On Tuesday 24 January 2006 22:45, Fabrice Bellard wrote:
> If you want to model the real PC speaker, the best to do is to generate
> a square signal and to pass it thru a low pass filter with a cut off
> frequency of a few kHz. Then you could even be able to play samples thru
> the simulated PC spe
Sebastian Kaliszewski wrote:
Joachim Henke wrote:
Ok, these are really strong arguments. Thanks a lot for your
interesting statements! I'll do some testing on square waves and will
post an updated patch, as I am also not totally satisfied with the
current sound myself.
One little sugges
Joachim Henke wrote:
Ok, these are really strong arguments. Thanks a lot for your
interesting statements! I'll do some testing on square waves and will
post an updated patch, as I am also not totally satisfied with the
current sound myself.
One little suggestion...
Real PC-speaker is rath
Ok, these are really strong arguments. Thanks a lot for your
interesting statements! I'll do some testing on square waves and will
post an updated patch, as I am also not totally satisfied with the
current sound myself.
Sincerely
Jo.
Sebastian Kaliszewski wrote:
Well, it sounds rather dul
Joachim Henke wrote:
I still prefer using a sine wave, it sounds more smooth and won't hurt
our ears (and speakers) too much.
Well, it sounds rather dull, and even worse, on non hi-fi computer speakers
(which is 90% of PC users use) low frequencies (<100Hz) are simply
unhearable (since equip
I still prefer using a sine wave, it sounds more smooth and won't
hurt our ears (and speakers) too much. I don't think that people want
to play square waves, when they have their PC connected to their hifi
system (c: The main purpose of the PC speaker is to generate tones
and many programme
Hi,
On Fri, 20 Jan 2006, Joachim Henke wrote:
> Thanks for the hint! I assume, the reason, why floating point calculations
> should be avoided, is to be compatible with processors like ARM, that don't
> necessarily have an FPU.
>
> Yes, I'll rewrite the waveform generation stuff to fit in fixed
malc wrote:
I'd like to not one thing, namely, you are using FPU to generate the
samples. This is something Fabrice expressed dissatisfaction with. In
the case of speaker it might be feasible to switch to fixed-point
calculation.
One more note about that. PC-speaker generates just plain square
Thanks for the hint! I assume, the reason, why floating point calculations
should be avoided, is to be compatible with processors like ARM, that don't
necessarily have an FPU.
Yes, I'll rewrite the waveform generation stuff to fit in fixed point. It
should also be faster then.
Jo.
malc wrote:
On Thu, 19 Jan 2006, Joachim Henke wrote:
Thanks a lot for your help! I wasn't aware that AUD_write could return zero.
Seems that I was just lucky, it didn't do that with 44100 Hz sample rate (c:
You are welcome.
Now it works fine with 32000 Hz, and I really encourage people (especially PC
Thanks a lot for your help! I wasn't aware that AUD_write could
return zero. Seems that I was just lucky, it didn't do that with
44100 Hz sample rate (c:
Now it works fine with 32000 Hz, and I really encourage people
(especially PC users) to test the attached patch. Call QEMU with the
swi
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