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Questions about period and frequency

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Hi! I've recently been working on making a simple mod player, and for the most part everything has been going well. However, translating the period of a note into its frequency has been giving me some clearly incorrect results. I've been using this page as my main reference:

It claims that the formula to find frequency is (7159090.0 / (2 * period)). But for most amiga period values, this produces a very high frequency. For example: C-1 is said to have a period of 856, which implies a frequency of 4181.7 Hz.

Since I'm not very good at reading other's code, I haven't been able to figure it out from other sources, like libopenmpt. My current guess is that my player interprets frequency differently (it's not derived from a mod player). If it helps, my code assumes a 2Hz frequency should try to play any sample (regardless of size) twice a second, which admittedly seems a bit off.

Saga Musix:

--- Quote ---C-1 is said to have a period of 856, which implies a frequency of 4181.7 Hz.
--- End quote ---
That sounds correct. I think it's just your interpretation of what frequency means in this context is a bit off. A frequency of 4181 Hz doesn't mean that the whole sample repeats 4181 times a second. The result of this formula is a sample rate, i.e. it's the frequency at which the sample position is incremented - it means that 4181 sampling points will play per second. So if your sample is 4181 samples long, its duration will be exactly one second (if it was 8362 points long, it would take two seconds to play, and so on).
What OpenMPT does is first translate the period into frequency (the same way you do), and then converts this frequency into a ratio of this sample mix rate and the output mix rate, and uses this to determine how fast to increment the sample. Let's choose more simple numbers for the sake of an example: Suppose your output mix rate is 48000 Hz (that would be a very common value, alternatively "CD quality" 44100 Hz), and period to frequency conversion gave you a sample rate of 4800 Hz (somewhere between D-1 and D#1). The ratio between those two numbers is 4800/48000 = 0.1, i.e. for every increment in the mixer's output, you advance the offset at which you read from your sample data at 0.1.
Small example:

--- Code: ---time  |  offset
0     |  0
1     |  0.1
2     |  0.2
3     |  0.3
9     | 0.9
10    | 1.0
11    | 1.1
after one second...
47999 | 4799.9
48000 | 4800.0
48001 | 4800.1

--- End code ---
Now the next thing you might be asking yourself is "how do I read sample data at offset 0.5", and the answer is "it depends". An old-skool MOD player will simply truncate the result, i.e. round down. A more sophisticated player will interpolate the output if the offset is fractional, for which it will use at least the previous and next sample point (linear interpolation) or even more surrounding sample points (e.g. sinc interpolation). But you shouldn't care about that for now, one step at a time. :)

Ah I was so close, it seems so obvious in retrospect. Thank you very much! Coincidentally, I recently learned about a couple kinds of interpolation in university, so I've already implemented it (linear at least)! On a related note, do you have a recommended way of storing sample data? I tried to use pointers and std::vectors, but had some weird issues. At the moment, I'm using the FastTracker 2 approach, which is just pointers to arrays, but I've heard that's unsafe.

Thank you as always!

Saga Musix:
I'd just use vectors in a modern codebase. Currently OpenMPT uses raw pointers for historical reasons, but this will probably change in the future.
One thing that's slightly more complicated with vectors is handling 8-bit and 16-bit data: You cannot simply cast a vector<int8> to vector<int16>, so you'd either have to cast the vector data each time you access it (which doesn't cost anything performance-wise but it can lead to mistakes if not done properly), or maybe for a more modern approach, use a std::variant<std::vector<int8>, std::vector<int16>> to be able to distinguish between 16-bit and 8-bit data.

Hi again! I have some more questions, unfortunately. My player is working very well actually, but implementing some effects has been a nightmare. In particular stuff like Slide to note (3xy) and Vibrato (4xy). It feels like I have to override the whole player just to accommodate. What do you think is a good way to work with these?


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