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GetID3 - Bloc informations de fichiers
9 avril 2013, par
Mis à jour : Mai 2013
Langue : français
Type : Image
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Sur d’autres sites (7894)
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python subprocess ffmpeg return code = 69
13 juin 2023, par Tim ChenI 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, promptingreturncode=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.wavafter fastapi handle request on the command line andsubprocess.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 VelagaI 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 'react'
import Navbar from '../../components/Navbar/Navbar'
import './Dashboard.css'

const CAPTURE_OPTIONS = {
 audio: true,
 video: true,
}

function Dashboard() {
 const [mute, setMute] = useState(false)
 const videoRef = useRef()
 const ws = useRef()
 const mediaStream = useUserMedia(CAPTURE_OPTIONS)

 let liveStream
 let liveStreamRecorder

 if (mediaStream && videoRef.current && !videoRef.current.srcObject) {
 videoRef.current.srcObject = mediaStream
 }

 const handleCanPlay = () => {
 videoRef.current.play()
 }

 useEffect(() => {
 ws.current = new WebSocket(
 window.location.protocol.replace('http', 'ws') +
 '//' + // http: -> ws:, https: -> wss:
 'localhost:3000'
 )

 ws.current.onopen = () => {
 console.log('WebSocket Open')
 }

 return () => {
 ws.current.close()
 }
 }, [])

 const startStream = () => {
 liveStream = videoRef.current.captureStream(30) // 30 FPS
 liveStreamRecorder = new MediaRecorder(liveStream, {
 mimeType: 'video/webm;codecs=h264',
 videoBitsPerSecond: 3 * 1024 * 1024,
 })
 liveStreamRecorder.ondataavailable = (e) => {
 ws.current.send(e.data)
 console.log('send data', e.data)
 }
 // Start recording, and dump data every second
 liveStreamRecorder.start(1000)
 }

 const stopStream = () => {
 liveStreamRecorder.stop()
 ws.current.close()
 }

 const toggleMute = () => {
 setMute(!mute)
 }

 return (
 <>
 <navbar></navbar>
 <div style="{{" classname="'main'">
 <div>
 
 </div>
 <div classname="'button-container'">
 <button>Go Live</button>
 <button>Stop Recording</button>
 <button>Share Screen</button>
 <button>Mute</button>
 </div>
 </div>
 >
 )
}

const useUserMedia = (requestedMedia) => {
 const [mediaStream, setMediaStream] = useState(null)

 useEffect(() => {
 async function enableStream() {
 try {
 const stream = await navigator.mediaDevices.getUserMedia(requestedMedia)
 setMediaStream(stream)
 } catch (err) {
 console.log(err)
 }
 }

 if (!mediaStream) {
 enableStream()
 } else {
 return function cleanup() {
 mediaStream.getVideoTracks().forEach((track) => {
 track.stop()
 })
 }
 }
 }, [mediaStream, requestedMedia])

 return mediaStream
}

export default Dashboard


-
How to improve web camera streaming latency to v4l2loopback device with ffmpeg ?
11 mars, par Made by MosesI'm trying to stream my iPhone camera to my PC on LAN.


What I've done :


- 

-
HTTP server with html page and streaming script :


I use WebSockets here and maybe WebRTC is better choice but it seems like network latency is good enough






async function beginCameraStream() {
 const mediaStream = await navigator.mediaDevices.getUserMedia({
 video: { facingMode: "user" },
 });

 websocket = new WebSocket(SERVER_URL);

 websocket.onopen = () => {
 console.log("WS connected");

 const options = { mimeType: "video/mp4", videoBitsPerSecond: 1_000_000 };
 mediaRecorder = new MediaRecorder(mediaStream, options);

 mediaRecorder.ondataavailable = async (event) => {
 // to measure latency I prepend timestamp to the actual video bytes chunk
 const timestamp = Date.now();
 const timestampBuffer = new ArrayBuffer(8);
 const dataView = new DataView(timestampBuffer);
 dataView.setBigUint64(0, BigInt(timestamp), true);
 const data = await event.data.bytes();

 const result = new Uint8Array(data.byteLength + 8);
 result.set(new Uint8Array(timestampBuffer), 0);
 result.set(data, 8);

 websocket.send(result);
 };

 mediaRecorder.start(100); // Collect 100ms chunks
 };
}


- 

-
Server to process video chunks






import { serve } from "bun";
import { Readable } from "stream";

const V4L2LOOPBACK_DEVICE = "/dev/video10";

export const setupFFmpeg = (v4l2device) => {
 // prettier-ignore
 return spawn("ffmpeg", [
 '-i', 'pipe:0', // Read from stdin
 '-pix_fmt', 'yuv420p', // Pixel format
 '-r', '30', // Target 30 fps
 '-f', 'v4l2', // Output format
 v4l2device, // Output to v4l2loopback device
 ]);
};

export class FfmpegStream extends Readable {
 _read() {
 // This is called when the stream wants more data
 // We push data when we get chunks
 }
}

