I use RMAA as your measuring engine.
http://audio.rightmark.org/index_new.shtml
We are trying for a good 16-17 bits total range from these 24 bit
A/D's. 20*log10(2^16) = 96 dB. That said, we NEED the 24 bit A/D's
to get those good 16 bits but we do not need to set the sound system to
24 bits to get the good 16-17 bits. Peter is trying to justify his
programming preferences which include not using the more complex tools
to open sound cards, for example using WAVEFORMATEXTENSIBLE in the MME
interface is quite a bit more complex and if you do not need it and
cannot justify it, why do it? On the other hand, those cards that use
the highest end chips, such as the Lynx L22, Emu 1212M, and Audiophile
192, can all use larger than 16 bits and there is no intermediate step,
you have to go all the way.
Bob
Everett Palmer wrote:
Hello Peter,
What software and/or hardware do you use to measure soundcards?
Thanks,
Everett, KG6RYB
At 01:06 PM 5/19/2006, Peter Martinez wrote:
From G3PLX:
I am just starting to do some serious tests with my new 24-bit Firebox. With
the audio input of a channel short-circuit, if I analyse the statistics of
the 24-bit data, I find the bottom 8 bits are ALWAYS random. In fact the
bottom 9-10 bits are random.
This tells me there is no point at all in having 24-bit audio. I might just
as well stay with 16-bits.
Has anyone here done any rigorous tests comparing 16 and 24-bit audio and
found any difference? Like patching the code to mask off the bottom 8 bits
completely, or replacing the bottom 8 bits with random data. Maybe someone
has added a checkbox to turn this sort of test feature on and off and see
what, if any, is the difference in the speaker.
73
Peter
--
AMSAT VP Engineering. Member: ARRL, AMSAT-DL, TAPR, Packrats,
NJQRP/AMQRP, QRP ARCI, QCWA, FRC. ARRL SDR Wrk Grp Chairman
Laziness is the number one inspiration for ingenuity. Guilty as charged!
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