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• FFmpeg "Non-monotonous DTS in output stream" error when processing video from Safari's MediaRecorder

  17 juillet 2024, par Hackermon

  I'm recording a video stream in Safari with MediaRecorder, then sending it to a remote server which then uses ffmpeg to reencode the video. When reencoding with FFmpeg, I get a lot of warnings and the final video is broken, frame are glitching and out of sync but the audio sounds fine.

  


  Here's my MediaRecorder script :

  


  const camera = await navigator.mediaDevices.getUserMedia({ audio: true, video: true });
const recorder = new MediaRecorder(camera, {
        mimeType: 'video/mp4', // Safari only supports MP4
        bitsPerSecond: 1_000_000,
});

recorder.ondataavailable = async ({ data: blob }) => {
        // open contents in new tab
        var fileURL = URL.createObjectURL(file);
         window.open(fileURL, '_blank');
};

recorder.start();
setTimeout(() => recorder.stop(), 5000);


  


  I download the video blob from Safari and use this command to reencode it :

  


  ffmpeg -i ./blob.mp4 -preset ultrafast -strict -2 -threads 10 -c copy ./output.mp4


  


  Logs :

  


  ffmpeg version 4.2.7-0ubuntu0.1 Copyright (c) 2000-2022 the FFmpeg developers
  built with gcc 9 (Ubuntu 9.4.0-1ubuntu1~20.04.1)
  configuration: --prefix=/usr --extra-version=0ubuntu0.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-nvenc --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 31.100 / 56. 31.100
  libavcodec     58. 54.100 / 58. 54.100
  libavformat    58. 29.100 / 58. 29.100
  libavdevice    58.  8.100 / 58.  8.100
  libavfilter     7. 57.100 /  7. 57.100
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  5.100 /  5.  5.100
  libswresample   3.  5.100 /  3.  5.100
  libpostproc    55.  5.100 / 55.  5.100
[mov,mp4,m4a,3gp,3g2,mj2 @ 0x559826616000] DTS 29 < 313 out of order
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from './chunk1.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2024-07-17T14:30:47.000000Z
  Duration: 00:00:01.00, start: 0.000000, bitrate: 3937 kb/s
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], 6218 kb/s, 33.36 fps, 600 tbr, 600 tbn, 1200 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
File './safari3.mp4' already exists. Overwrite ? [y/N] Output #0, mp4, to './safari3.mp4':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    encoder         : Lavf58.29.100
    Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p(progressive), 640x480 [SAR 1:1 DAR 4:3], q=2-31, 6218 kb/s, 33.36 fps, 600 tbr, 19200 tbn, 600 tbc (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Video
    Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 420 kb/s (default)
    Metadata:
      creation_time   : 2024-07-17T14:30:47.000000Z
      handler_name    : Core Media Audio
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10016, current: 928; changing to 10017. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10017, current: 1568; changing to 10018. This may result in incorrect timestamps in the output file.
[mp4 @ 0x559826644340] Non-monotonous DTS in output stream 0:0; previous: 10018, current: 2208; changing to 10019. This may result in incorrect timestamps in the output file.
...x100
frame=  126 fps=0.0 q=-1.0 Lsize=     479kB time=00:00:00.97 bitrate=4026.2kbits/s speed= 130x    
video:425kB audio:51kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.652696%


  


  Not sure what's happening or how to fix it. This issue only happens in Safari, videos from Chrome are perfectly fine.

  


  I've tried various flags :

  


  -fflags +igndts
-bsf:a aac_adtstoasc
-c:v libvpx-vp9 -c:a libopus
etc


  


  None of them seem to fix the issue.

