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Error Opening RTMP Stream through FFmpeg command when executed through exec package [closed]
3 octobre 2024, par AkhilI have been trying to transcode the live stream from RTMP server running on
rtmp://localhost:1936/live/test
with FFmpeg in a Go application usingos/exec
package, But seems to not work and gives the input/output error (I have attached below). The same exact ffmpeg command when I execute on terminal, works as its supposed to. Not Sure why that is, here is my code for reproducing and analyzing the mistakes.

ffmpegCmd := fmt.Sprintf("ffmpeg -nostdin -i rtmp://localhost:1936/live/%s -c:v libx264 -s %s -f %s %s/stream.mpd",
 streamKey, resolution, sp.OutputFormat, outputPath)
 log.Printf("Executing FFmpeg command: %s", ffmpegCmd)

 // Prepare the command execution with a timeout context
 ctx, cancel := context.WithTimeout(context.Background(), 60*time.Second) // Set a 60-second timeout
 defer cancel()

 cmd := exec.CommandContext(ctx, "bash", "-c", ffmpegCmd)



the ffmpeg command looks like this :

ffmpeg -nostdin -i rtmp://localhost:1936/live/test -c:v libx264 -s 1920x1080 -f dash output/test/1080p/stream.mpd


I get the following error :


Error opening input: Input/output error

Error opening input file rtmp://localhost:1936/live/test.

Error opening input files: Input/output error

Exiting normally, received signal 2.

signal: interrupt



I have already tried to break the command, and then execute it. Something like :


cmd := exec.CommandContext(ctx,
 "ffmpeg",
 "-nostdin",
 "-i", "rtmp://localhost:1936/live/"+streamKey,
 "-c:v", "libx264",
 "-s", resolution,
 "-f", sp.OutputFormat,
 outputPath+"/stream.mpd")



After running the ffmpeg command with -loglevel debug and -report :


Here is the logs and errors I get :


When I run it within the go application :


ffmpeg started on 2024-10-02 at 12:00:06
Report written to "ffmpeg-20241002-120006.log"
Log level: 48
Command line:
ffmpeg -loglevel debug -report -i rtmp://localhost:1936/live/test -c:v libx264 -s 1920x1080 -f dash ./output/test/1080p/stream.mpd
ffmpeg version 7.0.2 Copyright (c) 2000-2024 the FFmpeg developers
 built with Apple clang version 15.0.0 (clang-1500.3.9.4)
 configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0.2_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon
 libavutil 59. 8.100 / 59. 8.100
 libavcodec 61. 3.100 / 61. 3.100
 libavformat 61. 1.100 / 61. 1.100
 libavdevice 61. 1.100 / 61. 1.100
 libavfilter 10. 1.100 / 10. 1.100
 libswscale 8. 1.100 / 8. 1.100
 libswresample 5. 1.100 / 5. 1.100
 libpostproc 58. 1.100 / 58. 1.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-i' ... matched as input url with argument 'rtmp://localhost:1936/live/test'.
Reading option '-c:v' ... matched as option 'c' (select encoder/decoder ('copy' to copy stream without reencoding)) with argument 'libx264'.
Reading option '-s' ... matched as option 's' (set frame size (WxH or abbreviation)) with argument '1920x1080'.
Reading option '-f' ... matched as option 'f' (force container format (auto-detected otherwise)) with argument 'dash'.
Reading option './output/test/1080p/stream.mpd' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url rtmp://localhost:1936/live/test.
Successfully parsed a group of options.
Opening an input file: rtmp://localhost:1936/live/test.
[AVFormatContext @ 0x13f721f90] Opening 'rtmp://localhost:1936/live/test' for reading
[rtmp @ 0x13f6040e0] No default whitelist set
[tcp @ 0x13f7223d0] No default whitelist set
[tcp @ 0x13f7223d0] Original list of addresses:
[tcp @ 0x13f7223d0] Address ::1 port 1936
[tcp @ 0x13f7223d0] Address 127.0.0.1 port 1936
[tcp @ 0x13f7223d0] Interleaved list of addresses:
[tcp @ 0x13f7223d0] Address ::1 port 1936
[tcp @ 0x13f7223d0] Address 127.0.0.1 port 1936
[tcp @ 0x13f7223d0] Starting connection attempt to ::1 port 1936
[tcp @ 0x13f7223d0] Connection attempt to ::1 port 1936 failed: Connection refused
[tcp @ 0x13f7223d0] Starting connection attempt to 127.0.0.1 port 1936
[tcp @ 0x13f7223d0] Successfully connected to 127.0.0.1 port 1936
[rtmp @ 0x13f6040e0] Handshaking...
[rtmp @ 0x13f6040e0] Type answer 3
[rtmp @ 0x13f6040e0] Server version 13.14.10.13
[rtmp @ 0x13f6040e0] Proto = rtmp, path = /live/test, app = live, fname = test
[rtmp @ 0x13f6040e0] Window acknowledgement size = 5000000
[rtmp @ 0x13f6040e0] Max sent, unacked = 5000000
[rtmp @ 0x13f6040e0] New incoming chunk size = 4096
[rtmp @ 0x13f6040e0] Creating stream...
[rtmp @ 0x13f6040e0] Sending play command for 'test'
[rtmp @ 0x13f6040e0] Deleting stream...
[in#0 @ 0x13f721d40] Error opening input: Input/output error
Error opening input file rtmp://localhost:1936/live/test.
Error opening input files: Input/output error
Exiting normally, received signal 2.



