making a flowstone game

[Lex248]

wlangfor@uoguelph.ca

Try to adjust SFZ synth for the sound. It is based on the waves array.

viewtopic.php?f=2&t=34301

[wlangfor@uoguelph.ca]

wlangfor@uoguelph.ca

Try to adjust SFZ synth for the sound. It is based on the waves array.

http://www.dsprobotics.com/support/view ... =2&t=34301

Yes, ultimately that is My goal. But the problem is that it comes down to adding support for velocity zones.
Furthermore, in some instances; there is looping, and in others you wish to play the whole sound (without the requirement for holding a note down). So, as you might have gathered by that there is a need for a type of sampling other than just this method. And, if the same boolean pulse method were used then as you might expect it comes down to testing for any signal above zero.. so:

Code: Select all

sound=sound*(in!=0)&1;

to initiate the boolean to be turned on. It may be difficult to test for the quieter signals; and where that code was placed is just a guess for the time being.

In time I will have something finite to look at.

[wlangfor@uoguelph.ca]

So I have been making a sampler that allows loading audio files in tandem and sequence.

Furthermore, it unloads all files after use. Like, take for instance there is probably a few different instances of intro music/sound so you need to have a transition between the two. I made one for that with a slight crossfade.

Next I needed something that allowed to mix many wav's together to compile together a track that plays for the space of an hour. The trick IMO is to make it change based on the tension of the game. And that comes down to the threat levels in the game and risk.

That one has a few different samples. But here is the point I wanted to make: Wav samples are too large. I've tried many clients to bring the size of a wav down, but I recently tried NCH mixpad; I used wav, with pcm and then I chose a bitrate of 180. It meant a 40 mb file turns out to be 2.4 mb.

The savings alone, make wav a tangible realistic thing. But, it doesn't load within the FL studio version. However, it does load to VST's made with 3.0.6 (i think) & later. So, food for thought, If you make samplers etc you might want to check out mixpad, or try out the free version of wave pad before you buy it. If it does indeed convert to this in batch files, please let us all know. It's useful info!

Regards all,
Robert

[wlangfor@uoguelph.ca]

I noticed (in follow up) that the wav's that are formatted with MP3 like formatting sound a bit crispy. I was able to combat that by using the following code with the stream multiplies. it's a form of dezipping that has a nice effect. You lose a tiny bit of volume however.

Code: Select all

streamin in;
streamout out;

float ratio=0.995;
float denorm=1e-009;

movaps xmm0,out;
addps xmm0,denorm;
subps xmm0,denorm;
subps xmm0,in;
mulps xmm0,ratio;
addps xmm0,in;
movaps out,xmm0;


In the process of making a unique sample player btw. It's really handy, will post soon.

[trogluddite]

it's a form of dezipping that has a nice effect.

More generally, it's the code for a first-order low-pass filter (6 db/oct), where the 'ratio' determines the cutoff frequency. So we could say that dezipping is a form of low-pass filtering (though it's less obvious for the stock dezippers).

[wlangfor@uoguelph.ca]

it's a form of dezipping that has a nice effect.

More generally, it's the code for a first-order low-pass filter (6 db/oct), where the 'ratio' determines the cutoff frequency. So we could say that dezipping is a form of low-pass filtering (though it's less obvious for the stock dezippers).

Yes, but in the case of lower sample rates what it really comes down to is the differentiation between continuous time signals and discrete time signals.

It causes any source of multiplier to react on an intermittent continuous basis and this dezipper does not fully reach 1 so its subtractive nature seems much like the compression thresholds emulating rarefaction. So, it has a "vivid" effect. Ultimately though, less noticeable with a 192 sample rate, notably.

[trogluddite]

Ultimately though, less noticeable with a 192 sample rate, notably.

Indeed - I should have said; "...where alpha and the sample-rate determine the cutoff frequency."!

[wlangfor@uoguelph.ca]

Ultimately though, less noticeable with a 192 sample rate, notably.

Indeed - I should have said; "...where alpha and the sample-rate determine the cutoff frequency."!

Mm, yes. Before I started to really learn DSP I always wondered why people would use 192, but I guess it is closer to a continuous time signal. In fact too, I suppose a record player's needle because of it's constant connectivity also was one of the few mediums where a signal was something like continuous time.

[MichaelBenjamin]

.

[Spogg]

In ultimative HiFi mode the program would have a sample rate at 2 * ( 1 / ( 5.39 × pow(10,-44) ) ), which covers the frequency up to planck length. To get the most subsonic bass wave however, you would need to record for a very very long time. A problem still not solved in normal HiFi Apps. So most apps cut at something like 50hz bc that is the minimum usual recording time anyways (20 milliseconds) for the human to differentiate. Any sound longer than this may contain residue Low wave frequency content however, since the filter is never perfect. There are many more interesting problems in digital representations, this is just one.

Please excuse my ignorance but I just don’t understand what you wrote, sorry.

May I ask for a simpler explanation, suitable for an old fool like me?

I don’t really understand the connection between frequency, data rate and recording time.

Cheers

Spogg