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Encode/AAC – FFmpeg
https://trac.ffmpeg.org/wiki/Encode/AAC
Video coversion using ffmpeg losing audio quality - Stack ...
https://stackoverflow.com/questions/28451423/video-coversion-using-ffmpeg-losing-audio-quality
You haven't set any options for audio bitrate or quality, which means that ffmpeg will choose the best AAC encoder it is configured with (which will probably be libvo-aacenc or libfaac ), and it'll choose a default audio bitrate. Since you're converting something that's already compressed with a lossy codec, you'll lose even more quality.
FFmpeg Codecs Documentation
https://ffmpeg.org/ffmpeg-codecs.html
A description of some of the currently available audio encoders follows. 8.1 aac. Advanced Audio Coding (AAC) encoder. This encoder is the default AAC encoder, natively implemented into FFmpeg. 8.1.1 Options b. Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this option is unspecified it is set to ...
ffmpeg - How to get peak audio quality and lowest video ...
https://video.stackexchange.com/questions/29551/how-to-get-peak-audio-quality-and-lowest-video-quality
If you're looking to optimize the audio for voice, check out ffmpeg's highpass, lowpass, and anequalizer audio filters to filter out unwanted high end and low end frequencies, and to enhance vocal frequencies (probably around 1khz and 5kz). This line processes all of one file type in a folder. Just cd into the appropriate directory.
FFmpeg
https://ffmpeg.org/index.html
Licensing has always been an issue with encoding AAC audio as most of the encoders have had a license making FFmpeg unredistributable if compiled with support for them. The fact that there now exists a fully open and truly free AAC encoder integrated directly within the project means a lot to those who wish to use accepted and widespread standards.
Which audio encoders in FFmpeg support 8 kHz? - Super User
https://superuser.com/questions/1492214/which-audio-encoders-in-ffmpeg-support-8-khz
Which audio encoders in FFmpeg support 8 kHz? aac, aptx, aptx_hd, dca, flac, g723_1, libfdk_aac, libmp3lame, libopus, libspeex, libvorbis, real_144, wavpack, many pcm variants. There are probably others, but reporting of supported_samplerates is inconsistent. I would like to transcode it into something modern.
ffmpeg and libmp3lame produces bad audio quality? - Unix ...
https://unix.stackexchange.com/questions/7284/ffmpeg-and-libmp3lame-produces-bad-audio-quality
ffmpeg -i ~/"$2.flv" -sameq -acodec libmp3lame -vol 200 -ar 44100 -aq 300 -ab 2097152 ~/"$2.avi" I already set -aq (audio quality) to 300, but no difference to 100 or 200. Moreover 100 is the max. value in my opinion. -ar (frequecy) 44100 should be ok too and the bitrate in bit/s (-ab) should be 256kb/s (2097152 / 1024 / 8).
audio - FFmpeg - resampling from …
https://stackoverflow.com/questions/22822515/ffmpeg-resampling-from-av-sample-fmt-fltp-to-av-sample-fmt-s16-got-very-bad-so
I've also noticed that, in old ffmpeg, aac is recongized as FLT format and directly decoded into 16-bit PCM, while in new ffmpeg, aac is counted as FLTP format and produce still 32-bit IEEE float output. Thus the same code will produce quite different outputs with different versions of ffmpeg.
how to make audio quality lower : ffmpeg
https://www.reddit.com/r/ffmpeg/comments/ierdz2/how_to_make_audio_quality_lower/
I have a command for routing audio input to output, it's about as basic as it gets: ffmpeg -f alsa -acodec pcm_s32le -i hw:3,0 -f pulse out. And yet it will introduce about 4 seconds of latency. Do this in Audacity and it is instant. Let's try from pulse to pulse: ffmpeg -f pulse -acodec pcm_s16le -i default -f pulse out. 4 seconds delay.
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