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Filter Coefficients in ATRAC?
pat
post Dec 8 2003, 09:39
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Since there is a standard filter coefficients given in mp3(ISO 11172-3), I wonder if there is a standard filter coefficients I can find for ATRAC?

If there is no, how do i go about deciding the coefficients i'm going to use to build the ATRAC encoder?
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Gabriel
post Dec 8 2003, 10:13
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Do you have the specifications of the Atrac format?
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pat
post Dec 8 2003, 10:19
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No. I don't have something as detailed like the ISO 11172. So far I only have the overall block diagrams, without the detailed coefficients.

I only know the QMF have to divide the input signal into 3 subbands, 0 - 5.5kHz, 5.5 - 11kHz, and 11 - 22kHz.
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Gabriel
post Dec 8 2003, 10:42
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So how could you be able to write an encoder without specifications or reference source code?
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pat
post Dec 11 2003, 04:06
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Hm.....that's my problem. I'm actually doing this as my project for part of my coursework.....

Is there anyone who is familiar with ATRAC? I sincerely need help.....
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Gabriel
post Dec 11 2003, 10:12
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The problem is that we do not have access to Atrac specifications (perhaps you have them), so most of us are unable to help you on this matter.
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wkwai
post Dec 11 2003, 15:19
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Sony guards its commercial secrets very well.... No doubt that without the proper specs, you can't implement an ATRAC coder/decoder.. Even the huffman tables are unique.. even if it does not mention any specific quantization algorithm..
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menno
post Dec 11 2003, 15:30
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Filters are not really the problem. If you know the subband edges you can easily design them yourself smile.gif
Try searching for a book/paper on subband filter design.

BTW: I think ATRAC does 2 filterbanks. One that splits the signal in 2 subbands (0-11,11-22) and one that splits the first subband in 2 more (0-5.5,5.5-11). You can do the second filterbank by decimating the prototype filter of the first by 2. This leaves you to design just a prototype filter for the first filterbank that has only 2 subbands (so splitting signal in halve). This is the easiest filterbank possible smile.gif

Menno
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pat
post Dec 12 2003, 18:47
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Thanks guys. Think I still manage to handle the filter. But then when come to the Bit Allocation and the Spectral Quantization parts, i become more n more headache. As what Gabriel said, Sony guides their technology tightly that I couldn't find any of the standards carry on with my work. So i am kinda stuck now.

Does anyone can help me out????
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wkwai
post Dec 14 2003, 09:34
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QUOTE (menno @ Dec 11 2003, 06:30 AM)
Filters are not really the problem. If you know the subband edges you can easily design them yourself smile.gif
Try searching for a book/paper on subband filter design.


I think it is not as simple as that.. The reconstruction subband filterbanks MUST match the analysis subband filterbanks, otherwise, there could be aliasing artifacts in the decoded audio data.. ohmy.gif
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