Eric Welsh is the author of the most used patch set for timidity++, the MIDI to WAVE converter and player. As well as constantly refining the patch set, he is working on a mod2mid converter which is in the working prototype stage.

related links:
Patch set and info
Sound Fonts Guide



Eric A. Welsh July 4th, 2000 "What are sound patches, what are they used for, and how do I make a good one?" A patch is simply a collection of audio samples together with information that tells hardware/software how to use those samples to play music from a MIDI file. A long time ago, before most home computers could even play 8-bit stereo sound (long live the Amiga!), before there was even a General Midi standard, there were only expensive hardware MIDI tone generators and synthesizers. Professional musicians would pop in a floppy with their instruments on them and load them into the RAM on their hardware, then use them to play music on their keyboard that didn't sound like a plain old square wave. There were many different synths, each one with it's own proprietary patch format, yet each containing the same important information. I won't discuss any more history of midi hardware or midi standards here, save to mention a few important events related to playing MIDI on a home computer. In the early 90's, Advanced Gravis came out with the first PC sound card that I know of that used "wavetable" RAM to hold patches that could be used to play midi. For the first time, your average Joe could buy a cheap Gravis Ultrasound and start playing some decent sounding MIDI without having a hugely expensive tone generator. In the mid 90's, as computers became more powerful, "software synthesizers" began to appear that would read in these same patches and use them to play MIDI files just as a real wavetable sound card would, lowering the cost of playing midi even further and virtually removing any limitations on total memory and sample size. No matter what you use to play the midi file, whether it be a tone generator, wavetable soundcard, or software renderer, they all work the same way and use the same information. Some patch formats may support more features than the ones I will describe below, but this should cover most of the important ones and are the most likely to be supported in any format you choose to use. I'll start out by describing the different levels of information in a patch and what that information is used for. A generic patch is structured like this: Instrument Layer 1 Region 1 Sample 1 Region 2 Sample 2 etc ... etc ... (Layer2, etc...) not all formats support multiple layers Let's start out at the lowest level, the Sample. Most people probably know that this contains the raw audio data and the frequency it was sampled at. However it also contains other information that is vital to it's use as an instrument, such as the key, loop points, and loop type. The key tells it which note on an 11 octave scale it is to be associated with. When the player wants to play a note other than the one that the sample is associated with, it has to resample the sample to a different frequency to change it's pitch. It looks up a frequency table associated with each note, gets the new target frequency, then proceeds to do the resampling. I'll get to how this effects sound quality a little later on. The loop information is very important for use as an instrument. If a note is held longer than the sample length, the player needs to loop back and play part of the sample over again to keep the sound going for the requested amount of time. The loop type tells it whether it is a normal, bi-directional, or reversed loop. It can often be tricky to choose good loop points, I'll discuss this further down as well. Every sample is associated with a Region. Each region is associated with one sample, and tells the player almost everything else it needs to know about using the sample in an instrument. First, the player needs to know what range of notes is associated with this particular sample. If a sample is sampled at C4, you'll probably only want to use it to play notes between, say, C3 - C5, since resampling too far beyond the key note can stretch / shrink the sample too much and start to sound bad. The other important information stored in the region is all of the Envelope settings, including chorus, reverb, and panning overrides. The envelope settings are used by the player to embellish the sample and make it sound even better (or worse...) than it already does. Adding a good envelope to a patch can usually greatly improve how it sounds, by making it sound more realistic by setting volume ramps for Attack, Decay, Sustain, and Release regions of the sample, plus adding a base level of reverb. Each Layer is just a collection of regions. It can also contain global chorus, reverb, and panning information, but some patch formats / players do not understand this, so I'd not recommend trying it :) The assembled regions in the layer should cover all of the notes in the scale, or at least as many as are supported by the format (typically from C0 - C9). An instrument may have more then one layer, but not everything understands this. This is typically done so that one layer covers the right channel, while the other covers the left, so that two samples are played for each note and create a surround effect. This is not generally a good idea IMHO, but some sound card vendors do it in an attempt to make their MIDI sound better. I feel that it is better to just leave everything in mono and let the player do fancy stuff like add surround sound effects. Just use one layer, keep the samples mono, and don't