function main() {
 const ffmpeg = setupFFmpeg(V4L2LOOPBACK_DEVICE);
 serve({
 port: 8000,
 fetch(req, server) {
 if (server.upgrade(req)) {
 return; // Upgraded to WebSocket
 }
 },
 websocket: {
 open(ws) {
 console.log("Client connected");
 const stream = new FfmpegStream();
 stream.pipe(ffmpeg?.stdin);

 ws.data = {
 stream,
 received: 0,
 };
 },
 async message(ws, message) {
 const view = new DataView(message.buffer, 0, 8);
 const ts = Number(view.getBigUint64(0, true));
 ws.data.received += message.byteLength;
 const chunk = new Uint8Array(message.buffer, 8, message.byteLength - 8);

 ws.data.stream.push(chunk);

 console.log(
 [
 `latency: ${Date.now() - ts} ms`,
 `chunk: ${message.byteLength}`,
 `total: ${ws.data.received}`,
 ].join(" | "),
 );
 },
 },
 });
}

main();


After I try to open the v4l2loopback device


cvlc v4l2:///dev/video10


picture is delayed for at least 1.5 sec which is unacceptable for my project.


Thoughts :


- 

- Problem doesn't seems to be with network latency




latency: 140 ms | chunk: 661 Bytes | total: 661 Bytes
latency: 206 ms | chunk: 16.76 KB | total: 17.41 KB
latency: 141 ms | chunk: 11.28 KB | total: 28.68 KB
latency: 141 ms | chunk: 13.05 KB | total: 41.74 KB
latency: 199 ms | chunk: 11.39 KB | total: 53.13 KB
latency: 141 ms | chunk: 16.94 KB | total: 70.07 KB
latency: 139 ms | chunk: 12.67 KB | total: 82.74 KB
latency: 142 ms | chunk: 13.14 KB | total: 95.88 KB


150ms is actually too much for 15KB on LAN but there can some issue with my router


- 

- As far as I can tell it neither ties to ffmpeg throughput :




Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'pipe:0':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 creation_time : 2025-03-09T17:16:49.000000Z
 Duration: 00:00:01.38, start:
0.000000, bitrate: N/A
 Stream #0:0(und): Video: h264 (Baseline) (avc1 / 0x31637661), yuvj420p(pc), 1280x720, 4012 kb/s, 57.14 fps, 29.83 tbr, 600 tbn, 1200 tbc (default)
 Metadata:
 rotate : 90
 creation_time : 2025-03-09T17:16:49.000000Z
 handler_name : Core Media Video
 Side data:
 displaymatrix: rotation of -90.00 degrees

Stream mapping:
 Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))

[swscaler @ 0x55d8d0b83100] deprecated pixel format used, make sure you did set range correctly

Output #0, video4linux2,v4l2, to '/dev/video10':
 Metadata:
 major_brand : iso5
 minor_version : 1
 compatible_brands: isomiso5hlsf
 encoder : Lavf58.45.100

Stream #0:0(und): Video: rawvideo (I420 / 0x30323449), yuv420p, 720x1280, q=2-31, 663552 kb/s, 60 fps, 60 tbn, 60 tbc (default)
 Metadata:
 encoder : Lavc58.91.100 rawvideo
 creation_time : 2025-03-09T17:16:49.000000Z
 handler_name : Core Media Video
 Side data:
 displaymatrix: rotation of -0.00 degrees

frame= 99 fps=0.0 q=-0.0 size=N/A time=00:00:01.65 bitrate=N/A dup=50 drop=0 speed=2.77x
frame= 137 fps=114 q=-0.0 size=N/A time=00:00:02.28 bitrate=N/A dup=69 drop=0 speed=1.89x
frame= 173 fps= 98 q=-0.0 size=N/A time=00:00:02.88 bitrate=N/A dup=87 drop=0 speed=1.63x
frame= 210 fps= 86 q=-0.0 size=N/A time=00:00:03.50 bitrate=N/A dup=105 drop=0 speed=1.44x
frame= 249 fps= 81 q=-0.0 size=N/A time=00:00:04.15 bitrate=N/A dup=125 drop=0 speed=1.36
frame= 279 fps= 78 q=-0.0 size=N/A time=00:00:04.65 bitrate=N/A dup=139 drop=0 speed=1.31x


- 

-
I also tried to write the video stream directly to
video.mp4file and immediately open it withvlcbut it only can be successfully opened after 1.5 sec.

-
I've tried to use OBS v4l2 input source instead of vlc but the latency is the same








Update №1


When i try to stream actual
.mp4file toffmpegit works almost immediately with 0.2sec delay to spin up the ffmpeg itself :

cat video.mp4 | ffmpeg -re -i pipe:0 -pix_fmt yuv420p -f v4l2 /dev/video10 & ; sleep 0.2 && cvlc v4l2:///dev/video10


So the problem is apparently with streaming process


-
Syndiquer tout le site
