I'm tweaking a script to make it detect mov_text and encode it to srt, but also if present i want it to copy all other subtitle streams at the same time, is there a way to do this without specifying copy for every individual stream like -c:s:0 srt -c:s:allothers copy?

Bonjour tous.
Existe t il un moyen d'utiliser ffmpeg pour oter les images en double d'une vidéos avec mpdecimate, et que le son associé à chaque image soit aussi coupé, de manière à ne pas avoir de décalage entre l'audio et la video ?
Cordalement

Eric

Hi everyone, I'm trying to get a random opaque text color set in the video for each new production. What I want is to assign a different text color to the video output each time, but for this color not to be transparent.

My simple command:

ffmpeg -i input.mp4 -vf "drawtext=text='Welcome!':fontfile=Arial-Bold.ttf:
fontcolor=random:fontsize='min(40+(4*mod(n,180)),60)':
x=(w-tw)/2:y=(h-th)/2:enable='between(t,0,5)',scale=1280:720,setdar=16/9" output.mp4

Even though the font color=random setting provides random colors, it also creates transparent colors and transparent colors that aren't fully visible in the video appear. Is there a way to get just opaque colors other than this option?

Any help is greatly appreciated.

hello I am trying to stream using ffmpeg and hls and create a link to share it through my website but when I ever try to open the link using mpv player I get this following error I added the port using iptables but the issue still going

C:\Users\HP>mpv http://148.113.44.158:8088/hls/bbbbb.m3u8

[ffmpeg] tcp: Connection to tcp://148.113.44.158:8088 failed: Error number -138 occurred

Failed to open http://148.113.44.158:8088/hls/bbbbb.m3u8.

[ytdl_hook] ERROR: [generic] Unable to download webpage: <urllib3.connection.HTTPConnection object at 0x000001CEE3521330>: Failed to establish a new connection: [WinError 10061] No connection could be made because the target machine actively refused it (caused by TransportError('<urllib3.connection.HTTPConnection object at 0x000001CEE3521330>: Failed to establish a new connection: [WinError 10061] No connection could be made because the target machine actively refused it'))

[ytdl_hook] youtube-dl failed: unexpected error occurred

Exiting... (Errors when loading file)

When I use yt-dlp, I save the audio and video separately, and use --embed-thumbnail. But, when I use ffmpeg to mux, the embedded thumbnail does not automatically get imported. So, what is the ffmpeg command for importing an embedded thumbnail?

Wondering if anyone else has this issue:

The following pipes an ffmpeg command from python to the OS terminal:

import shlex
import subprocess

command = r'''ffmpeg -i "./Subfolder/video1.mp4" -ss 00:00:04.000 -to 00:00:06.000 -c copy "./Subfolder/video1_000004_000006.mp4"'''

try:
    print("Executing cmd....")
    output = subprocess.check_output(shlex.split(command + " -n" + " -hide_banner"),
                                                     stderr=subprocess.STDOUT,
                                                     universal_newlines=True)
    print("Finished successfully....")
except subprocess.CalledProcessError as e:
    print("FFMpeg stderr triggerred")
    print(e.output)

Since file already exists, and + " -n" is appended, there should be an error message saying file exists.

When I put the ffmpeg command directly into the Windows terminal it says: File... already exists. Exiting.

Within Ubuntu Linux terminal the python script is able to show File... already exists. Exiting., but not on Windows 10 Powershell.

Hello! I used streetwarp.com and wanted to try to reverse-engineer it.

I got this video without transition between the frames.

https://reddit.com/link/1js4tez/video/eg3v30pp11te1/player

And then I also got this video with smooth transition between the frames.

https://reddit.com/link/1js4tez/video/174quwjx11te1/player

How to achieve this kind of effect? Is there a specific name I should look for?

Thanks for the help!

