[sdiy] Wave table interpolation

[Mikko Helin]

For PCM1681 and it's 8-channel TDM interface you will need some STM32F42x
part at least. (427)

--Mikko

On Mon, Nov 13, 2017 at 5:32 PM, <paula at synth.net> wrote:

> <cough>
>
> 256 Kbyte of SRAM
>
> http://www.st.com/en/microcontrollers/stm32f412ve.html
>
>
> Still, that's quite a chunk of RAM.
>
> Paula
>
>
>
> On 2017-11-13 15:08, Tom Wiltshire wrote:
>
>> 100MHz?! 256mB RAM?! Easy life!! That’s no limitations at all. <insert
>> Monty Python's Yorkshireman paraphrasing here>
>>
>> Yes, you could certainly make linear interp work well in such a situation.
>>
>> You’ve got multiply instructions and probably multiply-accumulate too,
>> so linear interp is only a couple of instructions and you don’t have
>> to worry about whether it’s powers of 2 or not (although that does
>> always seem to make life easier…)
>>
>> Tom
>>
>> ==================
>>        Electric Druid
>> Synth & Stompbox DIY
>> ==================
>>
>> On 12 Nov 2017, at 22:50, Tim Ressel <timr at circuitabbey.com> wrote:
>>>
>>> I meant clean code-wise  :-)
>>>
>>> Target sample rate is 192KHz. TI makes a super cool 8 channel audio dac
>>> PCM1681. Target proc is the STM32F412VET6, cortex M4, 256 mB ram, 100MHz.
>>>
>>> So you think I can get away with linear?  That would make the interop
>>> code simple. Since I am filling in 256 points in between samples the math
>>> is all powers of 2.
>>>
>>> -t-i-m-b-o-
>>>
>>> On 11/12/2017 11:28 AM, Richie Burnett wrote:
>>>
>>>> Linear interpolation isn't particularly clean (it isn't band-limited)
>>>> but can work adequately if you oversample the data in the wave tables
>>>> first, and/or use a sufficiently high playback sample-rate like 96kHz.  In
>>>> the case of the former, if the data in the wavetables is already
>>>> oversampled considerably, then linear interpolation between the points
>>>> isn't such a bad approximation.  And the latter gives space for the images
>>>> from the imperfect interpolation to go, before they alias back into the
>>>> audio band where you will hear them...
>>>>
>>>> If you run your playback at 48kHz, then the demands on the interpolator
>>>> are very stringent.  It should be flat out to 20kHz or thereabouts, and
>>>> then be something like 80dB or more down by 28kHz if you aren't to hear
>>>> aliasing back into the audio band below 20kHz.  That's only 8kHz of
>>>> transition band for the interpolating filter to work it's magic.  However,
>>>> doubling the playback sample rate to 96kHz, means that you've got all the
>>>> space between 20kHz and 76 kHz for the interpolating filter's transition
>>>> band.  That's 7 times as wide, for a mere x2 oversampling during playback,
>>>> so the demands on the interpolating filter are greatly reduced!
>>>>
>>>> -Richie,
>>>>
>>>> -----Original Message----- From: Tim Ressel
>>>> Sent: Sunday, November 12, 2017 6:56 PM
>>>> To: SYNTH DIY
>>>> Subject: [sdiy] Wave table interpolation
>>>>
>>>> Hi,
>>>>
>>>> Another fun one. I've been looking at the WaveEdit software. It looks
>>>> like it produces waves that are 256 samples by 16 bits. Obviously you'll
>>>> want to interpolate between the samples. Linear interpolation works and
>>>> is cheap but is brute force and causes lots of distortion. I've looked
>>>> at cubic interpolation. Is that a viable approach? My app needs 4 tables
>>>> live at a time so I can interpolate (linearly) between them, so this all
>>>> has to happen on the fly. Cubic looks a bit math intensive.
>>>>
>>>> Thanks!
>>>>
>>>>
>>> --
>>> --Tim Ressel
>>> Circuit Abbey
>>> timr at circuitabbey.com
>>>
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>>
>>
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