On Mon, Jan 09, 2023 at 01:10:09PM -0700, Ashlen wrote: > > Although I need to finalize the Perl script I was using to do this (life gets > busy), in practice I was able to distinguish between samples created by > audio/sox and aucat(1) in informal AB/X testing on my 7th generation X1 Carbon > with HiFiMan Sundara headphones plugged in. To describe the circumstances + > outcome briefly: 9 out of 10 correct in 10 trials; randomly sampling from an > array containing the givens A and B to get an unknown X; comparing 15 seconds > of > audio; audio/sox as the playback software. In the future, I would do >=16 > trials, and perhaps conduct the tests from my desktop instead since it has a > discrete amp and DAC. > > In offline resampling from 48kHz to 44.1kHz, the highs were most affected and > that's what I was able to use to distinguish between samples. The percussion, > especially the cymbals, sounded different in particular because the clip > resampled by aucat had cymbal crashes that seemed to 'shimmer' much less (the > decay was more rapid). The spectrograms seemed to confirm that the highs were > most affected. > > Whether that means "low quality resampling" or merely that the results of the > two commands can be differentiated is something I'm uncertain of. Either way, > I > don't know enough about C or sndio internals to be useful in that domain yet. > As > an aside, I did find this to be a useful resource for learning about digital > audio resampling, and they recommend audio/sox there. > > https://ccrma.stanford.edu/~jos/resample/ > > I hadn't said anything about this earlier because I wanted to take the time to > finish + document the script, reproduce my results with a royalty free sample > at > a greater trial count, and then post. Given that I haven't done so yet, I can > at > least post the commands used to resample the audio for those that are > interested. > > > # This was originally an opus file downloaded with www/yt-dlp. > # Converting to WAV so both SoX and aucat can work with it. > $ opusdec input.{opus,wav} > > # Resample 16-bit 48kHz WAV file to 44.1kHz using both SoX and aucat(1). > # > # If I recall correctly, I converted to FLAC here because the WAV headers > # generated by aucat and SoX differed, and so SoX would refuse to play WAV > files > # created by aucat. > $ sox -G input.wav -t wav - rate -v 44100 | flac - -o output-sox.flac > $ aucat -i input.wav -h wav -r 44100 -e s16 -o - | flac - -o output-aucat.flac > > # Generate spectrograms for later inspection/comparison. > $ sox output-sox.flac -n spectrogram -o spectrogram-sox.png > $ sox output-aucat.flac -n spectrogram -o spectrogram-aucat.png >
Thank you for all the details. Do you mind sharing the youtube link (or the input.wav file) so I can experiment with the same file and examine the data > > I'd certainly be interested in the ability to play audio in a way that avoids > resampling altogether, similar to what a user can do on FreeBSD with the > following sysctl tweaks: > > # sysctl hw.snd.maxautovchans=0 dev.pcm.0.bitperfect=1 > This is equivalent to bypassing sndiod. The OpenBSD equivalent is as follows: chown <your_user_id> /dev/audio* rcctl stop sndiod Note that bit-perfect audio and avoiding resampling are not the same thing. The latter is more complicated but it would keep current features working (mixing, routing, device switching) by making sndiod dynamically set device sample rates to match the sample rate of a particular client (for instance an audio player), other programs (if any) would resample.
