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#ffmpeg

4 posts4 participants0 posts today
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2/3

A little more reading of some non-AI internet had me try just changing the file type suffix from webm to mkv.
That's #Matroska, #MKV - a container.

And the video (only) plays on the #SamsungTV base model code #JU6400.
And the TV tells me the audio format ( .opus) is not supported.

But #FFMPeg made short work of transcoding audio. Under 3 minutes instead of hours. Went to aac. Still renaming the webm file to a .mkv container.

This image stems from a mistaken perception of what #FFmpeg truly is. Contrary to viewing it as a fragile, makeshift tool like many #opensource projects, FFmpeg is more akin to the base of a pyramid, a solid foundation upon which the entire video industry relies. Its free, open-source framework powers countless #multimedia operations, from format conversion and compression to live streaming and real-time encoding, yet the industry often exploits this invaluable resource without reciprocation.

Need some Owncast help. I've been having this problem, consistently, for months now. I appreciate any troubleshooting advice anyone might have.

I'm using the out of the box ffmpeg from the Debian 12 repos, but suspect I might need to compile ffmpeg on my own, with some special options, but honestly have no idea where to start there, or what I'd be looking for.

github.com/owncast/owncast/dis

Once in a while, I'll start streaming and everything will be fine for a few minutes, then the video freezes and shows the loading animation. Stream Health shows latency and download spikes. Hardwar...
GitHub"no such file or directory" errors several minutes into stream · owncast owncast · Discussion #4110Once in a while, I'll start streaming and everything will be fine for a few minutes, then the video freezes and shows the loading animation. Stream Health shows latency and download spikes. Hardwar...
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Reference mix is on top; our mix is on bottom. These are normalized to -1 dB peak, because I am specifically evaluating the mix. (If I just wanted to compare how they sounded and which I preferred, then it would make sense to normalize to perceived loudness, the default for #ffmpeg-normalize.)

Our mix has more dynamic range, prob too much. That fits _my_ tastes, but it's not what the market prefers. You can see some limiting in the ref mix, but not excessive.

I was puzzling over why so many of my spoken word audio tracks were suddenly so muted, and I thought I was going to have to re-process all 70 of them.

Y'all, the bluetooth volume on my phone was set low.

XD

But at least now I know how to batch-detect quiet tracks with #ffmpeg!

Replied in thread

@Edent
If I got the video in binary form, I could read enough of the #ffmpeg man page to adjust the brightness and contrast myself. The results wouldn't be as good as a specialist could produce, certainly, but they'd be enough to tell me whether the enhanced video could plausibly have been derived from the original.

For non-geeks, my answer is quite different: we can't be sure of any evidence presented in court, because witnesses, police officers and experts could be lying. That's why we have perjury laws with such stiff penalties. If you can't get your hands on the evidence yourself, you ultimately have to trust the system.

Continued thread

#ffmpeg nachinstalliert. Das löste das Problem unter #Windows. Hat aber auch nicht geholfen. Wahrscheinlich bleibt mir nix anderes übrig, als extra dafür nun eine VM mit Windows aufzusetzen. Wollte ich eigentlich nicht.

Leider habe ich keine Video Schnittsoftware als #Opensource gefunden, die die Features von Cuttermaran bietet.
* Schnitt auch auf B-Frame
* Geschnittene Videos werden nur kopiert. Nur die Schnittstellen werden über den #Encoder geschleust.

Any idea?

Update: Thanks to @furicle for this suggestion. I think it's about perfect:

tmp $ AV_LOG_FORCE_NOCOLOR=true ffmpeg -hide_banner -i example.opus -filter:a volumedetect -f null /dev/null
Input #0, ogg, from 'example.opus':
  Duration: 02:13:19.89, start: 0.007500, bitrate: 118 kb/s
  Stream #0:0(eng): Audio: opus, 48000 Hz, stereo, fltp
      Metadata:
        encoder         : Lavf58.45.100
[Parsed_volumedetect_0 @ 0x563ea07eeb00] n_samples: 0
Stream mapping:
  Stream #0:0 -> #0:0 (opus (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
Output #0, null, to '/dev/null':
  Metadata:
    encoder         : Lavf61.7.100
  Stream #0:0(eng): Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
      Metadata:
        encoder         : Lavc61.19.101 pcm_s16le
[Parsed_volumedetect_0 @ 0x7f9920003c00] n_samples: 767987856
[Parsed_volumedetect_0 @ 0x7f9920003c00] mean_volume: -21.0 dB
[Parsed_volumedetect_0 @ 0x7f9920003c00] max_volume: -2.8 dB
[Parsed_volumedetect_0 @ 0x7f9920003c00] histogram_2db: 1
[Parsed_volumedetect_0 @ 0x7f9920003c00] histogram_3db: 70
[Parsed_volumedetect_0 @ 0x7f9920003c00] histogram_4db: 3872
[Parsed_volumedetect_0 @ 0x7f9920003c00] histogram_5db: 98331
[Parsed_volumedetect_0 @ 0x7f9920003c00] histogram_6db: 750534
[out#0/null @ 0x563ea084bf80] video:0KiB audio:1499976KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: unknown
size=N/A time=02:13:19.87 bitrate=N/A speed= 573x    

Dear sound/audio folks and engineers,

[Update: just for clarity: I'm looking for a command line utility that will help me decide which of 70 audio recordings need amplification/compression/normalization. Something that can print out media stats like average loudness, or something like that]

I have a directory with 3.5GiB of audio files (chiefly opus & m4a) which are spoken word recordings.

Some of them are quite low, and some of them are quite dynamic such that it's a whisper at times and nearly a shout at other times.

I've processed a lot of them with #audacity's compressor filter or #ffmpeg (ffmpeg -i audio.m4a -filter:a "speechnorm=e=50:r=0.0001:l=1" audio-normalized.m4a), but there are some unprocessed files in the collection, which are a pain to individually find and fix.

Is there a way from the #CommandLine to detect the loudness and/or dynamic range of audio files so that I can automatically flag them for processing with ffmpeg?

Thanks!!

Replied in thread

#ffmpeg template (in development) containing all commands for fast and simple editing of snippets, in this case with the files to create the audio file above this comment:
leak_bitPicRecovery_mdCC.mp3
media.tupambae.org/colaborator…

original template:
media.tupambae.org/colaborator…

@deusfigendi
So viel zu den quantenphysischen Momenten wenn ein Kommntar ueber duckduckgo-AI im Stream just in den work flow passt .. :)

@aiquez @crossgolf_rebel @mina

I made a video and an audio track sound. I synchronized the two with ffmpeg. They are well in sync on my (linux) and on a friend's (mac) laptop.

But when I load and read it on my (android) phone, the audio is slightly (.1-.2 sec?) but visibly out of sync. I can't upload this on Loops. Annoying. Any one has an idea of what I can do?

I always find that TikTok downloadable videos have a great size to quality ratio. With dark magic, a friend figured out what #ffmpeg options they use. Here you go:

ffmpeg -i input.mp4 \
-vf "scale=576:1024:flags=lanczos" \
-c:v libx264 -profile:v high -level 3.1 -preset slow \
-b:v 385k -maxrate 420k -bufsize 800k \
-c:a aac -b:a 37k -ac 2 -ar 44100 \
-movflags +faststart \
output_tiktok_style.mp4

I did this amazing recording of a tree swaying in the wind and knocking at another tree the other day for my upcoming mini exhibition at Notam. It was recorded by sticking a Geofin in between the roots to get the sound of the wind from the perspective of the tree.

I will resynthesize this as a vibrator motor signal for the exhibition too. Pretty excited!