8 Apr
2025
8 Apr
'25
2:07 p.m.
#1. most computers cannot do better than 7 or 8 miliseconds AD-DA latency...
Buffer size is different in different interfaces: PCI, USB, FireWire...
ADC & DAC chips have more latency if are 1st gen "older",
USB: 256x2 is much better vs. 128x5
has the same audio latency measured with an oscilloscope,
but CPU works relaxed with 256x2 = less Xruns.
#2. Generic Kernel is useless for Audio.
install lowlatency kernel &/or liquorix kernel.
#3. RME hdsp 9632 pci alsa driver works ok up to 2013,
since 2013, source code has small "Cosmic Ray Bit Flips",
RME Alsa driver has weird glitches,
because Alsa driver is Not activated permanently,
driver activates every-time i pay an audio file, and deactivates when audio stops,
the first 7 seconds of audio buffer are corrupted.
but Jack is Activated continuously,
it does Not matter.
#4. when using low latency:
requires CPU running Fast all the time "Performance" mode,
install cpupower-gui
or similar
and
CPU-X
to verify CPU speed.
some Linux Generic Kernels, when are "On Demand"
does Not allow Single core CPU to reach 100%, are limited to 50%.
"to save battery power."
#5. Pulse audio requires manual configuration to change sample rate to 48KHz or more,
and Bit Depth from 16 to 24 or more.
if you play a 48KHz-24-Bits file using PulseAudio,
PulseAudio will convert to 44.1KHz,-16-Bits
same:
some versions of VLC only play at 44.1KHz has a Bug,
even if PulseAudio .config is set ok.
#6.
i have the temptation to buy a Digidesign 003
would be interesting to test.
Digi002 had terrible sound, requires too much modifications.
Digi001 had nice sound, transparent, PCI but does Not have Alsa driver.
________________________________________
From: Florian Paul Schmidt <mista.tapas(a)gmx.net>
Sent: Tuesday, April 8, 2025 2:02 AM
To: linux-audio-user(a)lists.linuxaudio.org; Jay Thomas
Subject: [LAU] Re: Jack Frames/Period setting with Firewire
On April 7, 2025 8:36:15 PM GMT+02:00, Jay Thomas <jayt0808(a)gmail.com> wrote:
Hello
I am having difficulty setting Frames/Period for Jack at 128 to keep the
system from having stuttering audio quality. The highest I can go is 64
but that is not high enough to resolve the stuttering issue. Jack
complains of ALSA not being able to set the parameters: "ERROR: ALSA:
cannot set hardware parameters for capture"
The driver being used is snd-fireworks. This module has a config parameter
called resp_buf_size that can be set to 4096 maximum. So when I set this
parameter to 4096, I still have trouble setting the Frames/Period in Jack
to 128. It would seem that one would expect that 4096 would be good enough
since I am using an AudioFire device with 16 channels being sent over the
FW bus. 1 sample = 4 bytes. 4 bytes x 16 channels x 128 = 4096. So this
figure of 4096 bytes is enough to hold 128 samples of 16 channels of audio.
Is there another driver module configuration parameter that I need to
change instead of the snd-fireworks parameter such that ALSA will be able
to set the requested buffer size that propagates down from Jack config?
Hi, sorry for top-posting.
A buffer needs to hold at least two periods for jackd. And two times 64 is 128.
Regards,
FPS
_______________________________________________
Linux-audio-user mailing list -- linux-audio-user(a)lists.linuxaudio.org
To unsubscribe send an email to linux-audio-user-leave(a)lists.linuxaudio.org