Hi all,
A little venting today about something I've worked on as a sideline while doing an
interesting test with phaser/filter banks and different sample rate multicompression for
extracting much better reconstructable audio from A grade music.
As most will know, I like, to some extend, running digital audio processing prototype
setups on Linux, using Ladspa plugins and some more programs, that are even available on
the humble Raspberry Pi and such, with the "Jack" (jackd deamon) streaming audio
processing, connected to the Alsa "sound system". This works good in that you can take for
instance a Usb Audio Class 2.0 audio device (in my case like a Yamaha MGxxXu series mixer,
or a Xmos based high quality ground separated DIY audio device), connect the jack server
to the Alsa device with proper settings (sample rate buffer size and number #channels) and
you'll be able to use jack-based processing (for instance with "jack-rack") where you have
perfect sync with the audio card/device, no sample conversions or sample clock issues.
This way, you also will know (in most cases), while pushing the processing limits such
that the processor gets hot with the amount of audio processing, when so called "x-runs"
are taking place, indicating the your computer is starting not to keep up with the amount
of audio work. so far, so good.
No, I've a need for multiple sampling domains, as in that I want parts of my processing to
take place in 96kHz sampling rate, some at 48.0/44.1kHz and some at 192 or even 384 (the
highest I've tried, and even an I-3930 at 4.5GHz runs out of juice at some point that
way), with specific rate conversion types in between. Now, for instance there's a tool
(for those interested, called alsa_out/alsa_in) that allows you to connect from a jack
domain running at a certain rate to connect to an Alsa sound device running at another
rate, and do a sort of PID controlled stretch based conversion fo quality 1 through 4
(heavy uni-core load). For connecting a sound card to a lower frequency sampling domain
that works fine.
Now, I've used a virtual "Loopback" device, with this tool as well, and audio
player/playback with streaming conversion, and sync based and other off line file
conversion, and even analog methods (quality DAC out possibly through filter, to quality
ADC in), but I'd like to easily setup a few Jack based sampling domains, with exact given
live conversions (back and forth) in between, all synced to one (1) specific clock, either
a software clock or hpet clock or (preferably as well) a single high quality usb audio
clock. Appears to be not prepared in the given software...
T. verelst
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