This is what i get when i run the same command on terminal :


<same as="as" but="but" please="please" scroll="scroll" further="further">

[rtmp @ 0x1437144c0] No default whitelist set
[tcp @ 0x143604f20] No default whitelist set
[tcp @ 0x143604f20] Original list of addresses:
[tcp @ 0x143604f20] Address ::1 port 1936
[tcp @ 0x143604f20] Address 127.0.0.1 port 1936
[tcp @ 0x143604f20] Interleaved list of addresses:
[tcp @ 0x143604f20] Address ::1 port 1936
[tcp @ 0x143604f20] Address 127.0.0.1 port 1936
[tcp @ 0x143604f20] Starting connection attempt to ::1 port 1936
[tcp @ 0x143604f20] Connection attempt to ::1 port 1936 failed: Connection refused
[tcp @ 0x143604f20] Starting connection attempt to 127.0.0.1 port 1936
[tcp @ 0x143604f20] Successfully connected to 127.0.0.1 port 1936
[rtmp @ 0x1437144c0] Handshaking...
[rtmp @ 0x1437144c0] Type answer 3
[rtmp @ 0x1437144c0] Server version 13.14.10.13
[rtmp @ 0x1437144c0] Proto = rtmp, path = /live/test, app = live, fname = test
[rtmp @ 0x1437144c0] Window acknowledgement size = 5000000
[rtmp @ 0x1437144c0] Max sent, unacked = 5000000
[rtmp @ 0x1437144c0] New incoming chunk size = 4096
[rtmp @ 0x1437144c0] Creating stream...
[rtmp @ 0x1437144c0] Sending play command for 'test'
[flv @ 0x143604b30] Format flv probed with size=2048 and score=100
[flv @ 0x143604b30] Before avformat_find_stream_info() pos: 13 bytes read:2263 seeks:0 nb_streams:0
Transform tree:
 mdct_inv_float_c - type: mdct_float, len: 64, factors[2]: [2, any], flags: [unaligned, out_of_place, inv_only]
 fft32_ns_float_neon - type: fft_float, len: 32, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_inv_float_c - type: mdct_float, len: 64, factors[2]: [2, any], flags: [unaligned, out_of_place, inv_only]
 fft32_ns_float_neon - type: fft_float, len: 32, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_pfa_15xM_inv_float_c - type: mdct_float, len: 120, factors[2]: [15, any], flags: [unaligned, out_of_place, inv_only]
 fft4_fwd_float_neon - type: fft_float, len: 4, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_inv_float_c - type: mdct_float, len: 128, factors[2]: [2, any], flags: [unaligned, out_of_place, inv_only]
 fft_sr_ns_float_neon - type: fft_float, len: 64, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_pfa_15xM_inv_float_c - type: mdct_float, len: 480, factors[2]: [15, any], flags: [unaligned, out_of_place, inv_only]
 fft16_ns_float_neon - type: fft_float, len: 16, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_inv_float_c - type: mdct_float, len: 512, factors[2]: [2, any], flags: [unaligned, out_of_place, inv_only]
 fft_sr_ns_float_neon - type: fft_float, len: 256, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_pfa_15xM_inv_float_c - type: mdct_float, len: 960, factors[2]: [15, any], flags: [unaligned, out_of_place, inv_only]
 fft32_ns_float_neon - type: fft_float, len: 32, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_inv_float_c - type: mdct_float, len: 1024, factors[2]: [2, any], flags: [unaligned, out_of_place, inv_only]
 fft_sr_ns_float_neon - type: fft_float, len: 512, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
Transform tree:
 mdct_fwd_float_c - type: mdct_float, len: 1024, factors[2]: [2, any], flags: [unaligned, out_of_place, fwd_only]
