LinuxMusicians

I use a 2nd audio interface on my jackd with I can use the q-parm (quality) up to 4. 0 means linear interpolation, 4 means sinc best quality and a whole bunch of CPU load that needs -p512 (as I found in jackd's source code).

The resampling is only needed because the 2 interfaces cannot be synced. But since the signal gets resampled from 48000 to 48000 I doubt that the quality parm is important. There is nothing to be interpolated as I understand.

So I guess -q0 is perfect in this case. I can't hear a difference to -q4, though. What do you think?

It gives you the magic binary code music that catapults your music into antimatter, ~Luzifer_hack

No, I have no idea. I think a good sound card brings more with less CPU load.
Shm do you use? RAM instead of hard disk.

The previous post was more of a ~humor.

Is this possibly due to a higher (=better) resolution of the floating point number (https://en.wikipedia.org/wiki/Floating-point_arithmetic), which of course also means a higher calculation effort (higher CPU load)?
This improves the calculation of the individual music processing steps (real, close to, for example, music to be realized with analog hardware), which enables an exact / more precise reproduction of your originally intended / wanted music?

@q-parm Does anyone have any background knowledge?

khz wrote:@q-parm Does anyone have any background knowledge?

Default is q=0 and the lowest, and q=4 the highest quality. It uses the Secret Rabbit Code aka libsamplerate. A list of what the option choses:

• q=0: SRC_LINEAR
• q=1: SRC_ZERO_ORDER_HOLD
• q=2: SRC_SINC_FASTEST
• q=3: SRC_SINC_MEDIUM_QUALITY
• q=4: SRC_SINC_BEST_QUALITY

You can see the results of a few tests at: http://src.infinitewave.ca/

THX4--info \o/

But this doesn't help much in the special case of resampling from 48000 to 48000 to work around the missing sync of two audio interfaces. Even -q0 sounds fine for me.

I synchronized my RME cards with SPDIF. The cards also have a word clock and an internal sync port.
Is there anything about this in the manual of your used sound card?
(Or do you mean something else?)
https://en.wikipedia.org/wiki/Word_clock
https://www.rme-audio.de/english/techinfo/sync.htm (German: https://www.rme-audio.de/techinfo/sync.htm); (https://www.rme-audio.de/download/hdspeaio_e.pdf)
http://manual.ardour.org/synchronization/

I use 2 Focusrite Scarletts (USB). They can't be synchronized so I have to load audioadapter or use other stuff that does resampling.

I think you need to synchronize the soundcards over one of the possibilities mentioned above. If it doesn't offer your sound card it won't work. The audioadapter does not do this. IMHO
I have audioadapter +

khz wrote:I synchronized my RME cards with SPDIF. The cards also have a word clock and an internal sync port.

khz wrote:I think you need to synchronize the soundcards over one of the possibilities mentioned above. If it doesn't offer your sound card it won't work. The audioadapter does not do this. IMHO
I have audioadapter +

khz wrote:I synchronized my RME cards with SPDIF. The cards also have a word clock and an internal sync port.

Jackd's audioadapter, alsa_out, zita_j2a are workarounds for those cases when the audio adapters cannot be synced. I am happy that there are such workarounds.

I am also glad that there are such workarounds.
In the audio field I have little knowledge. That's why I find sharing knowledge so excellent.