For example, just this :

Input #0, matroska,webm, from 'xxxxxxxxxxxxxxxxxxx':
  Duration: 00:04:00.08, start: 0.000000, bitrate: 1068 kb/s
  Stream #0:0: Video: h264 (Main), yuv420p(progressive), 854x480 [SAR 1:1 DAR 427:240], 25 fps, 25 tbr, 1k tbn (default)
  Stream #0:1: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp (default)
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native))
  Stream #0:1 -> #0:1 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':
  Stream #0:0: Video: wrapped_avframe, cuda(progressive), 854x480 [SAR 1:1 DAR 427:240], q=2-31, 200 kb/s, 25 fps, 25 tbn (default)
  Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)

instead of this log full of garbage. I just want a -hide_metadata switch

Input #0, matroska,webm, from 'xxxxxxxxxxxxxxxxxxx':
  Metadata:
    MINOR_VERSION   : 0
    COMPATIBLE_BRANDS: iso6avc1mp41
    MAJOR_BRAND     : dash
    ENCODER         : Lavf61.3.103
  Duration: 00:04:00.08, start: 0.000000, bitrate: 1068 kb/s
  Stream #0:0: Video: h264 (Main), yuv420p(progressive), 854x480 [SAR 1:1 DAR 427:240], 25 fps, 25 tbr, 1k tbn (default)
    Metadata:
      HANDLER_NAME    : ISO Media file produced by Google Inc.
      VENDOR_ID       : [0][0][0][0]
      DURATION        : 00:04:00.080000000
  Stream #0:1: Audio: aac (HE-AAC), 44100 Hz, stereo, fltp (default)
    Metadata:
      HANDLER_NAME    : ISO Media file produced by Google Inc.
      VENDOR_ID       : [0][0][0][0]
      DURATION        : 00:04:00.001000000
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> wrapped_avframe (native))
  Stream #0:1 -> #0:1 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to 'pipe:':
  Metadata:
    encoder         : Lavf62.0.100
  Stream #0:0: Video: wrapped_avframe, cuda(progressive), 854x480 [SAR 1:1 DAR 427:240], q=2-31, 200 kb/s, 25 fps, 25 tbn (default)
    Metadata:
      encoder         : Lavc62.0.100 wrapped_avframe
  Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s (default)
    Metadata:
      encoder         : Lavc62.0.100 pcm_s16le

When Adding Chapters to an MP4 Video, I'm getting: Unsupported codec with id 98314 for input stream 5

Is there anything wrong with how I'm adding the Chapter Markers?

The Unsupported codec with id 98314 for input stream 5 is as follows:

Stream #0:5[0x6](eng): Data: bin_data (text / 0x74786574)
Metadata:
handler_name    : SubtitleHandler
Unsupported codec with id 98314 for input stream 5

I'm adding the Chapters with the command:

ffmpeg -i in.mp4 -i chapters.txt -map 0 -map_chapters 1 -c copy out.mp4

chapters.txt is:

;FFMETADATA1
[CHAPTER]
TIMEBASE=1/1000
START=0
END=300000
title=Chapter 1
[CHAPTER]
TIMEBASE=1/1000
START=300000
END=600000
title=Chapter 2
[CHAPTER]
TIMEBASE=1/1000
START=600000
END=900000
title=Chapter 3
[CHAPTER]
TIMEBASE=1/1000
START=900000
END=1200000
title=Chapter 4

And f I extract the bin_data, its:

  Chapter 2
encd  Chapter 3
encd  Chapter 4
encd

We have a film scanner that will be connected to this PC to do encoding 4k 16bit and 2k 10bit FFV1/MKV.

From my understanding Ffv1 is less about GPU and more about CPU.

I was thinking of the following specs

Ryzen Threadripper Pro 7995WX

256GB 8x32 DDR5-5600 ECC

4TB x 4 NVME Raid0

2TB Main OS NVME

RTX 6000 ADA 48GB

Nvidia Mellonx Nic 10/25 SFP 28 connected to our storage Server

Is this decent enough?

Hi, I am new to programming and building a video making script in Python. I am stitching some simple images into a static video, adding some subtitles and a VoiceOver Audio. It's a simple project, and working absolutely fine in my Mac, but when I am dockerizing this script and running the image in a container, the output video has a very high-pitch distorted sound. I am using native AAC decoder, locally it's working fine, I wanted to use libfdk_aac but could not use it as it's not free.

I wanted to know how to resolve this Audio issue, is there something I can do.

For reference, here is the python code which is responsible to attaching audio to video:

audio_cmd = f"ffmpeg -y -i {subtitled_video} -i {audio_path} -map 0:v -map 1:a -c:v copy -c:a aac -b:a 192k -ar 44100 -ac 1 -shortest {final_output}"

subprocess.call(audio_cmd, shell=True, stdout=subprocess.DEVNULL)

There was some bitrate mismatch in the audio and subtitle_video, but that above code should take care of it as per ChatGPT. Can someone please help me with this? It would be great

Hey folks! 