 fft_sr_ns_float_neon - type: fft_float, len: 512, factor: 2, flags: [aligned, inplace, out_of_place, preshuf]
[NULL @ 0x144124920] nal_unit_type: 7(SPS), nal_ref_idc: 3
[NULL @ 0x144124920] Decoding VUI
[NULL @ 0x144124920] nal_unit_type: 8(PPS), nal_ref_idc: 3
[NULL @ 0x144124920] Decoding VUI
[h264 @ 0x144124920] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x144124920] Decoding VUI
[h264 @ 0x144124920] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 @ 0x144124920] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x144124920] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 @ 0x144124920] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 0x144124920] Decoding VUI
[h264 @ 0x144124920] Format yuv420p chosen by get_format().
[h264 @ 0x144124920] Reinit context to 1280x720, pix_fmt: yuv420p
[h264 @ 0x144124920] no picture 
[flv @ 0x143604b30] All info found
[flv @ 0x143604b30] rfps: 29.666667 0.016552
[flv @ 0x143604b30] rfps: 29.750000 0.009347
[flv @ 0x143604b30] rfps: 29.750000 0.009347
[flv @ 0x143604b30] rfps: 29.833333 0.004197
[flv @ 0x143604b30] rfps: 29.916667 0.001104
[flv @ 0x143604b30] rfps: 29.916667 0.001104
[flv @ 0x143604b30] rfps: 30.000000 0.000067
[flv @ 0x143604b30] rfps: 30.000000 0.000067
[flv @ 0x143604b30] rfps: 60.000000 0.000270
[flv @ 0x143604b30] rfps: 60.000000 0.000270
[flv @ 0x143604b30] rfps: 120.000000 0.001079
[flv @ 0x143604b30] rfps: 120.000000 0.001079
[flv @ 0x143604b30] rfps: 240.000000 0.004316
[flv @ 0x143604b30] rfps: 240.000000 0.004316
[flv @ 0x143604b30] rfps: 29.970030 0.000204
[flv @ 0x143604b30] rfps: 29.970030 0.000204
[flv @ 0x143604b30] rfps: 59.940060 0.000814
[flv @ 0x143604b30] rfps: 59.940060 0.000814
[flv @ 0x143604b30] After avformat_find_stream_info() pos: 496783 bytes read:496783 seeks:0 frames:179
Input #0, flv, from 'rtmp://localhost:1936/live/test':
 Metadata:
 |RtmpSampleAccess: true
 Server : NGINX RTMP (github.com/arut/nginx-rtmp-module)
 displayWidth : 1280
 displayHeight : 720
 fps : 30
 profile : 
 level : 
 Duration: 00:00:00.00, start: 6.742000, bitrate: N/A
 Stream #0:0, 138, 1/1000: Audio: aac (LC), 48000 Hz, stereo, fltp, 163 kb/s
 Stream #0:1, 41, 1/1000: Video: h264 (High), 1 reference frame, yuv420p(tv, bt709, progressive, left), 1280x720 [SAR 1:1 DAR 16:9], 0/1, 2560 kb/s, 30 fps, 30 tbr, 1k tbn
Successfully opened the file.
Parsing a group of options: output url ./output/test/1080p/stream.mpd.
Applying option c:v (select encoder/decoder ('copy' to copy stream without reencoding)) with argument libx264.
Applying option s (set frame size (WxH or abbreviation)) with argument 1920x1080.
Applying option f (force container format (auto-detected otherwise)) with argument dash.
Successfully parsed a group of options.
Opening an output file: ./output/test/1080p/stream.mpd.
[out#0/dash @ 0x123707480] No explicit maps, mapping streams automatically...
[vost#0:0/libx264 @ 0x123707d60] Created video stream from input stream 0:1
detected 10 logical cores
[h264 @ 0x123607b70] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x123607b70] Decoding VUI
[h264 @ 0x123607b70] nal_unit_type: 8(PPS), nal_ref_idc: 3
[aost#0:1/aac @ 0x144028080] Created audio stream from input stream 0:0
Transform tree:
 mdct_inv_float_c - type: md

