Sample Sound

[Martha Vanschaick]

Soundpool is specific to the project(limited slots) and allows insertion of samples complete with parameter locks along side current samples on the same track without changing them, without them any new samples inserted would take on the current track parameters. Sound banks are universal and when used replace the current sample with a new one complete with parameter locks, only one per track.

Be careful using the soundpool between different projects: if you use a sound with a sample in slot 57 in one project, it will use sample slot 57 in another project which might be a totally different sample.

You can then plock the sample slot to change a sounds sample for that trig. Hold the trig and use the sample slot pot to select the sample. You may also plock a different, distinct sound from the sound pool.

To be able to load a sample into the SOUND POOL it must be saved as a sound in the PROJECT and accessible in the SOUND MANAGER.

First load the sound from your computer onto your Rytm. Then load that sample into your PROJECT. Assign the SAMPLE to TRACK. Then hit YES/SAVE & MUTE/SOUND and a list of available slots to assign the SOUND appear. Hit YES/SAVE and name the SOUND. You can now access it in the SOUND MANAGER where you can LOAD it to the SOUND POOL.

If you have a particular sound that is embedded into a wave file containing other sounds, the only mechanism anyone has for isolating that sound and excluding all other sounds is through filtering BUT this probably won't work in a general sense because....

A particular sound may be (say) a flute and its characteristic timbre is produced by a series of harmonics related to the base pitch of the note - these harmonics will extend through the audio spectrum and overlap other instruments that are also in the wave file. Disentangling a flute harmonics from other sounds is very, very hard to do effectively.

We, as humans, may recognize that a flute is being played (in the presence of other sounds) and if we can play the flute we could recreate that sound fairly well BUT it would be an interpretation of the flute part of the wave file.

If the sound you wish to isolate is limited to a certain range of the spectrum and other sounds in the wave file do not encroach into that part of the spectrum then simple band-pass filtering should be successful. Anything more complex and the process is rather like trying to turn a baked-cake back into it's original constituents.

You can use equalization as stated above. You can also adjust the stereo imaging by playing with the phase. Adjusting the phase could potentially allow you to isolate vocals from the rest of the track if the vocals are mixed hard center. I've used this process successfully to hide the bass and vocals and hear what else is happening in a song.

Here's what I'd do: 1. Clone your track twice (you'll have 3 channels of the same track)2. Flip the phase (reverse polarity) on one channel. Hard pan one left, hard pan one right.3. Turn down your headphones or monitors and slowly move the center channel up and down. Adjusting the level will cause cancellation of the center channel and either bring up or remove elements from the center of the mix.

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I have tried playing with equalization, hi-pass/low-pass, and various filters, but I can't seem to get anywhere close to this effect, where the "s" sounds come off as static, and the annunciation of consonants is muted (possibly an attack filter?). I can't find any documentation through Google search, either.

I know very little about the sound architecture of the NES, other than that music was composed with Square and Triangle waves, and the "samples" were presented through a separate sound channel. On NES sound emulators I use, this channel is usually labeled "PCM," (Pulse Code Modulation?) The NES system itself was 8-Bit, but I do not know if its sound system was.

Like Shane said, use at most 8-bits resolution, but also [re-]sample it around 4kHz or less. No lowpass filter necessary as you want the higher frequencies to wrap "poorly" while sampling at the low sample rate. (Though, some re-sampling actions on audio editors may do that lowpass for you).

That sound is due to a combination of low sample rate and low bit depth. Those are the 2 parameters you can control to diminish the data size on PCM data, which they had to do for the old game consoles because of their limited storage and processing power. A very low sample rate without pre-filtering introduces strong aliasing, and a very low bit depth introduces distortion on the signal.

