When I try to play Amazon Prime videos on my Fedora laptop, they never start. Instead, the screen gets stuck on a loading animation (spinning circle) indefinitely. How can I fix this? Is there a step-by-step guide or checklist I can follow? [Edit July 31] This appears to be hardware dependent. It works okay on my desktop (Intel i7-4790 with GeForce GT 730) but fails on my laptop (Intel Celeron N4500 with Jasperlake embedded GPU) despite having followed the checklist on the Fedora website. Just today, I found the

utility. On both systems, it says in the properties section:

SUBSYSTEM drm
DEVTYPE drm-minor

I'm trying to do streaming from a source (tv box transmitting in multicast in this source rtp://@X.X.X.X:Y) to Internet (to my mobile phone as example or another device inside of my LAN) but I can not achieve it. The command I'm using is something like this ffmpeg -i rtp://@X.X.X.X:Y -vcodec copy -f mpegts udp://127.0.0.1:1234 But it does not work as I expected, I mean, I'm able to open vlc and play the streaming in the same machine I'm running ffmpeg but not in another machine in the same LAN. Somebody can help me? Thank you! EDIT: Finally I solved installing a software called "udpxy" that forwards multicast content to the clients. I installed in a raspberry and it works perfect for this purpose. Thank you for all your explanations. It helped me to understand what I want to do and the limitations I have using a transcoder. I guess I can do the same with udpxy with ffmpeg but I can publish directly the TV Box IPs.

I want to encode a videostream to H.264. But when I start the gstreamer program I get the following error. No Element >>x264enc>x264<< ```

I am trying to make a perfect mobile webcam fitting my needs with the RPI and camera module I have here already. I run Arch both on the RPI and on the Laptop I intend as the "client". I have the cam running and can pipe its data from raspivid somewhere, and I am able to ssh back and forth. Now I am trying to get the transmission via the local network to work. Ideally, I would start the "server" script on the RPI and be able to connect as many clients as often as I want until I decide to close the server again. And, ideally, the delay from having a separate device and wireless transmission is almost unnoticeable so that I don't have to compensate and try to sync up video and audio. So from what I read, using VLC is out of the question. Also ideally, the "client" would make the stream available as /dev/video1 or whatever — though currently I just want to get it into OBS *somehow*. Until now I tried to follow some tutorials particularly involving netcat, but the process stopped immediately or as soon as a disconnect happened, as I was unable to play the stream on the client. Any hints on where to look or how to plan this? I thought about using something like a media server for that dynamic client (dis)connect thing, but it would need to be super lightweight.

I am trying to play back a dash manifest using ffmpeg but I keep running into the issue of ffmpeg ending with the error

[http @ 0x671e580] Stream ends prematurely at 2250, should be 18446744073709551615

The dash manifest is for a livestream. I have tested the live stream in VLC player on Windows and MX player on Android and it plays perfectly fine.  The issue seems to be just with ffmpeg. Has anyone had this issue before or know what the cause may be? I have tried telling ffmpeg to reconnect but it just loops this error.  I have tried two different MPD live streams: * $Time$ based, and * $Number$ based The issue happens only with the Number based live stream.

I'm looking for an existing Linux command-line tools that can accomplish the following: Consider there is an input stream /dev/inputstream which spews an infinite stream of data that looks like this: A A B A C Z A . . . **NOTE:** (Each line is one character from the alphabet). The tool is used in the following fashion: cat /dev/inputstream | tool The output of the tool should be like the one from "top" command. It should be a continuously updated list of top N frequently occurring items. Eg: A -- 10 times B -- 8 times Z -- 7 times C -- 2 times D -- 1 time If the stream were not infinite I could have accomplished this using sort and uniq.