I’ve been having issues with transcoding v210 mov files into mkv using ffmpeg. Every time I try I get the following error: 

[in#0/mov,mp4,3gp,3pg2,mj2 @ 0x600000b5cd00] Error during demuxing : Device not configured

[[in#0/mov,mp4,3gp,3pg2,mj2 @ 0x600000b5cd00] Error retrieving a packet from demuxer: Device not configured

[out#0/matroska @ 0x60000025c780] Error closing file: Bad file descriptor 

This is issue has been happening on an iMac with an OS of Sequoia 15.3.2 and the same error has occurred with ffmpeg versions 7.1.1 and 6.1.2. On a different iMac running Mojave 10.14.6 OS and ffmpeg version 5.0 the transcode goes through fine. 

This is the script I’m using:  

ffmpeg -i input_file -map 0 -dn -c:v ffv1 -level 3 -g 1 -slicecrc 1 -slices 16 -c:a copy output_file.mkv -f framemd5 -an framemd5_output_file

Any ideas as to what the issue is and how to fix it? Any insight is appreciated!

Is there a way to keep the audio so that it sounds normal when output is slowed down? Maybe by multiplying the audio bitrate? Or will doing so changes the pitch?

For example, a long Vlog, and I speed up the video 5x, but when i see parts im interested in, i can slow down the video and the audio wont be patchy. This is handled video-wise by multiplying the framerate so ffmpeg doesnt drop any frame. For audio? Also is there a way to speed up the the subtitle without burning it? Thanks.

Also, what exactly preset does? Should I use ultrafast? (In my specific case storage isnt an issue)

Hi,

i have this script

-c:v hevc_amf -rc cqp -qp_p 20 -qp_i 20 -qp_b 20 -c:a copy -c:s copy -map 0 -map_metadata 0

and it works fine

How can i use 10 bit?

Thanks

They have one for YouTube but I can't find recommended upload settings for Instagram or Facebook?

I’m trying to convert a 16bit WAV file with a bitrate of 1536kbps and a sample rate of 48khz to a PCM file useing the command

ffmpeg -ar 48000 -ac 1 -f s16le -I track1.wav output.pcm

but I keep getting the errors “sample rate too large” and “could not write header (Incorrect codec parameters ?): invalid argument”

ffmpeg also states that the bitrate is 768kbps, the previously listed bitrate was reported by windows. What do I need to do to get this to work?

Here's 2 (related) questions about FFmpeg that Goolge / Bing, & so on are of no help in answering.

(Q1) Does FFmpeg have an "Auto tone" function (for the video, not the audio)?

(Q2) If it does, how do I apply "Auto-tone" to a video? By that, I mean what is the range of values? Specifically, what value(s) would equal minimum low & high, without equaling zero?

I've got a series with HD video but wrong language and this same series in SD but with right audio, so I want to put the right language on the HD files.
I came up with this :
ffmpeg -i videoHD.mkv -i videoSD.mkv -c:v copy -c:a copy output.mkv
but I don't know how to tell ffmpeg that I want it to take the audio from the second file. Also, the second file has 2 audio tracks and I want to use the second one, so there should be a -map 0:a:1 somewhere, right ?

I've messed around with various settings and the video tag in mdx won't work on Iphone Chrome and Safari. It works everywhere else - Android, Linux, Windows.