  


• python subprocess ffmpeg return code = 69

  13 juin 2023, par Tim Chen

  I try to call ffmpeg through the subprocess.run(['ffmpeg', '-i', file_name, output_file_name], capture_output=True, text=True) command in python to convert the audio file incoming from the front end to wav format file. The backend code is as follows, using python+fastapi :

  


  @app.post("/api/upload/convert")
async def convert_upload_file(request: Request, file: UploadFile = File(...)):
    token = uuid.uuid4().hex
    tmpFileName = os.path.join(os.path.dirname(__file__), token)
    with open(tmpFileName, "wb") as buffer:
        buffer.write(await file.read())
    await file.seek(0)
    output_path = tmpFileName + '-output.wav'
    command = ['ffmpeg', '-i', tmpFileName, output_path]
    result = subprocess.run(command, capture_output=True, text=True)


  


  This code usually works, but there are some scenarios where it doesn't work. The audio file is recorded by js code (specifically navigator.mediaDevices.getUserMedia({audio: true})).
The code of the audio recorded in windows chrome can run normally and get the converted wav file, but the audio recorded from ios15 safari for more than 3 seconds cannot be converted, prompting returncode=69. The error message is as follows :

  


  CompletedProcess(args=['ffmpeg', '-i', '5cfb52c503a646bda0f422b517c8014a', '5cfb52c503a646bda0f422b517c8014a-output.wav'], returncode=69, stdout='', stderr="
ffmpeg version 4.4.2-0ubuntu0.22.04.1 Copyright (c) 2000-2021 the FFmpeg developers
built with gcc 11 (Ubuntu 11.2.0-19ubuntu1)
configuration: --prefix=/usr --extra-version=0ubuntu0.22.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librabbitmq --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libsrt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opencl --enable-opengl --enable-sdl2 --enable-pocketsphinx --enable-librsvg --enable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
libavutil      56. 70.100 / 56. 70.100
libavcodec     58.134.100 / 58.134.100
libavformat    58. 76.100 / 58. 76.100
libavdevice    58. 13.100 / 58. 13.100
libavfilter     7.110.100 /  7.110.100
libswscale      5.  9.100 /  5.  9.100
libswresample   3.  9.100 /  3.  9.100
libpostproc    55.  9.100 / 55.  9.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '5cfb52c503a646bda0f422b517c8014a':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    creation_time   : 2023-06-11T16:36:53.000000Z
  Duration: 00:00:07.06, start: 0.000000, bitrate: 187 kb/s
  Stream #0:0(und): Audio: aac (LC) (mp4a / 0x6134706D), 48000 Hz, mono, fltp, 184 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
Stream mapping:
  Stream #0:0 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, wav, to '5cfb52c503a646bda0f422b517c8014a-output.wav':
  Metadata:
    major_brand     : iso5
    minor_version   : 1
    compatible_brands: isomiso5hlsf
    ISFT            : Lavf58.76.100
  Stream #0:0(und): Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s16, 768 kb/s (default)
    Metadata:
      creation_time   : 2023-06-11T16:36:53.000000Z
      handler_name    : Core Media Audio
      vendor_id       : [0][0][0][0]
      encoder         : Lavc58.134.100 pcm_s16le
size=       2kB time=00:00:00.00 bitrate=N/A speed=N/A    
[aac @ 0x55f1f8f19fc0] Sample rate index in program config element does not match the sample rate index configured by the container.
[aac @ 0x55f1f8f19fc0] Too large remapped id is not implemented. Update your FFmpeg version to the newest one from Git. If the problem still occurs, it means that your file has a feature which has not been implemented.
[aac @ 0x55f1f8f19fc0] If you want to help, upload a sample of this file to https://streams.videolan.org/upload/ and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)
Error while decoding stream #0:0: Not yet implemented in FFmpeg, patches welcome
[aac @ 0x55f1f8f19fc0] Multiple frames in a packet.
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Number of bands (18) exceeds limit (13).
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.
[aac @ 0x55f1f8f19fc0] Prediction is not allowed in AAC-LC.
Error while decoding stream #0:0: Invalid data found when processing input
[aac @ 0x55f1f8f19fc0] Reserved bit set.