I am trying to simulate a radio with a PTT button. Connected users should be able to live voice chat in the group, one at a time, while holding a button. I have a Ubuntu 22.04 server running, with nginx, but unfortunately nothing seems to work. I don't know exactly what a good method should be. I tried using sockets but i keep getting console errors (on Firefox) like: Media resource blob:https://mywebsite could not be decoded, error: Error Code: NS_ERROR_DOM_MEDIA_METADATA_ERR (0x806e0006) How exactly would a live audio browser transmission protocol look like? async function startTransmission() { if (!mediaStream) { try { mediaStream = await navigator.mediaDevices.getUserMedia({ audio: true }); mediaRecorder = new MediaRecorder(mediaStream, { mimeType: "audio/webm;codecs=opus" }); mediaRecorder.ondataavailable = (event) => { if (event.data.size > 0) { socket.emit("audio-stream", event.data); } }; mediaRecorder.start(300); // Send audio in 300ms chunks logMessage("🔊 Transmitting Audio..."); } catch (error) { console.error("Microphone access denied:", error); logMessage("⚠️ Microphone access denied."); } }} and server: socket.on("audio-stream", (audioData) => { io.emit("receive-audio", { audioData, mimeType: "audio/webm;codecs=opus", senderId: socket.id // Send the sender's ID so they can ignore their own audio });});

Using Debian 12 VM. Is there any way to mount in the filesystem, a singular remote file (typically media, e.g. mkv) accessed via HTTP/S path? I've tried various solutions: rclone - won't mount a singular file, only directories httpfs2 - doesn't seem to work with ssl I specifically want to mount these files on demand, so that they may be played via a local media server once indexed, and such media servers are file based, they cannot play remote media/HTTP streams. Performance is also paramount here. Any help or advice would be appreciated.

I prefer streaming Youtube videos via mpv, mpv https://www.youtube.com/watch?v= However, for long videos sometimes the audio breaks down in the middle, leaving error messages like > [ffmpeg] tls: Error in the pull function. Restarting the video is starting it from the beginning. How can I either directly in the command line start from, let's say, minute 30, or alternatively jump in the mpv interface to minute 30 *without* pressing 30 times the up button (which also has issues if done too fast)? I haven't found an answer related to that issue in the man page. Is it possible at all? Also alternative suggestions for streaming are welcome.

I am working with a mod_cgi application and a client that sends data in chunked form over a period of time. But the issue is on the server, my cgi script can't reply until the client terminates. I can use wireshark with curl to send chunked requests and validate that I am sending them correctly:

for i in $(seq 5); do date; sleep 1; done | \
 curl --noproxy '*' -H "Transfer-Encoding: chunked" \
     -H "Content-Type: application/json" -X POST -T - \
     https://example.com/test.cgi 

I can see the request start, and chunked messages start streaming. But, on the server, running Apache 2.4.57, I am using mod_cgi with the following flags:

Options +ExecCGI
	AddHandler cgi-script .cgi

The application is invoked but, no reply is possible until after the request is done. How can I get Apache to invoke the CGI script after the connection is established, request made, but data is incomplete? I found a link here with someone saying it is possible, but I see nowhere else online saying how to do it: https://stackoverflow.com/a/284869/2926815 Edit, I also found this link of someone who pointed out how to do the curl client-side with this https://unix.stackexchange.com/a/469060/510722 (Which is how I got my arguments for testing) - and I did validate the curl is actually sending the chunked data at the correct times.

When streaming videos on YouTube using firefox-esr and firefox, the video playback is choppy with an uneven frame-rate. CPU utilisation is high; 20% - 25% when streaming a 4K video on a Ryzen 5800X, as reported by tops. Internet connection is strong with 80Mbp/s. OS: Debian 11 CPU: Ryzen 7 5800X GPU: AMD Asus ROG Strix 6900XT

I need to work on the framebuffer to record what is being rendered. Now, I figure out how to convert the framebuffer in an image, I'm currently unable to obtain a constant stream of data from the fb to create a video. /dev/fb > fb When I do something like this my redirection just drops me and I honestly don't know how to work in another way with the framebuffer. How I can obtain a stream?

What is a handy way to record live audio stream (mp3) from terminal, using command line? The OS is Ubuntu 16.04.6 LTS

I'm capturing an RTMP stream as HLS (saving playlist files) using FFMPEG wihout any stream processing, merely copying it. Thanks to such setup I'm able to run tens of processes (I've tried 80 at most), with memory and CPU consumption not exceeding 30%. This is my command:

ffmpeg -i  -hls_init_time 10 -hls_time 10 -hls_list_size 0 -f hls -strftime 1 -hls_segment_filename '%H-%M-%S-%s.ts' -hls_flags split_by_time -codec:v copy -codec:a copy ./play$(date -d "today" +"%M%S-%s").m3u8

Unfortunately, as the number of processes grows, I start losing some HLS segments (HLS consists of a playlist file, and segments composing that file). For example the HLS plays fine from 0:00 until 0:40, but then freezes at 0:40 for 10s (my segments are 10s long) and unfreezes at 0:50. With 20 FFMPEG processes running, the loss rate is about 5%, at 80 processes its close to 50%. I'm performing those benchmarks at AWS EC2 instances using Ubuntu 22.x, and various machine types. No matter the machine type, the error's frequency remains the same, thus I assume it's not CPU or Memory related. I suspect that's something related to I/O and file writes, either an EC2 or Ubuntu setup. Any ideas?

