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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 - 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
I've included an audio conversion to high bitrate aac in all of the commands, but this will not actively improve your audio. 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 ...
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 Convert audio files to mp3 using ffmpeg same …
https://stackoverflow.com/questions/36012173/ffmpeg-convert-audio-files-to-mp3-using-ffmpeg-same-quality
MP3 is a lossy codec. If you want to minimize quality loss, use a high bit rate. For example ffmpeg -i audio.ogg -c:a libmp3lame -q:a 0 newfile.mp3 will use the highest quality VBR (variable bit rate) preset from the Lame MP3 encoder, which will usually give you a file with an average bit rate between 220 and 260 kb/s. If your input file has a relatively low bit rate (e.g. …
Repair Corrupt Video Files using FFmpeg - Stellar Data ...
https://www.stellarinfo.com/blog/repair-corrupt-videos-using-ffmpeg/
FFmpeg is a command-line tool that uses commands to convert audio or video formats. For example: To extract sound from a video and save it as MP3, you can use command line: $ ffmpeg -i video1.avi -vn -ar 44100 -ac 2 -ab 192 -f mp3 audio3.mp3
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.
Convert AC3 audio track to AAC from a video : ffmpeg
https://www.reddit.com/r/ffmpeg/comments/9dzzpy/convert_ac3_audio_track_to_aac_from_a_video/
Try ffmpeg -i input -vcodec copy -acodec aac -strict -2 output.mkv. It seems like it’s only converting the audio because no video is defined. I’d also recommend defining the audio bitrate:-ab 128K or -ab 320K for better audio.
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