  


  For the abnormal code, I tried to execute ffmpeg -i input output.wav after fastapi handle request on the command line and subprocess.run(['ffmpeg', '-i', file_name, output_path], capture_output =True, text=True), all succeeded, which means that the final file must be normal, otherwise the subsequent verification work will get the same error.

  


  This confuses me, is there some information I'm missing ?

  


• how to add audio using ffmpeg when recording video from browser and streaming to Youtube/Twitch ?

  26 juillet 2021, par Tosh Velaga

  I have a web application I am working on that allows the user to stream video from their browser and simultaneously livestream to both Youtube and Twitch using ffmpeg. The application works fine when I don't need to send any of the audio. Currently I am getting the error below when I try to record video and audio. I am new to using ffmpeg and so any help would be greatly appreciated. Here is also my repo if needed : https://github.com/toshvelaga/livestream

  


  Here is my node.js server with ffmpeg

  


  const child_process = require('child_process') // To be used later for running FFmpeg
const express = require('express')
const http = require('http')
const WebSocketServer = require('ws').Server
const NodeMediaServer = require('node-media-server')
const app = express()
const cors = require('cors')
const path = require('path')
const logger = require('morgan')
require('dotenv').config()

app.use(logger('dev'))
app.use(cors())

app.use(express.json({ limit: '200mb', extended: true }))
app.use(
  express.urlencoded({ limit: '200mb', extended: true, parameterLimit: 50000 })
)

var authRouter = require('./routes/auth')
var compareCodeRouter = require('./routes/compareCode')

app.use('/', authRouter)
app.use('/', compareCodeRouter)

if (process.env.NODE_ENV === 'production') {
  // serve static content
  // npm run build
  app.use(express.static(path.join(__dirname, 'client/build')))

  app.get('*', (req, res) => {
    res.sendFile(path.join(__dirname, 'client/build', 'index.html'))
  })
}

const PORT = process.env.PORT || 8080

app.listen(PORT, () => {
  console.log(`Server is starting on port ${PORT}`)
})

const server = http.createServer(app).listen(3000, () => {
  console.log('Listening on PORT 3000...')
})


const wss = new WebSocketServer({
  server: server,
})

wss.on('connection', (ws, req) => {
  const ffmpeg = child_process.spawn('ffmpeg', [
    // works fine when I use this but when I need audio problems arise
    // '-f',
    // 'lavfi',
    // '-i',
    // 'anullsrc',

    '-i',
    '-',

    '-f',
    'flv',
    '-c',
    'copy',
    `${process.env.TWITCH_STREAM_ADDRESS}`,
    '-f',
    'flv',
    '-c',
    'copy',
    `${process.env.YOUTUBE_STREAM_ADDRESS}`,
    // '-f',
    // 'flv',
    // '-c',
    // 'copy',
    // `${process.env.FACEBOOK_STREAM_ADDRESS}`,
  ])

  ffmpeg.on('close', (code, signal) => {
    console.log(
      'FFmpeg child process closed, code ' + code + ', signal ' + signal
    )
    ws.terminate()
  })

  ffmpeg.stdin.on('error', (e) => {
    console.log('FFmpeg STDIN Error', e)
  })

  ffmpeg.stderr.on('data', (data) => {
    console.log('FFmpeg STDERR:', data.toString())
  })

  ws.on('message', (msg) => {
    console.log('DATA', msg)
    ffmpeg.stdin.write(msg)
  })

  ws.on('close', (e) => {
    console.log('kill: SIGINT')
    ffmpeg.kill('SIGINT')
  })
})

const config = {
  rtmp: {
    port: 1935,
    chunk_size: 60000,
    gop_cache: true,
    ping: 30,
    ping_timeout: 60,
  },
  http: {
    port: 8000,
    allow_origin: '*',
  },
}

var nms = new NodeMediaServer(config)
nms.run()


  


  Here is my frontend code that records the video/audio and sends to server :