Its possible that the reason that you are having problems recreating the sound you want is that the samples you are trying to recreate might have been recorded/played back using the NES sample channel's DCPM (differential pulse-code modulation) capabilities

This is where instead storing each audio sample directly the chip stores the first sample and then the delta's (how much the signal goes up or down between samples). I read a blog somewhere (I'll try to find it and link) where somebody analysed the behavior of this channel and found that it introduced lots of strange artifacts into the sounds

Now my idea; with the new auto sampler I could somehow sample the sound from the hardware device, right?

As I never routed from GP to the RD2000, just from the RD2000 to GP I have no idea how to wire it?

You might use Auto Sampler to capture audio files, but you probably should continue your work in full featured sampler, like Kontakt, which offers all the tools required to map sample files to zones on a keyboard and help with all the transitions, loop points detection, etc.

I have managed it already quite good with the audio file import function of Omnisphere.

It works very good in a certain key range.

I recorded C4 and it sounds good in about 2 octaves up and down.

If I would add a audio file of C2 and C6, and layer them in Omnisphere it may gives a better result.

You need to write down the loop point and root key somewhere, as well as the rate. Hit Export WAV file and save it as something easily identifiable. I like to include the loop point, root key and sampling rate in the file name for easy reference.

Before we import the sample into the game, we need to know whether or not the sampling rate needs to be adjusted to be played properly in-game. If you miss this step you may find that your sample plays at the wrong pitch, which may not be pleasant. Or it might be really pleasant, depending on what the interval is :P:

Just icing on the cake. Sound sample is much larger than other resources like graphic and text. It will run out of free ROM space soon. It can be optimized with compression. Graphic uses LZ77 compression, text uses Huffman compression and sound sample uses DPCM compression.

I made an installer for m4a_hq_mixer based on Improved Sound Mixer.

However, since m4a_hq_mixer has larger code than Improved Sound Mixer, I had to increase the number of bytes to copy to RAM.

And unfortunately, the routine to adjust the end of the compressed wav song was placed at the end of the code.

So that part of the code had not been copied and was in a mass of nop, which resulted in the code running out of control.

Although I'm not aware of a way that Java can produce audio samples for us at this time, if that changes in the future, this can be a place for it. I know that JavaFX has some stuff like this, for example AudioSpectrumListener, but still not a way to access samples directly.

You can use any processing supported by the mixer or its component lines, by querying for Control objects and then setting the controls as the user desires. Typical controls supported by mixers and lines include gain, pan, and reverberation controls.

Don't assume the bytes are meaningful audio samples! Unless you happen to have an 8-bit AIFF file, they aren't. (On the other hand, if the samples are definitely 8-bit signed, you can do arithmetic with them. Using 8-bit is one way to avoid the complexity described here, if you're just playing around.)

So instead, I'll enumerate the types of AudioFormat.Encoding and describe how to decode them yourself. This answer will not cover how to encode them, but it's included in the complete code example at the bottom. Encoding is mostly just the decoding process in reverse.

(Notice that the most positive value in two's complement representation is 1 less than the most negative value. This is a minor detail to be aware of. For example if you're clipping audio and forget this, the positive clips will overflow.)

To decode PCM samples, we don't care much about the sample rate or number of channels, so I won't be saying much about them here. Channels are usually interleaved, so that if we had an array of them, they'd be stored like this:

The byte array contains the sample frames split up and all in a line. This is actually very straight-forward except for something called endianness, which is the ordering of the bytes in each sample packet.

The two's complement sign must be extended. This means that if the most significant bit (MSB) is set to 1, we fill all the bits above it with 1s. The arithmetic right-shift (>>) will do the filling for us automatically if the sign bit is set, so I usually do it this way:

Java does sign-extension automatically when converting from one integral type to a larger type, for example byte to int. If you know that your input and output format are always signed, you can use the automatic sign-extension while concatenating bytes in the earlier step.

Recall from the section above (How do I coerce the byte array in to meaningful data?) that we used b & 0xFF to prevent sign-extension from occurring. If you just remove the & 0xFF from the highest byte, sign-extension will happen automatically.

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