<it simply="simply" starts="starts" working="working">
</it></same>


I am not sure if there is something to do with Permissions.


-
Stream sent via FFMPEG (NodeJS) to RTMP (YouTube) not being received
10 décembre 2024, par QumberI am writing a very basic chrome extension that captures and sends video stream to a nodeJS server, which in turns sends it to Youtube live server.


Here is my implementation of the backend which receives data via WebRTC and send to YT using FFMPEG :


const express = require('express');
const cors = require('cors');
const { RTCPeerConnection, RTCSessionDescription } = require('@roamhq/wrtc');
const { spawn } = require('child_process');

const app = express();
app.use(express.json());
app.use(cors());

app.post('/webrtc', async (req, res) => {
 const peerConnection = new RTCPeerConnection();

 // Start ffmpeg process for streaming
 const ffmpeg = spawn('ffmpeg', [
 '-f', 'flv',
 '-i', 'pipe:0',
 '-c:v', 'libx264',
 '-preset', 'veryfast',
 '-maxrate', '3000k',
 '-bufsize', '6000k',
 '-pix_fmt', 'yuv420p',
 '-g', '50',
 '-f', 'flv',
 'rtmp://a.rtmp.youtube.com/live2/MY_KEY'
 ]);

 ffmpeg.on('error', (err) => {
 console.error('FFmpeg error:', err);
 });

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

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

 // Handle incoming tracks
 peerConnection.ontrack = (event) => {
 console.log('Track received:', event.track.kind);
 const track = event.track;

 // Stream the incoming track to FFmpeg
 track.onunmute = () => {
 console.log('Track unmuted:', track.kind);
 const reader = track.createReadStream();
 reader.on('data', (chunk) => {
 console.log('Forwarding chunk to FFmpeg:', chunk.length);
 ffmpeg.stdin.write(chunk);
 });
 reader.on('end', () => {
 console.log('Stream ended');
 ffmpeg.stdin.end();
 });
 };

 track.onmute = () => {
 console.log('Track muted:', track.kind);
 };
 };

 // Set the remote description (offer) received from the client
 await peerConnection.setRemoteDescription(new RTCSessionDescription(req.body.sdp));

 // Create an answer and send it back to the client
 const answer = await peerConnection.createAnswer();
 await peerConnection.setLocalDescription(answer);

 res.json({ sdp: peerConnection.localDescription });
});

app.listen(3000, () => {
 console.log('WebRTC to RTMP server running on port 3000');
});




This is the output I get, but nothing gets sent to YouTube :




FFmpeg stderr: ffmpeg version 7.0.2 Copyright (c) 2000-2024 the FFmpeg developers
 built with Apple clang version 15.0.0 (clang-1500.3.9.4)