I also need an option for stronger compression so that it doesn't make a 25MB mp4 from a 3MB webm, but about the same size.

``` ffmpeg -i overview.webm -c:v libx264 -c:a aac -strict -2 -b:a 192k -preset fast -movflags +faststart overview.mp4

<video width="160" hight="90" controls autoplay loop muted playsinline> <source src={OverviewMP4} type="video/mp4" /> </video> ```

I am writing cpp code for encoding a stream into abr hls stream of segment size 4 seconds , I want to add scte markers in stream , I am writing scte marker in manifest.m3u8 but there is a need to break .ts file if marker comes between start and end time of .ts file, is there a way I can split .ts file of 4 seconds for e.g in 1.5 and 2.5 sec segments .

Looking for a ffmpeg expert to join an ongoing product. If interested, dm me :)

So I've been playing with aevalsrc to create sample waveforms that sound interesting. I found some weird behaviour when writing to ogg files. I was playing with use pow to produce waves that are "squarer".

The following filter works

ffplay -f lavfi -i 'aevalsrc=pow(sin(120*t*2*PI)\,1.1)'

As does,

ffmpeg -filter_complex 'aevalsrc=pow(sin(120*t*2*PI)\,1.1), atrim=duration=10s' output.wav ffplay output.wav

But if I use an .ogg file instead of a .wav file. I get silence.

ffmpeg -filter_complex 'aevalsrc=pow(sin(120*t*2*PI)\,1.1), atrim=duration=10s' output.ogg ffplay output.ogg

But then if I remove the pow and create a pure sin wave it works. I can also convert the wav to an ogg file without problem.

ffmpeg -filter_complex 'aevalsrc=sin(120*t*2*PI), atrim=duration=10s' output.ogg ffplay output.ogg

(It also works if the exponent in pow is an integer). Some experiment suggests this is something to do with values not being "aligned" with samples.

If I use aformat=s32 in the pipeline the ogg file is not silent

ffmpeg -filter_complex 'aevalsrc=pow(sin(120*t*2*PI)\,1.1), aformat=s32, atrim=duration=10s' output.ogg

Any ideas on what's going on? I would quite to be able to see the sample formats at different parts of the filterchain.

For a bit more confusion, if I use -sample_fmt libvorbis demands this be fltp - but this still produces silence - as does using format=fltp in the chain

I'm not that good at coding But im trying to combine the given commands together without a success,

ffmpeg -hwaccel d3d11va -i input.mp4 -vf "hflip" -c:v hevc_amf output1.mp4 #mirror effect

ffmpeg -hwaccel d3d11va -i input.mp4 -vf "colorbalance=rm=0.05:gm=0.02" -c:v hevc_amf output2.mp4 #slight_warmth effect

ffmpeg -hwaccel d3d11va -i input.mp4 -vf "setpts=PTS/1.2" -filter:a "atempo=1.2" -c:v hevc_amf output3.mp4 #speed 1.2x

ffmpeg -hwaccel d3d11va -i input.mp4 -vf "scale=iw1.2:ih1.2,crop=iw/1.2:ih/1.2:(iw-iw/1.2)/2:(ih-ih/1.2)/2" -c:v hevc_amf output4.mp4 #zoom 1.2x

ffmpeg -hwaccel d3d11va -i input.mp4 -filter_complex "[0:v]crop=w=iw:h=8:x=0:y=(ih-8)/2,gblur=sigma=10[blurred_strip];[0:v][blurred_strip]overlay=x=0:y=(H-8)/2" -c:v hevc_amf output.mp4 #adds a blurred line

I have used multiple ai models with reasoning but none of those results worked. If I merge these commands into a single command nothing happens on cmd.

note that im trying to use hardware accleration using an amd vega6 igpu and i have properly build ffmpeg with amf support as all of these commands works well individually.

Can someone please help me ?

I am developing a real time speech to text system. I split the work in two steps:

Step 1 - Receive the video, extract the audio, send into speech-to-text model, and obtain words from the speech to text system. Everything in a real time manner, by calling the ffmpeg command with the flag -re. I can see that this is working since my python scripts start to return some .srt segments after some seconds.

Step 2 - Burn the .srt segments from step 1, as hard captions, in the video and stream (through RTMP or HLS). For this, I am using the ffmpeg command below, with video filter for subtitles. The subtitles file is a named pipe, which is receiving words from step 1

````
ffmpeg -i input.mp4 -vf "subtitles=named.pipe.srt" -c:v libx264 -c:a copy -f flv rtmp://localhost:1935/live/stream
````

However, the ffmpeg command only starts after the script of step1 is completed, losting the real time beahviour. It seems it waits the end of the close of the named pipe to be able to read instead of start reading as the program starts.

I am not surprised since it seems that ffmpeg is not that preprared for real time captions. But do you no if I am doing something stupid or if I should use other approach? What you recommend?

I want to avoid the CEA-608 and CEA-708 captions, but I already know that ffmpeg does't do this.