Changes
Created page with 'Voice Generator The voice generator relies on the Amplitude, Pitch, F0..F5, and B0..B5 registers, which are processed like so: Amplitude --> F0 --> F1 --> F2 --> F3 --> F4 -…'
Voice Generator
The voice generator relies on the Amplitude, Pitch, F0..F5, and B0..B5 registers, which are processed like so:
Amplitude --> F0 --> F1 --> F2 --> F3 --> F4 --> F5 --> PWM --> External
Pitch/Noise B0 B1 B2 B3 B4 B5 5kHz Filter
Another important register is the Repeat counter, which indicates when the next opcode shall be executed (and which may then load new values into the above registers).
Sample Rate and Repeat Timings
The SP0256 is (usually) driven by a 3.12MHz oscillator, and it uses 7bit PWM output. So the sample rate should be 3.12MHz/128, probably further divided by two:
Sample Rate = 3.12MHz/256 = 12.1875kHz ;82.051ns per sample
Which means one sample is 82.051ns long, that value multiplied by 64 or 91 gives the following timings per repeat:
5.251ms per repeat (noise and pause), or
7.466ms per repeat (tone with pitch=91)
That (guessed/undocumented) samplerate does more or less match the specifications that say that (unwanted PWM-) noise is above 10kHz, and that wanted pitch can be up to 5kHz. And the repeat timings do more or less match the timings shown in the allophone list.
Amplitude/Pitch/Repeat
The 8bit amplitude register defines the volume in floating point form,
Amplitude = lower5bit SHL upper3bit
The pitch defines the frequency, counted in numbers of samples per period.
For pitch=91, one HIGH sample (amplitude) is output, followed by 90 zero samples (null). That pattern is repeated as many times as specified in the repeat count, for example, with repeat=3:
__ Amplitude level (+)
| | |
|________|________|________ __ Zero level PITCH
<-Pitch-> __ Amplitude level (-)
<--------repeat=3--------->
As shown above, the generated waveform is NOT a square wave (which would have 50% high, and 50% low). After applying filters, the final waveform may look somewhat like so:
__ Amplitude level (+)
| | |
|_|_.____|_|_.____|_|_.____ __ Zero level PITCH+FILTERS
| | | | | |
| | | __ Amplitude level (-)
Note that (aside from noise) the AL2 ROM uses only one pitch value: 5Bh aka 91 decimal (meaning that all vowels are using the same base frequency, and they differ only by using different filter settings).
Amplitude/Noise/Repeat
Noise is activated when setting pitch=0. The timings are then same as when pitch=64, but instead of outputting HIGH and NULL levels, the hardware does now randomly output HIGH or LOW levels, for example, pitch=0 and repeat=5:
__ Amplitude level (+)
||| || | | | || | || ||| |
|||_|| |__|_|__||_|___|| |||_| __ Zero level NOISE
| | || | || | ||| | |
<-64->| || | || | ||| | | __ Amplitude level (-)
<----------repeat=5---------->
The exact random algorithm is unknown (probably some shift/xor stuff?), the random levels seem to be output on each sample (not only on the first sample of a repeat). Like normal pitch, the noise is passed to the 6 filters.
Pause/Repeat
The pause command sets amplitude=0. The timings are then same as when pitch=64, but the output is always NULL, for example, pause and repeat=5:
__ Amplitude level (+)
______________________________ __ Zero level PAUSE (SILENCE)
<-64-> __ Amplitude level (-)
<----------repeat=5---------->
Pause does reset the filters to 0, so the silence is not affected by filters.
Digital Filters
As shown above, the amplitude/pitch/noise output is passed through six digital filter stages (using the F0..F5 and B0..B5 registers), each stage looks like so:
_____ _____
------------------>| |------------------->| |-----+----->
_____ | SUB | ______ | SUB | |
+--->| *B |--->|_____| +--->| *2*F |-->|_____| |
| |_____| _____ | |______| _____ |
+---------------|OLDER|<---+---------------| OLD |<----+
|_____| |_____|
Ie. the incoming samples are adjusted like so:
for i=0 to 5 ;filter number
sample = sample - quant_table[F.i] * OLD.i * 2 ;F0..F5 registers
sample = sample - quant_table[B.i] * OLDER.i ;B0..B5 registers
OLDER.i = OLD.i
OLD.i = sample
next i
Whereas, quant_table is a non-linear translation table that translates the signed 8bit registers to signed 10bit factors (with 9bit fractional part, ie. 511 means 0.99), with following entries:
0 ,9 ,17 ,25 ,33 ,41 ,49 ,57 ,65 ,73 ,81 ,89 ,97 ,105,113,121
129,137,145,153,161,169,177,185,193,201,209,217,225,233,241,249
257,265,273,281,289,297,301,305,309,313,317,321,325,329,333,337
341,345,349,353,357,361,365,369,373,377,381,385,389,393,397,401
405,409,413,417,421,425,427,429,431,433,435,437,439,441,443,445
447,449,451,453,455,457,459,461,463,465,467,469,471,473,475,477
479,481,482,483,484,485,486,487,488,489,490,491,492,493,494,495
496,497,498,499,500,501,502,503,504,505,506,507,508,509,510,511
Above shows only positive values for index 0..127. Values for index -1..-128 should be 0..-511, or maybe -9..-512.
