Hi everyone,
I just found out that I have sent signals between 0..1 instead of -1..1 to [dac] (!!!), and I am wondering... what happens in that case?
Does Pd take care of that in any way? Or, Is that the job of portaudio/asio/jack? Or, Is that the job of the audio interface? Or, nothing happens and you just send non-negative values to a speaker?
I attach a minimal noise test patch to show this (I only hear a difference in loudness on macos using portaudio and a motu...)
Thanks in advance,
f
I think the DAC/audiobackend/soundcard will accept those values "As is".
The Amp and the Speaker will just work on "one half" of their normal area. (50% of its audio loudness).
It shouldn't hurt the amp or the speaker as long as you don't try to pump up the volume. You might break the speaker if you increase too much the volume. (it will sound soft but you are pushing it too much to one of its extremes).
Mensaje telepatico asistido por maquinas.
On 3/1/2020 8:24 PM, ffdd cchh wrote:
Hi everyone,
I just found out that I have sent signals between 0..1 instead of -1..1 to [dac] (!!!), and I am wondering... what happens in that case?
Does Pd take care of that in any way? Or, Is that the job of portaudio/asio/jack? Or, Is that the job of the audio interface? Or, nothing happens and you just send non-negative values to a speaker?
I attach a minimal noise test patch to show this (I only hear a difference in loudness on macos using portaudio and a motu...)
Thanks in advance,
f
Pd-list@lists.iem.at mailing list UNSUBSCRIBE and account-management -> https://lists.puredata.info/listinfo/pd-list
Hi,
software like Pd or jack doesn't care because it's not forbidden to send only positive or even constant values. You might use your DAC to send control voltages for a modular synthesizer, for example. Most audio interfaces, however, apply some kind of highpass filter to the outputs to get rid of very low frequencies, but there are some "DC coupled" audio interfaces which pass the output unchanged. If you use such an interface with speakers, you have to be careful :-) But then again, you speakers might actually have a highpass filter...
Personally, I like to play it safe and add a [hip~ 5] as a DC filter before my [dac~] :-)
Christof
On 02.03.2020 00:24, ffdd cchh wrote:
Hi everyone,
I just found out that I have sent signals between 0..1 instead of -1..1 to [dac] (!!!), and I am wondering... what happens in that case?
Does Pd take care of that in any way? Or, Is that the job of portaudio/asio/jack? Or, Is that the job of the audio interface? Or, nothing happens and you just send non-negative values to a speaker?
I attach a minimal noise test patch to show this (I only hear a difference in loudness on macos using portaudio and a motu...)
Thanks in advance,
f
Pd-list@lists.iem.at mailing list UNSUBSCRIBE and account-management -> https://lists.puredata.info/listinfo/pd-list
Thanks Christof and Lucas for your replies!
The Amp and the Speaker will just work on "one half" of their normal area. (50% of its audio loudness).
Exactly, that is the difference I hear in loudness.
It shouldn't hurt the amp or the speaker as long as you don't try to pump up the volume. You might break the speaker if you increase too much the volume. (it will sound soft but you are pushing it too much to one of its extremes).
Indeed, this is an important warning that should be in the docs (if it's there other than getting rid of DC offset, I must have missed it)
Personally, I like to play it safe and add a [hip~ 5] as a DC filter before my [dac~] :-)
I do this, too, just by (blindly) following the pd manual and helpfiles... perhaps this calls for a tiny adjustment to the [dac] object to always implement a hipass (with perhaps a flag to revert to a non-hipass-5 enabled [dac]). Just a thought, but one that could be easily argued against given pd's agnostic qualities.
You might use your DAC to send control voltages for a modular synthesizer, for example.
I have never tried this, but I guess it might be an edge case, and quite an interesting one.
So, if I want to get more technical as to what exactly happens to the speakers when sending such "malformed" or "halfformed?" signals, do you know where I can find good sources that would explain this? I guess there's a math foundation here that would justify the need for a -1..1 correct signal range... an age old one that probably relates to the Nyquist theorem... but I can't see how.
In any case, thanks again for the quick replies!
Best,
f
I guess there's a math foundation here that would justify the need for a -1..1 correct signal range...
Is just a convenience.
16bit or 24bit audio files have totally different numbers to represent the start, middle and end of the same sine wave.
Why would you want Pd to have 0 as one side of the sample's limits?
Mensaje telepatico asistido por maquinas.
On 3/2/2020 12:42 AM, Fede Camara Halac wrote:
Thanks Christof and Lucas for your replies!
The Amp and the Speaker will just work on "one half" of their normal area. (50% of its audio loudness).
Exactly, that is the difference I hear in loudness.
It shouldn't hurt the amp or the speaker as long as you don't try to pump up the volume. You might break the speaker if you increase too much the volume. (it will sound soft but you are pushing it too much to one of its extremes).
Indeed, this is an important warning that should be in the docs (if it's there other than getting rid of DC offset, I must have missed it)
Personally, I like to play it safe and add a [hip~ 5] as a DC filter before my [dac~] :-)
I do this, too, just by (blindly) following the pd manual and helpfiles... perhaps this calls for a tiny adjustment to the [dac] object to always implement a hipass (with perhaps a flag to revert to a non-hipass-5 enabled [dac]). Just a thought, but one that could be easily argued against given pd's agnostic qualities.
You might use your DAC to send control voltages for a modular synthesizer, for example.
I have never tried this, but I guess it might be an edge case, and quite an interesting one.
So, if I want to get more technical as to what exactly happens to the speakers when sending such "malformed" or "halfformed?" signals, do you know where I can find good sources that would explain this? I guess there's a math foundation here that would justify the need for a -1..1 correct signal range... an age old one that probably relates to the Nyquist theorem... but I can't see how.
In any case, thanks again for the quick replies!
Best,
f
Pd-list@lists.iem.at mailing list UNSUBSCRIBE and account-management -> https://lists.puredata.info/listinfo/pd-list
Le 02/03/2020 à 04:42, Fede Camara Halac a écrit : ...
You might use your DAC to send control voltages for a modular synthesizer, for example.
I have never tried this, but I guess it might be an edge case, and quite an interesting one.
(un)fortunately, most soundcard are DC filtered. This will not work, unless you have a very specific sound card.
This also mean than adding a hip~ 5 is not mandatory in most situation.
cheers
(un)fortunately, most soundcard are DC filtered. This will not work, unless you have a very specific sound card.
I think I've pointed that out:
Most audio interfaces, however, apply some kind of highpass filter to the outputs to get rid of very low frequencies, but there are some "DC coupled" audio interfaces which pass the output unchanged.
Christof
On 02.03.2020 08:11, cyrille henry wrote:
Le 02/03/2020 à 04:42, Fede Camara Halac a écrit : ...
You might use your DAC to send control voltages for a modular synthesizer, for example.
I have never tried this, but I guess it might be an edge case, and quite an interesting one.
(un)fortunately, most soundcard are DC filtered. This will not work, unless you have a very specific sound card.
This also mean than adding a hip~ 5 is not mandatory in most situation.
cheers
Pd-list@lists.iem.at mailing list UNSUBSCRIBE and account-management -> https://lists.puredata.info/listinfo/pd-list
-
Christof Ressi -
cyrille henry -
Fede Camara Halac -
ffdd cchh
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Lucas Cordiviola