpeg/  Copyright (c) 2007-2019 the FFmpeg developers&#xA;  built with gcc 6.3.0 (Debian 6.3.0-18&#x2B;deb9u1) 20170516&#xA;  configuration: --enable-gpl --enable-version3 --enable-static --disable-debug --disable-ffplay --disable-indev=sndio --disable-outdev=sndio --cc=gcc-6 --enable-fontconfig --enable-frei0r --enable-gnutls --enable-gmp --enable-libgme --enable-gray --enable-libaom --enable-libfribidi --enable-libass --enable-libvmaf --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librubberband --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libvorbis --enable-libopus --enable-libtheora --enable-libvidstab --enable-libvo-amrwbenc --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libdav1d --enable-libxvid --enable-libzvbi --enable-libzimg&#xA;  libavutil      56. 31.100 / 56. 31.100&#xA;  libavcodec     58. 54.100 / 58. 54.100&#xA;  libavformat    58. 29.100 / 58. 29.100&#xA;  libavdevice    58.  8.100 / 58.  8.100&#xA;  libavfilter     7. 57.100 /  7. 57.100&#xA;  libswscale      5.  5.100 /  5.  5.100&#xA;  libswresample   3.  5.100 /  3.  5.100&#xA;  libpostproc    55.  5.100 / 55.  5.100&#xA;/tmp/Fanfare60.wav: Invalid data found when processing input&#xA;&#xA;    at ChildProcess.<anonymous> (/var/task/node_modules/fluent-ffmpeg/lib/ffprobe.js:233:22)&#xA;    at ChildProcess.emit (events.js:314:20)&#xA;    at ChildProcess.EventEmitter.emit (domain.js:483:12)&#xA;    at Process.ChildProcess._handle.onexit (internal/child_process.js:276:12)&#xA;</anonymous>

  


  I am guessing it doesn't support wav format, but internet searches provide no proof of that.

  


  A point to note here is, I was able to get the duration of a local file when I ran this code on my local machine, but I have a windows machine, so perhaps only linux executable of ffprobe has issue ?

  


  Possible Solutions I am looking for

  


  

  1. Is there a way to specify format ?

  


  2. Can I use a different library (code example for the same) ?

  


  3. Any possible way to get duration of a wav file in the mentioned scenario (AWS Lambda NodeJS and S3 file (private file) ?

  


  


• How to Simply Remove Duplicate Frames from a Video using ffmpeg

  29 janvier 2017, par Skeeve

  First of all, I’d preface this by saying I’m NO EXPERT with video manipulation,
  although I’ve been fiddling with ffmpeg for years (in a fairly limited way). Hence, I’m not too flash with all the language folk often use... and how it affects what I’m trying to do in my manipulations... but I’ll have a go with this anyway...

  I’ve checked a few links here, for example :
  ffmpeg - remove sequentially duplicate frames

  ...but the content didn’t really help me.

  I have some hundreds of video clips that have been created under both Windows and Linux using both ffmpeg and other similar applications. However, they have some problems with times in the video where the display is ’motionless’.

  As an example, let’s say we have some web site that streams a live video into, say, a Flash video player/plugin in a web browser. In this case, we’re talking about a traffic camera video stream, for example.

  There’s an instance of ffmpeg running that is capturing a region of the (Windows) desktop into a video file, viz :-

  ffmpeg -hide_banner -y -f dshow ^
  
        -i video="screen-capture-recorder" ^
  
        -vf "setpts=1.00*PTS,crop=448:336:620:360" ^
  
        -an -r 25 -vcodec libx264 -crf 0 -qp 0 ^
  
        -preset ultrafast SAMPLE.flv

  Let’s say the actual ’display’ that is being captured looks like this :-

  123456789 XXXXX 1234567 XXXXXXXXXXX 123456789 XXXXXXX
  
  ^---a---^ ^-P-^ ^--b--^ ^----Q----^ ^---c---^ ^--R--^ 

  ...where each character position represents a (sequence of) frame(s). Owing to a poor internet connection, a "single frame" can be displayed for an extended period (the ’X’ characters being an (almost) exact copy of the immediately previous frame). So this means we have segments of the captured video where the image doesn’t change at all (to the naked eye, anyway).

  How can we deal with the duplicate frames ?... and how does our approach change if the ’duplicates’ are NOT the same to ffmpeg but LOOK more-or-less the same to the viewer ?

  If we simply remove the duplicate frames, the ’pacing’ of the video is lost, and what used to take, maybe, 5 seconds to display, now takes a fraction of a second, giving a very jerky, unnatural motion, although there are no duplicate images in the video. This seems to be achievable using ffmpeg with an ’mp_decimate’ option, viz :-

       ffmpeg -i SAMPLE.flv ^                      ... (i)
  
          -r 25 ^
  
          -vf mpdecimate,setpts=N/FRAME_RATE/TB DEC_SAMPLE.mp4

  That reference I quoted uses a command that shows which frames ’mp_decimate’ will remove when it considers them to be ’the same’, viz :-

       ffmpeg -i SAMPLE.flv ^                      ... (ii)
  
          -vf mpdecimate ^
  
          -loglevel debug -f null -

  ...but knowing that (complicated formatted) information, how can we re-organize the video without executing multiple runs of ffmpeg to extract ’slices’ of video for re-combining later ?

  In that case, I’m guessing we’d have to run something like :-

  • user specifies a ’threshold duration’ for the duplicates
    (maybe run for 1 sec only)
  • determine & save main video information (fps, etc - assuming
    constant frame rate)
  • map the (frame/time where duplicates start)->no. of
    frames/duration of duplicates
  • if the duration of duplicates is less than the user threshold,
    don’t consider this period as a ’series of duplicate frames’
    and move on
  • extract the ’non-duplicate’ video segments (a, b & c in the
    diagram above)
  • create ’new video’ (empty) with original video’s specs
  • for each video segment
    extract the last frame of the segment
    create a short video clip with repeated frames of the frame
    just extracted (duration = user spec. = 1 sec)
    append (current video segment+short clip) to ’new video’
    and repeat

  ...but in my case, a lot of the captured videos might be 30 minutes long and have hundreds of 10 sec long pauses, so the ’rebuilding’ of the videos will take a long time using this method.

  This is why I’m hoping there’s some "reliable" and "more intelligent" way to use
  ffmepg (with/without the ’mp_decimate’ filter) to do the ’decimate’ function in only a couple of passes or so... Maybe there’s a way that the required segments could even be specified (in a text file, for example) and as ffmpeg runs it will
  stop/restart it’s transcoding at specified times/frame numbers ?

  Short of this, is there another application (for use on Windows or Linux) that could do what I’m looking for, without having to manually set start/stop points,
  extracting/combining video segments manually...?

  I’ve been trying to do all this with ffmpeg N-79824-gcaee88d under Win7-SP1 and (a different version I don’t currently remember) under Puppy Linux Slacko 5.6.4.

  Thanks a heap for any clues.