  


  import React, { useState, useEffect, useRef } from &#x27;react&#x27;&#xA;import Navbar from &#x27;../../components/Navbar/Navbar&#x27;&#xA;import &#x27;./Dashboard.css&#x27;&#xA;&#xA;const CAPTURE_OPTIONS = {&#xA;  audio: true,&#xA;  video: true,&#xA;}&#xA;&#xA;function Dashboard() {&#xA;  const [mute, setMute] = useState(false)&#xA;  const videoRef = useRef()&#xA;  const ws = useRef()&#xA;  const mediaStream = useUserMedia(CAPTURE_OPTIONS)&#xA;&#xA;  let liveStream&#xA;  let liveStreamRecorder&#xA;&#xA;  if (mediaStream &amp;&amp; videoRef.current &amp;&amp; !videoRef.current.srcObject) {&#xA;    videoRef.current.srcObject = mediaStream&#xA;  }&#xA;&#xA;  const handleCanPlay = () => {&#xA;    videoRef.current.play()&#xA;  }&#xA;&#xA;  useEffect(() => {&#xA;    ws.current = new WebSocket(&#xA;      window.location.protocol.replace(&#x27;http&#x27;, &#x27;ws&#x27;) &#x2B;&#xA;        &#x27;//&#x27; &#x2B; // http: -> ws:, https: -> wss:&#xA;        &#x27;localhost:3000&#x27;&#xA;    )&#xA;&#xA;    ws.current.onopen = () => {&#xA;      console.log(&#x27;WebSocket Open&#x27;)&#xA;    }&#xA;&#xA;    return () => {&#xA;      ws.current.close()&#xA;    }&#xA;  }, [])&#xA;&#xA;  const startStream = () => {&#xA;    liveStream = videoRef.current.captureStream(30) // 30 FPS&#xA;    liveStreamRecorder = new MediaRecorder(liveStream, {&#xA;      mimeType: &#x27;video/webm;codecs=h264&#x27;,&#xA;      videoBitsPerSecond: 3 * 1024 * 1024,&#xA;    })&#xA;    liveStreamRecorder.ondataavailable = (e) => {&#xA;      ws.current.send(e.data)&#xA;      console.log(&#x27;send data&#x27;, e.data)&#xA;    }&#xA;    // Start recording, and dump data every second&#xA;    liveStreamRecorder.start(1000)&#xA;  }&#xA;&#xA;  const stopStream = () => {&#xA;    liveStreamRecorder.stop()&#xA;    ws.current.close()&#xA;  }&#xA;&#xA;  const toggleMute = () => {&#xA;    setMute(!mute)&#xA;  }&#xA;&#xA;  return (&#xA;    &lt;>&#xA;      <navbar></navbar>&#xA;      <div style="{{" classname="&#x27;main&#x27;">&#xA;        <div>&#xA;          &#xA;        </div>&#xA;        <div classname="&#x27;button-container&#x27;">&#xA;          <button>Go Live</button>&#xA;          <button>Stop Recording</button>&#xA;          <button>Share Screen</button>&#xA;          <button>Mute</button>&#xA;        </div>&#xA;      </div>&#xA;    >&#xA;  )&#xA;}&#xA;&#xA;const useUserMedia = (requestedMedia) => {&#xA;  const [mediaStream, setMediaStream] = useState(null)&#xA;&#xA;  useEffect(() => {&#xA;    async function enableStream() {&#xA;      try {&#xA;        const stream = await navigator.mediaDevices.getUserMedia(requestedMedia)&#xA;        setMediaStream(stream)&#xA;      } catch (err) {&#xA;        console.log(err)&#xA;      }&#xA;    }&#xA;&#xA;    if (!mediaStream) {&#xA;      enableStream()&#xA;    } else {&#xA;      return function cleanup() {&#xA;        mediaStream.getVideoTracks().forEach((track) => {&#xA;          track.stop()&#xA;        })&#xA;      }&#xA;    }&#xA;  }, [mediaStream, requestedMedia])&#xA;&#xA;  return mediaStream&#xA;}&#xA;&#xA;export default Dashboard&#xA;