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I am trying to change audiobook .aaxc to mp3 or any other audio format but conversion failed is showing.
ffmpeg -i test.aaxc -c:a aac test.mp3
I am beginner using ffmpeg on windows. there is so much errors so also guide to how to analyse there errors. one error is like Error while decoding stream #0:0: Invalid data found when processing input .
As llogan ask for i am sharing complete log file
Following information from this patch to ffmpeg, you need a key/iv pair, which is unique for each book, to decrypt aaxc files. This key/iv, pair is provided by a special license request to the audible api.
The key/iv pair have to be decrypted from the license request response. Please take a look on this issue in my audible github package for more informations how to obtain the key/iv pair.
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4-byte activation key.
ffmpeg -activation_bytes 1CEB00DA -i input.aax output.mp3
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please, who can help me here with a small doubt...
I have a PNG image (720x1280) and I would like it to be an overlay of a MP4 video.
I have a MP4 video (720x405) with 10 minutes leght.
This MP4 video I would like to put in position it at coordinates x/y 0x437 of this overlay (align center).
I want to use the ffmpeg command to create a new MP4 video.
So, MP4 video with 10 minutes lenght will hav a PNG overlay image with 10 minutes length too.
Could anyone support this idea?
Thanks so much.
I create a PNG overlay and a MP4 video.
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i need to watermark a video file.
i have FF-MPEG Linux server and ff-mpeg is installed by default
plz reply me the command of watermarking for below details
1. The source file is Input.mp4
2. the Target file is Output.mp4
3. the watermark file Watermark.png
please send me the Linux commands.
is there anyone know, please reply soon
According to this site it should be something like:
ffmpeg -i Input.mp4 -i Watermark.png -filter_complex "overlay=10:10" Output.mp4
10:10 are the coordinates from the top and from the right for the overlay. (In this example 10 pixels from the top and 10 pixels from the right)
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I have a Lumix camera which, like most new cameras, record video in AVCHD format. The files get segmented into 2 or 4 GiB segments because of the limitations of the filesystem used on the memory card.
When I transfer the files to my linux computer to edit them I naturally want to have each video in a single file, which is no problem at all for linux's filesystems. So, how can I losslessly join these segments, maintaining a/v-sync?
(With Avidemux 2.6.8 I can append these segments, but it leads to nasty distortions at the cut point.)
The solution, which seems to work with my files at least, turned out to be very simple:
ffmpeg -i "concat:00000.MTS|00001.MTS|00002.MTS" -c copy output.mts
One still has to figure out which of the files belong together, though.
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I'm trying to compress an mp4 file to reduce the file size. I am using a Debian box.
The file is around 701 MB. First, I tried bzip2, which compressed it down to like 700MB, then I tried lrzip with zpaq, took like 5 mins to compress, and only brought it down to like 695 MB... Am I doing something wrong?
mp4 (or MPEG4) is already a very compressed video format, which uses advanced coding specific for video.
You won't be able to compress it more using loseless zip algorithms like bzip. What you can do is transcode the file to lower its video quality using some kind of transcoder (such as HandBrake).
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how do I get a list of amplitudes from a audio file using a linux command line tool ?
Do you mean getting all the individual samples as text? SoX can do that.
$ sox file.wav file.dat
will take an audio file file.wav, and generate a text file file.dat with a column for the timebase in seconds, and a column for each audio channel scaled by the maximum possible value.