| Frodigi 6 - Assisted MP3 by Algotech | ||||||||
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popularity : 49% |
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added on the 2015-08-23 05:33:26 by algorithm  |
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comments
Impressive packing this into 202 blocks.
rulez added on the 2015-08-23 09:25:31 by w00t! 
o_______0 49.9K??? Black hole soon!
I enjoy Algorithm's work. These are always impressive. Congrat!
I wish I could do things like that on Thomson machines. Does this demo uses specific C64/SID tricks or is it mainly a cpu-only effect ?
I wish I could do things like that on Thomson machines. Does this demo uses specific C64/SID tricks or is it mainly a cpu-only effect ?
Been following the Frodigi project from the start. Never getting tired of it, and I hope to hear even more in the future.
The previous versions relied on the SID chip entirely to produce the audio. This version mixes and produces the audio entirely in software hence can be used on any other machine provided that it can output 8 bit samples (although can be adjusted to output at lower bits, but that would drastically destroy the audio even more)
The software approach gives more flexible control over the individual sines. One of the main issues with using the hardware oscillators of the sid was that the sustain
was not fully controllable resulting in heavy clicking or having to adjust the encoder to only force higher sustain if required.
The result of this was that amplitudes were not as intended making some channels louder/quieter than others enhancing the warbling effect.
The same issue arises also if attempting to use adsr curves to approximate the sustain or fixed sustain per channel and changing the master volume.
Mixing a small amount of sines is also very much sensitive to the source audio. In nearly all complex audio cases there will be issues with warbling and the underwater syndrome when using this very small amount of sines to recreate the audio.
I still think it does a good job considering the low amount of sines. I have compared it to some specific vst based plugins (e.g pitchtech) that can convert to given amount of sines and this performs rather poorly in comparison although i am not sure if the "melody extractor" in the plugin performs the psychoaccoustic masking.
The frodigi method is really only aimed at extremely low bitrate audio. quality can be increased by more sine channels or more updates, but this would make it far less useful when
other more efficient and higher quality methods (at higher bitrates) can be used instead .
For speech only, there is less chance of artifacts. I have provided a link to an example below
https://www.dropbox.com/s/pkukeme7ex8h73x/fro6test.prg?dl=1
Adding drums, bassline, chords, synth together with synth makes it choke however (ideally would need minimum 16 sines) - working well with the amiga based decoder, but would be too much for the c64 to mix these channels at 8khz.
There are some improvements that can be made. In particular the mixing. would have probably been able to mix 6 sines in non badline mode (using this current full decode in nmi implementation)
Fastest method of mixing was to prescale the sines from 0-63 unsigned (using 16 amplitudes) and then merely to add these together to recreate the 8bit unsigned.
this does result in the lower bits being distorted using the 8bit playback method (as its more 6.5bits or so scaled out to 8bit) - the digimax mode in the demo will give higher playback quality
the low order byte is added using a 8 step pitch table which gives me the steps to match the fft encode (5hz increments) this ensures seamless looping, however amplitude changes can result in clicking (which would have been more easily rectified if i had used the nmi to play back a buffer and used the main loop to decode and interpolate between transitions.
Further experiments will also involve multiple waveforms such as noise, pulse etc. due to the software approach, the phase can also be used if required.
The software approach gives more flexible control over the individual sines. One of the main issues with using the hardware oscillators of the sid was that the sustain
was not fully controllable resulting in heavy clicking or having to adjust the encoder to only force higher sustain if required.
The result of this was that amplitudes were not as intended making some channels louder/quieter than others enhancing the warbling effect.
The same issue arises also if attempting to use adsr curves to approximate the sustain or fixed sustain per channel and changing the master volume.
Mixing a small amount of sines is also very much sensitive to the source audio. In nearly all complex audio cases there will be issues with warbling and the underwater syndrome when using this very small amount of sines to recreate the audio.
I still think it does a good job considering the low amount of sines. I have compared it to some specific vst based plugins (e.g pitchtech) that can convert to given amount of sines and this performs rather poorly in comparison although i am not sure if the "melody extractor" in the plugin performs the psychoaccoustic masking.
The frodigi method is really only aimed at extremely low bitrate audio. quality can be increased by more sine channels or more updates, but this would make it far less useful when
other more efficient and higher quality methods (at higher bitrates) can be used instead .
For speech only, there is less chance of artifacts. I have provided a link to an example below
https://www.dropbox.com/s/pkukeme7ex8h73x/fro6test.prg?dl=1
Adding drums, bassline, chords, synth together with synth makes it choke however (ideally would need minimum 16 sines) - working well with the amiga based decoder, but would be too much for the c64 to mix these channels at 8khz.
There are some improvements that can be made. In particular the mixing. would have probably been able to mix 6 sines in non badline mode (using this current full decode in nmi implementation)
Fastest method of mixing was to prescale the sines from 0-63 unsigned (using 16 amplitudes) and then merely to add these together to recreate the 8bit unsigned.
this does result in the lower bits being distorted using the 8bit playback method (as its more 6.5bits or so scaled out to 8bit) - the digimax mode in the demo will give higher playback quality
the low order byte is added using a 8 step pitch table which gives me the steps to match the fft encode (5hz increments) this ensures seamless looping, however amplitude changes can result in clicking (which would have been more easily rectified if i had used the nmi to play back a buffer and used the main loop to decode and interpolate between transitions.
Further experiments will also involve multiple waveforms such as noise, pulse etc. due to the software approach, the phase can also be used if required.
youtube link here.. https://www.youtube.com/watch?v=v5Mp58lj2bE&feature=youtu.be
Thanks for the explantations :D (I like technical stuff)
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