set any panning or chorus in their envelopes either. This is just my opinion though, there are plenty of SF2 banks out there that would beg to differ with me :) Now that I've discussed how a patch is structured, I'll give a quick list of things to do and NOT to do when creating one. The most important part of the patch is, of course, the sample itself. The easiest pitfall to avoid, and unfortunately one of the more common ones I see, is oversampling the sample. Don't record the sample at a volume that maxes out the amplitude of the sample. This will give flat topped peaks and create a large amount of static. Record the sample so that it's maximum amplitudes are just under the limits of the data format. The next important factor is that the vibrato, pitch, volume, and general tone character of the sample should remain constant throughout the sample. There should really be no vibrato at all if it can be avoided, since resampling to different frequencies will cause the vibrato to either stretch out into weird sounding pulses or shrink into high frequency twitters as lower or higher notes are played. The same goes for pitch. Any deviations from constant pitch will be distorted the further away from the root key you get. The sample needs to have dead on perfect pitch too, since the more out of tune a sample is, the shorter it's useable range of notes that it can play without the human ear hearing it stretch out of tune. Samples can be retuned by adjusting the sampling rate to raise or lower the pitch slighty, but the closer the sample is to the correct pitch to begin with, the better. Pitch problems can also surface in note attacks, which will often be flat or sharp, then bend to the correct pitch for the sustained note. This has the effect of causing fast notes to be pitched differently than sustained notes, since only the attack is heard on the fast notes before the note ends. Attacks also need to be very rapid. A volume swell at the beginning of a sample can be very bad for both short and low notes. In a short note, the sample will not have enough time to rise to a good volume before the note is killed, and on a low note the volume swell will be stretched out so that it takes a long time for the sample to reach full volume. The low notes will either have a "sluggish" feel, like they are lagging behind the higher notes, or they will be very quiet, since none of them will have had time to reach their normal volume. The constant tone character comes more in to play when looping the sample, but it is always a good idea that the sample sound the same thoughout it's entire length rather than change it's tone. Looping samples can be a difficult thing to do well. Practice makes perfect, and that just comes with experience. However, there are some samples that just can not be looped well, at least not without substantial editing and manipulation to get them to do so, which I have never attempted. A good loop needs to be reasonably long and maintain a constant vibrato, pitch, volume, and general tone character. If any one of these criteria are not met, the loop will wind up sounding like a fast pulsating mess rather than a nice long sustained note. Volume changes, tone character changes, and loops that are too short will all result in a vibrato effect, one which generally sounds bad no matter what frequency the note is played at. The phasing of the loop points must also be aligned, so that you align the peaks and troughs of the waveforms at the beginning of the loop with those at the end. This includes any higher order patterns that are visible when you view the sample at different zoom levels, creating different interference patterns. While misalignment of loop points usually results in a pulsating or growling effect, it could also result in a change in pitch. Making a loop a little too long or too short can have the effect of changing the frequency of the loop region, so even if the loop itself sounds normal, it may be playing at a different effective pitch than the non-looped portion of the sample. It may sound like it can be a little tedious to make a good loop, and it IS if the sample is not a good one, but if the sample meets all the criteria given above in the sampling section, it should be fairly easy to select a section near the end of the sample that will make a good loop. "Now that you know how to create the perfect sample, how do you go about using that sample to create a good instrument?" Real instruments sound different depending on what octave they are being played in. As an example, a trumpet sample of a high C will not sound like a real trumpet when that sample is stretched down two octaves. To overcome this problem, multiple samples are used in the patch, with each sample assigned to a different region of the scale. The closer in pitch the samples are to each other, the more realistic the final patch will sound. Unfortunately, using many samples that are very close to each other can produce a very large patch. A compromise must be reached between size and realism. I have found that it is best to create samples 5 half-steps apart. Each sample is then assigned a region of the scale +/- 2 half-steps from it's root key. If a particular instrument does not lend itself to producing samples every 5th half-step, then it is usually acceptable to create samples 6 or 7 half-steps apart where necessary. If you space them any further apart than this, then there is a good chance that a note played at the upper extreme of one sample's range will have a noticeably different tone character than the next higher note, which is at the lower extreme of it's