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  • DXVA2/D3D11 encoders not available in FFmpeg for no apparent reason

    3 juillet 2020, par user1134621

    this is a weird one. I am trying to set up FFmpeg for hardware encoding of a live input into h264. I got it working fine on Linux with VAAPI. On Windows I was probably successful as well with NVENC but I do not have a NVENC capable card to check. What has been giving me a headache is encoding through DXVA2 or D3D11VA. FFmpeg picks up available hardware but it will not give me any suitable encoders. I even built FFmpeg myself to make sure that all the CONFIG_DXVA_HWACCEL switches are enabled, I can grep the binary for h264_dxva2 etc. and see that they are there. However, calling avcodec_find_encoder_by_name always returns NULL. h264_vaapi or h264_nvenc work fine though. I am obviously missing something stupid. Do you have any suggestions where to look ? Thanks.

    


  • Raw Audio Issue / Settings [on hold]

    31 octobre 2016, par GumboMcGee

    I have a raw audio file that i am trying to convert to WAV, I have run it though ffmpeg several times with different arguments (see below), and i have got it to a point where it sounds right, except for weird jumping / skipping.

    I have looked around on the internet for a fair while and cannot find any information that might help me figure out the issue,

    Hopefully someone here will be able to help me out in finding the fix.

    I have included a link to a section of the output WAV file from ffmpeg, i advise you skip to about the 2 minute mark, as it has the best example of sounding like an actual music but still does the skipping.

    ffmpeg command for WAV output :

    ffmpeg -f s16le -ar 22050 -ac 2 -i music.raw output.wav

    Link to the output WAV file

  • Merge 3 ffmpeg audio tracks into one file without normalized audio

    29 juillet 2022, par Meh.

    I'm trying to essentially get the command line version of dropping files into Audacity, where they all start at the same time and when I export it, it will export as one file.

    


    I'm using this command ffmpeg -i <file> -i <file> -i <file> -filter_complex amix=inputs=3 <out></out></file></file></file> from another question and it makes the audio/end product softer. I can definitely tell a difference without any checking. I am working with files from the musdb18 database so I can make instrumentals manually for a project.

    &#xA;

    How can I combine multiple audio files without ffmpeg doing something weird like changing the db or normalizing the audio ?

    &#xA;

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