dfpwm

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+ | DFPWM (Dynamic Filter Pulse Width Modulation) is an audio codec created by Ben "GreaseMonkey" Russell in 2012, originally to be used as a voice codec for asiekierka's pixmess, a C remake of 64pixels. It is a 1-bit-per-sample codec which uses a dynamic-strength one-pole low-pass filter as a predictor. Due to the fact that a raw DPFWM decoding creates a high-pitched whine, it is often followed by some post-processing filters to make the stream more listenable. | ||

+ | |||

+ | DFPWM is recognisable, but also creates a fair bit of noise. It is suitable for brostep, however. | ||

+ | |||

+ | It depends on the implementation as to what all the parameters are. Testing has shown that 8 bits per sample works better than 16 bits. The codec is not frequency-dependent, and can work at any frequency. It is a mono codec, but stereo can be achieved by running two streams in parallel. | ||

+ | |||

+ | If necessary, you can get away with converting your streams to raw 1-bit, or you could possibly assume a static-strength low-pass filter. | ||

+ | |||

+ | The implementation in Computronics handles streams using Ri=7, Rd=20, 8 bits per sample, and LSB stored first. | ||

+ | |||

+ | ==== Specification ==== | ||

+ | |||

+ | DFPWM, at its simplest, works like this: | ||

+ | Decoding: | ||

+ | smp(t) <- Predictor(bit(t)) | ||

+ | Encoding: | ||

+ | if smp(t) > LastPredictor OR (smp(t) = LastPredictor = 127): | ||

+ | bit(t) <- Predictor(1) | ||

+ | else: | ||

+ | bit(t) <- Predictor(0) | ||

+ | |||

+ | where bit is either LOW or HIGH, and smp ∈ [LOW, HIGH]. (Apologies for using the ∈, it's just that if I try to do "less than or equal", dokuwiki gives me an "is implied by" symbol instead. --GM) | ||

+ | |||

+ | === State === | ||

+ | |||

+ | Firstly we need to give some types: | ||

+ | * Let q,s be signed integers. | ||

+ | * Let b' be a single bit, either 0 or 1. | ||

+ | * Let Ri,Rd be chosen constant signed integers. ((7,20) is reasonable for 8 bits per sample.) | ||

+ | |||

+ | Then we need to assign meanings: | ||

+ | * Let q be the "charge", initialised to 0. | ||

+ | * Let s be the "strength", initialised to 1. | ||

+ | * Let Ri be the strength increase. | ||

+ | * Let Rd be the strength decrease. | ||

+ | |||

+ | Ri and Rd are constant (and, until someone discovers a better set of values, will always be 7 and 20 respectively). q and s vary. | ||

+ | |||

+ | From here we can define the predictor. | ||

+ | |||

+ | === Predictor === | ||

+ | |||

+ | To simplify this, we will define this in terms of signed 8-bit samples. | ||

+ | |||

+ | == Input comprehension == | ||

+ | Let b' be the previous instance of b, initialised to 0 to simplify implementation. | ||

+ | |||

+ | Let t be the "target", -128 if b is 0, and 127 if b is 1. (In other words, LOW and HIGH respectively.) | ||

+ | |||

+ | == Charge adjustment == | ||

+ | Let q' be an integer such that | ||

+ | > q' <- q + (s*(t - q) + 128)/256 | ||

+ | |||

+ | If q == q', and q != t, then: | ||

+ | > If t < q: q' <- q' - 1 | ||

+ | > If t > q: q' <- q' + 1 | ||

+ | |||

+ | This is done to ensure that the "charge" can always reach its target. | ||

+ | |||

+ | Then set q <- q'. | ||

+ | |||

+ | == Strength adjustment == | ||

+ | Let r,z be integers such that: | ||

+ | > If b == b', then r = Ri, z = 255 | ||

+ | > If b != b', then r = Rd, z = 0 | ||

+ | |||

+ | Let s' be an integer such that | ||

+ | > s' <- s + (r*(z - s) + 128)/256 | ||

+ | |||

+ | If s == s', and s != z, then: | ||

+ | > If z < s: s' <- s' - 1 | ||

+ | > If z > s: s' <- s' + 1 | ||

+ | |||

+ | Then set s <- s'. | ||

+ | |||

+ | === Filtering === | ||

+ | |||

+ | You can do anything here, within reason. These methods are by no means the best way to deal with the noise you get. | ||

+ | |||

+ | These notes are based on the C implementation. | ||

+ | |||

+ | == Antijerk == | ||

+ | |||

+ | If the current target and the previous target are different, output the average of the current and previous results from the predictor; otherwise, output the value directly. | ||

+ | |||

+ | == Low-pass filter == | ||

+ | |||

+ | outQ <- outQ + (expectedOutput - outQ) * 100/256, essentially. | ||

dfpwm.txt · Last modified: 2015/06/25 11:36 (external edit)