While trying to stream online with my friends, I had an issue where everyone else's computer can watch the stream but mine, and even my android device can watch it. I found [this test page from GitHub Mozilla](https://mozilla.github.io/webrtc-landing/pc_test_no_h264.html) that revealed I don't have H.264 support for WebRTC, while all my friends did. - I tested this on Firefox 112.0.1, both the Debian package firefox and the build from the official website. - I tested this on Chromium 112.0.5615.138 (Official Build) built on Debian 12.0, running on Debian 12.0 (64-bit) (provided by Debian package chromium). - I have installed libx264-160 and libx264-164. - I have installed libopenh264-7. Replacing it with competitor libopenh264-cisco7 makes no improvement. - I can watch H.264 encoded videos offline with MPV and VLC. - ffmpeg -codec lists h264 support: DEV.LS h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 Nearly all documents and SE questions about H.264 support for Firefox are dated from around 2014-2017, which I infer this means this is an old, solved issue. If that's the case, why isn't Firefox detecting support for H.264? I use Debian Unstable (next release bookworm).

For some reason clients are dropped / playing hangs between tracks. As if server side doesn't use back2back/circular buffer, and just drops buffer content when changing song. Anyone experienced this? Not sure if it's in ices2 or icecast2. Can't get support for these open source projects.

Does a streaming version of split exist somewhere in Linux? I'm trying to back up a large amount of data via SSH, but SSH's single threaded encryption is limiting the transfer. This machine doesn't have hardware AES support so I'm using ChaCha encryption, but the cpu is still not keeping up with the network. So I thought I could solve this by splitting the data stream in two, and sending each over a separate SSH connection, and then merging the streams together at the destination. That way the encryption load can be shared over multiple cpu cores. This looked like a general enough idea that it should already exist, but I can't find it. *edit*: for some numbers, I'm backing up data from an old computer, a few hundred GB over a gigiabit wired network. I'm copying an image from a partition, as that is faster than doing individual file access on the spinning rust drive, so technically it is random access data, but it is too large to treat it as such. I tried compressing it, but that doesn't help a lot. The data isn't very compressible. So what I'm looking for is a split (and corresponding merge) that will split a stream of binary data into multiple streams (probably splitting by fixed chunks).

I have an embedded system running a custom distro built using Yocto. Later down the line, I will have a usb camera plugged into the system, but for not I would like to use a .mp4 file as a "dummy" camera. My goal is to stream the video file over USB as a USB UVC device to a computer so that the computer sees my embedded Linux system as a USB webcam: I can't find any documentation on the subject and I am a bare metal c developer dipping his toes into embedded Linux development. I would like to know what to look for and the broader steps to get this working. (including modifications to the distro through Yocto.) Thank you.

I'm trying to concatenate multiple input files/streams into one stream (using the imaginary command stream-cat), pipe that stream into ssh and on the remote host separate it back into individual files/streams (stream-sep), like in this example, which is for demonstration purposes only:

stream-cat ( zfs receive tank/vm@snapshot ) somefile.txt"

Explanation of the example: zfs send outputs a large stream of data whose size isn't known in advance (that's why tar can't handle it). That data stream is concatenated with the content of a regular file somefile.txt. The resulting stream is piped into ssh, where it is separated again. The first stream gets piped into zfs receive, while the second is written to a regular file. Such a program should be straightforward to implement by reading non-seekable streams in chunks and always writing the chunk size followed by the data, until the end of the stream is reached. The overhead would be minimal. Does such a program exist already?

I can play without any issue 1080p videos on Netflix via an extension. Is there something similar for Firefox? I've tried searching the web but can't seem to find anything. I have the latest version of Firefox and have only 2 extensions: uBlock Origin and DuckDuckGo Privacy Essentials. I have tried disabling them but it doesn't fix the issue...