is there such a thing as down sampling or up sampling the [outlet~] or [dac~]? i have a bunch of .wav file which is in 11025hz & i dont have a batch file converter to up sample it to 44100hz.
so basically, the files playsback at a chipmunk speed as i think my computer soundcard doesn't accept lower sampling rate for .wav. i tried to lowered the sampling rate at the audio setting but no audio playback. -j
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Down sampling?
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something like [block~ 64 1 0.25] might work
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hmmm.., sounds like it playback at the right speed now but I can hear some artifacts. perhaps i need to down sample on each individual loaded sample file & not the overall output. btw, could you explain what's [block~ 64 1 .25]. i tried the browser help but don't really get. thanks for the reply hardoff
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[block~ 64 1 0.25]
the '64' is the amount of samples pd will process in one 'block'. that is the default blocksize...you could set it to any power of 2 if you wanted to. i'm not too sure, but i think that by processing a block of samples at a time, rather than processing sample by sample, pd can cut cpu usage.
the '1' is the oversampling (overlap) amount. if it's just 1, then samples will be processed one block at a time, but if you set it to 2 or 4, or whatever, then the sound will be processed 2 or 4 times in parallel, with an offset. sorry,,,,it's hard for me to explain how that works.
but importantly here, the '0.25' is the downsampling rate. because pd defaults to 44100hz sampling rate, your 11025hz samples will be played 4 times too fast, therefore you need to downsample to get the correct rate.
but what object are you using to playback your sounds? [tabread4~] / [tabplay~ ] / [readsf~ ]
i think there are ways for all of these to playback samples at the correct rate without changing [block~] settings.
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thanks for the elaborative block~ explanation.
i'm playing the sample file using [tabplay~] -
you might want to check the help files on [switch~] and [samphold~].
any technology distinguishable from magic
is insufficiently advanced. -
mmhh about the artefact
i have read you have to filter the sound (before/after ?) resampling it.
but i don't know whereHK
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thanks for the replies...
im actually just trying a patch i downloaded & the wav files sampling rate is 11025hz. ill probably convert it 1st to 44100 then re-uploaded it to pd. it might be expensive to do everything in pd...