The voice generator relies on the Amplitude, Pitch, F0..F5, and B0..B5 registers, which are processed like so:
Amplitude --> F0 --> F1 --> F2 --> F3 --> F4 --> F5 --> PWM --> External
Pitch/Noise B0 B1 B2 B3 B4 B5 5kHz Filter
Another important register is the Repeat counter, which indicates when the next opcode shall be executed (and which may then load new values into the above registers).
Sample Rate and Repeat Timings
The SP0256 is (usually) driven by a 3.12MHz oscillator, and it uses 7bit PWM output. So the sample rate should be 3.12MHz/128, probably further divided by two:
Sample Rate = 3.12MHz/256 = 12.1875kHz ;82.051ns per sample
Which means one sample is 82.051ns long, that value multiplied by 64 or 91 gives the following timings per repeat:
5.251ms per repeat (noise and pause), or
7.466ms per repeat (tone with pitch=91)
That (guessed/undocumented) samplerate does more or less match the specifications that say that (unwanted PWM-) noise is above 10kHz, and that wanted pitch can be up to 5kHz. And the repeat timings do more or less match the timings shown in the allophone list.
Amplitude/Pitch/Repeat
The 8bit amplitude register defines the volume in floating point form,
Amplitude = lower5bit SHL upper3bit
The pitch defines the frequency, counted in numbers of samples per period.
For pitch=91, one HIGH sample (amplitude) is output, followed by 90 zero samples (null). That pattern is repeated as many times as specified in the repeat count, for example, with repeat=3:
__ Amplitude level (+)
| | |
|________|________|________ __ Zero level PITCH
<-Pitch-> __ Amplitude level (-)
<--------repeat=3--------->
As shown above, the generated waveform is NOT a square wave (which would have 50% high, and 50% low). After applying filters, the final waveform may look somewhat like so:
__ Amplitude level (+)
| | |
|_|_.____|_|_.____|_|_.____ __ Zero level PITCH+FILTERS
| | | | | |
| | | __ Amplitude level (-)
Note that (aside from noise) the AL2 ROM uses only one pitch value: 5Bh aka 91 decimal (meaning that all vowels are using the same base frequency, and they differ only by using different filter settings).
Amplitude/Noise/Repeat
Noise is activated when setting pitch=0. The timings are then same as when pitch=64, but instead of outputting HIGH and NULL levels, the hardware does now randomly output HIGH or LOW levels, for example, pitch=0 and repeat=5:
__ Amplitude level (+)
||| || | | | || | || ||| |
|||_|| |__|_|__||_|___|| |||_| __ Zero level NOISE
| | || | || | ||| | |
<-64->| || | || | ||| | | __ Amplitude level (-)
<----------repeat=5---------->
The exact random algorithm is unknown (probably some shift/xor stuff?), the random levels seem to be output on each sample (not only on the first sample of a repeat). Like normal pitch, the noise is passed to the 6 filters.
Pause/Repeat
The pause command sets amplitude=0. The timings are then same as when pitch=64, but the output is always NULL, for example, pause and repeat=5:
__ Amplitude level (+)
______________________________ __ Zero level PAUSE (SILENCE)
<-64-> __ Amplitude level (-)
<----------repeat=5---------->
Pause does reset the filters to 0, so the silence is not affected by filters.
Digital Filters
As shown above, the amplitude/pitch/noise output is passed through six digital filter stages (using the F0..F5 and B0..B5 registers), each stage looks like so:
_____ _____
------------------>| |------------------->| |-----+----->
_____ | SUB | ______ | SUB | |
+--->| *B |--->|_____| +--->| *2*F |-->|_____| |
| |_____| _____ | |______| _____ |
+---------------|OLDER|<---+---------------| OLD |<----+
|_____| |_____|
Ie. the incoming samples are adjusted like so:
for i=0 to 5 ;filter number
sample = sample - quant_table[F.i] * OLD.i * 2 ;F0..F5 registers
sample = sample - quant_table[B.i] * OLDER.i ;B0..B5 registers
OLDER.i = OLD.i
OLD.i = sample
next i
Whereas, quant_table is a non-linear translation table that translates the signed 8bit registers to signed 10bit factors (with 9bit fractional part, ie. 511 means 0.99), with following entries:
0 ,9 ,17 ,25 ,33 ,41 ,49 ,57 ,65 ,73 ,81 ,89 ,97 ,105,113,121
129,137,145,153,161,169,177,185,193,201,209,217,225,233,241,249
257,265,273,281,289,297,301,305,309,313,317,321,325,329,333,337
341,345,349,353,357,361,365,369,373,377,381,385,389,393,397,401
405,409,413,417,421,425,427,429,431,433,435,437,439,441,443,445
447,449,451,453,455,457,459,461,463,465,467,469,471,473,475,477
479,481,482,483,484,485,486,487,488,489,490,491,492,493,494,495
496,497,498,499,500,501,502,503,504,505,506,507,508,509,510,511
Above shows only positive values for index 0..127. Values for index -1..-128 should be 0..-511, or maybe -9..-512.
