pt., 14 paź 2022, 01:49 użytkownik Andre Smagin <a...@smagin.com> napisał:
> On Thu, 13 Oct 2022 22:14:33 +0200 > Alexandre Ratchov <a...@caoua.org> wrote: > > > On Thu, Oct 13, 2022 at 03:11:50AM +0000, s...@skolma.com wrote: > > > in summary, audio works.. just not bit-perfectly :) > > > does anyone know if SNDIO supports such mode ? and how i might > configure it. > > > > bit-perfect is practical for one thing only: avoid questionings about > > whether the processing adds audible noise & distortion. I've tryed > > various hacks, including bypassing sndiod and neither was very > > practical. > > > > IMHO, the sndiod resampler covers 99% of the cases. To handle the > > remaining 1%, I just resample the files off-line. audio/sox is > > excellent for that. > > > > So, I'd suggest you to add "-e s24" to sndiod_flags and resample > > off-line when needed. > > > > HTH > > >>There is possibly one more use case for "bit-perfect". I agree, one more and not the only one. I have a small > >>collection of surround sound (5.1, 4.1, quad, etc) recordings extracted > >>from various DVDs, SACDs, and other sources. They are encoded in DTS > >>and Dolby Digital formats, as plain WAV files, and "compressed" to flac > >>format to prevent "smart" applications, such as ffmpeg, mpd, etc. from > >>trying to decode them and convert to stereo. > > >>My desktop is connected to a receiver via optical SPDIF cable. To get > >>the surround sound, I use mpd with 'device "snd/0"' option and Ario to > >>control the mpd daemon. mpd decodes the top layer (flac), but stops > >>there and sends DTS-wav to the sndiod without mangling it further. > >>However, if sndiod's sample rate does not match that of the recording, > i..t resamples the stream, which ruins the DTS and results in white noise. > > I made some test with my use case, and several cli players (siren, flac123, sox itself etc.). Using the USB Audioquest Dragonfly Red with led indicating actual sample rating. > >>I found out that I have to restart sndiod with either > >>'sndiod_flags="-m play -r 44100"' or 'sndiod_flags="-m play -r 48000"' > >>flags in /etc/rc.conf.local depending on the files I am playing, > >>and then it gets to the receiver without issues. > I have plenty of audio (stereo) files encoded with different bitrates and bit depths, mostly 24/96 for "hi-res" and 16/44.1 for ripped cds of my collection. I can confirm, that using above flags and restarting sndio it is possible to force DAC to work with its native supported modes (including bit depth) and succeed, with "proper" led colours too. There was one exception - setting 88200 is possible but no audible sound then. Not even using "native" 24/88,2 files. These files were only audible at 24/96 mode. Or default 48000 and 44100 set up by hand. > >>I have each music directory annotated with the sample rate used, like so: > > >>HAMLET: /storage $ ls music/dts/Pink\ Floyd/ > >>(1970) Atom Heart Mother (Quadrophonic Vinyl Conversion) (Dolby Digital > Quad 16-48) > >>(1973) Dark Side of the Moon (Alan Parson's Mix) (DVD-Audio) (DTS 4.1 > 24-48) > >>(1971) Echoes (Original 4.0 Quad Mix) (From Pink Floyd the Early Years > 1965-1972, Volume 5) (DTS Quad 16-48) > >>(1973) Dark Side of the Moon (Analogue Transfer From SACD) (DTS 5.1 > 16-44.1) > >>(1971) Meddle (From Pink Floyd the Early Years 1965-1972, Volume 5) (DTS > 5.1 16-48) > >>(1994) The Division Bell (2014, Warner Music Group, 20th Anniversary > Edition) (DTS 5.1 16-48) > >>Live: (1974) Live at Pompeii (DTS Quad 24-48) > > >>For '16-48' and '24-48' (bit depth-samplerate), I start sndiod with > >> sndiod_flags="-m play -r 48000" > >>for '16-44.1', I restart sndiod with > >> sndiod_flags="-m play -r 44100" > It seems to be sort of, say, "workaround" for "bit-perfect" audio "purists". While it is possible to detect encoding parameters of a particular file with e.g. soxi I can imagine script resetting sndiod flags and daemon itself for expected purposes. Even if it is possible (leave for a moment question about necesserity) it is not much "elegant". I was hoping (and tried) to use audioctl -f /dev/audioX rate=ZZZZZ but no success / effect. > >>Bit depth does not seem to matter. I don't care about "bit-perfect", but > >>only about sending the dts stream to the receiver as-is, which works. > But it brings audioquest colours to proper shades, and I am affraid that is how it works in "consider-myself-to-be-the-audiophile" world, despite of fact if it is reasonable / non-audible. People choose "hi-res" files expecting better(?) audio quality and usually they get what they expect and belive to be truth. Even if it is not truth. Or not whole-truth. In my opinion telling: "it makes no sense at all, go resample or sox that stuff back to <<standard>> parameters, you won't hear the difference" is a bit simplifying. As much as "stereo is good enough for allmost everything", is it?. All in all it sounds like "go mess with volumio, or some pi-based-stuff, whatever, it is out of our scope". Seems to be not much elegant, but propably fair and honest from OpenBSD perspective, and I accept that argument in advance. Tomasz >>-- > >>Andre > > > > >