FFmpeg stderr: configuration: --prefix=/opt/homebrew/Cellar/ffmpeg/7.0.2_1 --enable-shared --enable-pthreads --enable-version3 --cc=clang --host-cflags= --host-ldflags='-Wl,-ld_classic' --enable-ffplay --enable-gnutls --enable-gpl --enable-libaom --enable-libaribb24 --enable-libbluray --enable-libdav1d --enable-libharfbuzz --enable-libjxl --enable-libmp3lame --enable-libopus --enable-librav1e --enable-librist --enable-librubberband --enable-libsnappy --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libtesseract --enable-libtheora --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libxvid --enable-lzma --enable-libfontconfig --enable-libfreetype --enable-frei0r --enable-libass --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libspeex --enable-libsoxr --enable-libzmq --enable-libzimg --disable-libjack --disable-indev=jack --enable-videotoolbox --enable-audiotoolbox --enable-neon

FFmpeg stderr: libavutil 59. 8.100 / 59. 8.100
 libavcodec 61. 3.100 / 61. 3.100
 libavformat 61. 1.100 / 61. 1.100
 libavdevice 61. 1.100 / 61. 1.100

FFmpeg stderr: libavfilter 10. 1.100 / 10. 1.100
 libswscale 8. 1.100 / 8. 1.100
 libswresample 5. 1.100 / 5. 1.100
 libpostproc 58. 1.100 / 58. 1.100





I do not understand what I am doing wrong. Any help would be appreciated.



Optionally Here's the frontend code from the extension, which (to me) appears to be recording and sending the capture :


popup.js & popup.html




document.addEventListener('DOMContentLoaded', () => {
 document.getElementById('openCapturePage').addEventListener('click', () => {
 chrome.tabs.create({
 url: chrome.runtime.getURL('capture.html')
 });
 });
});






 
 <code class="echappe-js"><script src='http://stackoverflow.com/feeds/tag/popup.js'></script>




StreamSavvy













capture.js & capture.html




let peerConnection;

async function startStreaming() {
 try {
 const stream = await navigator.mediaDevices.getDisplayMedia({
 video: {
 cursor: "always"
 },
 audio: false
 });

 peerConnection = new RTCPeerConnection({
 iceServers: [{
 urls: 'stun:stun.l.google.com:19302'
 }]
 });

 stream.getTracks().forEach(track => peerConnection.addTrack(track, stream));

 const offer = await peerConnection.createOffer();
 await peerConnection.setLocalDescription(offer);

 const response = await fetch('http://localhost:3000/webrtc', {
 method: 'POST',
 headers: {
 'Content-Type': 'application/json'
 },
 body: JSON.stringify({
 sdp: peerConnection.localDescription
 })
 });

 const {
 sdp
 } = await response.json();
 await peerConnection.setRemoteDescription(new RTCSessionDescription(sdp));

 console.log("Streaming to server via WebRTC...");
 } catch (error) {
 console.error("Error starting streaming:", error.name, error.message);
 }
}

async function stopStreaming() {
 if (peerConnection) {
 // Stop all media tracks
 peerConnection.getSenders().forEach(sender => {
 if (sender.track) {
 sender.track.stop();
 }
 });

 // Close the peer connection
 peerConnection.close();
 peerConnection = null;
 console.log("Streaming stopped");
 }
}

document.addEventListener('DOMContentLoaded', () => {
 document.getElementById('startCapture').addEventListener('click', startStreaming);
 document.getElementById('stopCapture').addEventListener('click', stopStreaming);
});






 
 <code class="echappe-js"><script src='http://stackoverflow.com/feeds/tag/capture.js'></script>




StreamSavvy Capture















background.js (service worker)




chrome.runtime.onInstalled.addListener(() => {
 console.log("StreamSavvy Extension Installed");
});

chrome.runtime.onMessage.addListener((message, sender, sendResponse) => {
 if (message.type === 'startStreaming') {
 chrome.tabs.create({
 url: chrome.runtime.getURL('capture.html')
 });
 sendResponse({
 status: 'streaming'
 });
 } else if (message.type === 'stopStreaming') {
 chrome.tabs.query({
 url: chrome.runtime.getURL('capture.html')
 }, (tabs) => {
 if (tabs.length > 0) {
 chrome.tabs.sendMessage(tabs[0].id, {
 type: 'stopStreaming'
 });
 sendResponse({
 status: 'stopped'
 });
 }
 });
 }
 return true; // Keep the message channel open for sendResponse
});