sample. Different instruments have different tolerances for how far apart their samples can be spaced without creating bad transitions, but 5 half-steps is generally the best distance to aim for. Even if all of the above guidelines are followed, a multi-sample patch can still be ruined by the sampling conditions. While each sample may be a perfect sample by itself, it still may not blend well with any of the other samples in the patch, or it could be full of background hiss from a noisy amp. There are a few simple things you can do to keep these problems from happening. In order for the samples to blend well with each other, they all need have to have the same volume and basic tone character. Even if the person playing the instrument plays each note exactly like the others, they can still sound different if the microphone is in a slightly different place for each sample. The best solution here is to fix the microphone in place and keep the instrument as close to the same position as possible each time a note is sampled. You should also experiment with different microphone positions to determine where the microphone needs to be to get the best sound. All samples should be made in a single sitting, so that as many parameters can be held constant as possible. If the sampling hardware is prone to noise, most of the noise can be eliminated by post processing the sample with a sample editing program. These programs work by selecting a portion of the sample as "noise", then removing that noise from the sample. The longer the section of noise is, the more accurately the noise is modeled. It is good to leave a second or two of background noise at the beginning and end of a sample, so that you will have long regions of "high quality noise" to select from. These regions also act as buffer regions when the noise reduction algorithm is applied, since the ends of the sample can sometimes be distorted by the process. The sample should only be cropped to it's final size after all noise reduction and any other digital signal processing has been completed. Envelopes are used to apply the finishing touches to a patch. The best way to get a feel for how to make a good envelope is simply to load up a wide variety of patches and look at the envelopes that other people have created. The release envelope is probably the most useful of all the envelope parameters. This controls how quickly the sample decays after the note has been turned off. Melodic percussion instruments often have long releases, to better model the reverberations that persist long after the instrument has been hit. The sustain parameter sets the volume at which the sample is sustained, while the decay envelope controls how quickly the sample volume decays to the sustain volume level. The attack envelope controls how quickly the volume ramps up to 100%, before being effected by the decay envelope. There are many more parameters beyond these four that I have mentioned, including chorus (not recommended) and reverb effects (7% is a good amount), but these are the ones that are probably the most useful. With these, you can greatly enhance the way a sample sounds when it is played. However, you can also greatly detract from the sample by choosing unwise envelope settings. I don't know how many times I've seen decay envelopes that ramp the sample down to 0 before it's even reached the end of the sample. While this can be OK for percussive instruments, it usually sounds awful for anything else, like electric guitars, brass, woodwinds, etc.. It's generally a good idea to just add a little pseudo-reverb with the release envelope, and don't add any attack, decay, or sustain parameters. "Now that I know how to make patches, how can I use them?" There are many different "wavetable" soundcards on the market that let you load user created patches. Along with this variety comes a variety of patch formats that each vendor has created, such as the Gravis PAT, AWE (and others) SF2, the up and coming 94B format, and several others too numerous to list here. Most cards come with software (Win32) that lets you create patches in their own custom format. Some even allow you to import patches from other formats. There is a good commercial Win32 product called Awave that will edit and convert between just about any patch format. For those of the Linux persuasion, there is the free Smurf Sound Font Editor. Whatever you use to create/convert the patches, make sure that you have enough RAM on the card to enable you to use them. A high quality patch set might take >= 32 megs of RAM. For those without a wavetable card, without one that works under Linux, or without enough RAM to load large patches, there are a few software solutions. There are some commercial products that will allow Win32 users to load WAV files, maybe even PAT/SF2 files, and render midi as a Win32 MIDI device. While this is a nice thing, it might be better just to get a wavetable card or buy more RAM, since these programs aren't necessarily cheap. They also won't be of ANY use to Linux users. I have heard good things about CSound, a powerful sound/music tool with free ports to just about every operating system, but I've never tried it myself. I use TiMidity++. It's good, fast, has ports for most platforms, is easily customizable, and has free source code to hack on if you want to. Check it out. Whatever you use to play them, I hope this little guide can help you make some really good MIDI instruments. Eric A. Welsh >< Center for Molecular Design Center for Computational Biology Washington University St. Louis, MO

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