Re: [PD] headroom in Pd
I know. My response was tongue in cheek as I knew this ahead of time but, so far, it hasn't been a problem with the music that I make ... On Jan 1, 2014, at 2:08 AM, pd-list-requ...@iem.at wrote: From: Simon Wise simonzw...@gmail.com Subject: Re: [PD] headroom in Pd Date: January 1, 2014 at 12:50:55 AM GMT+1 To: pd-list@iem.at On 31/12/13 08:30, Dan Wilcox wrote: Ouch. I guess alot of us don't have serious projects :D (Out of curiosity, does Max do soft clipping also?) the point was that OSX was messing with the sound between the software, presumably any software, and the audio output ... which may perhaps be called a feature while listening to songs in iTunes but is a big worry if you are trying to use the system seriously as a musician. It is a problem that comes up on this list from time to time, I don't recall any reply saying it could be bypassed or turned off. Simon Dan Wilcox @danomatika danomatika.com robotcowboy.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2013-12-31 19:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! A DAC can only go to 0dBFS by definition. If it appears to go beyond that then something is scaling the input to be less than full scale at full scale. For instance a 24-bit DAC could be sent 16 16-bit full-scale streams and not clip. Only if 16-bits is considered full scale does that make it +12dBFS. Martin ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
Nope, the DAC can freely construct intersample peaks as it sees fit and those can easily exceed 0 dBFS. It has been common practice in the industry for more than a decade to reconstruct clipped samples well above 0 dBFS - partially to make up for shitty mixing and mastering prevalent in music, and also because it's the right way to do it. 16 bits full scale and 24 bits full scale are the same 0dBFS signal. The bits are added at the bottom not the top. On Wed, Jan 1, 2014 at 1:34 PM, Martin Peach martin.pe...@sympatico.cawrote: On 2013-12-31 19:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! A DAC can only go to 0dBFS by definition. If it appears to go beyond that then something is scaling the input to be less than full scale at full scale. For instance a 24-bit DAC could be sent 16 16-bit full-scale streams and not clip. Only if 16-bits is considered full scale does that make it +12dBFS. Martin ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/ listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2014-01-01 19:50, Chris Clepper wrote: Nope, the DAC can freely construct intersample peaks as it sees fit and those can easily exceed 0 dBFS. It has been common practice in the industry for more than a decade to reconstruct clipped samples well above 0 dBFS - partially to make up for shitty mixing and mastering prevalent in music, and also because it's the right way to do it. +1 nevertheless you cannot send digital values to the DAC that exceed 0dBFS. the intersample peaks are *purely* analog. FS stands for full scale and refers to the full range of the digianl fix-point values. thus - by definition - 0dBFS only refers to digital values, and can never be exceeded. however on the analog side, the nominal 0dB can easily be exceeded in the reconstruction. btw, zexy's [limiter~] tries to take intersample peaks into account by upsampling the signal prior to limiting,... so if you use it to limit between -1..+1, then the reconstructed analog signal should not exceed the nominal analog output range. (for practical reasons, upsampling is limited, so in some borderline cases you could still construct a signal that exceeds the 0dB analog) gfmadsr IOhannes signature.asc Description: OpenPGP digital signature ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
I can see how a filter circuit following a DAC can swing more than the DAC for example if two successive samples are full-scale, but there's no way a DAC can output beyond its own full scale except momentarily while it's settling to a value inside its range. The scaling has to be done before the DAC. You just can't reconstruct a clipped signal unless the clipping is very mild or the signal is very simple, like a sine wave. What if the signal is +12dBFS white noise? I meant that if you take 16 bits to be full-scale but you have a 24-bit DAC you _could_ use the 16 LSBs of the DAC as full scale, then you have a lot of headroom but your signal to noise ratio is not as good, and maybe something like this is happening in the default MacOS headphone driver. Martin On 2014-01-01 13:50, Chris Clepper wrote: Nope, the DAC can freely construct intersample peaks as it sees fit and those can easily exceed 0 dBFS. It has been common practice in the industry for more than a decade to reconstruct clipped samples well above 0 dBFS - partially to make up for shitty mixing and mastering prevalent in music, and also because it's the right way to do it. 16 bits full scale and 24 bits full scale are the same 0dBFS signal. The bits are added at the bottom not the top. On Wed, Jan 1, 2014 at 1:34 PM, Martin Peach martin.pe...@sympatico.ca mailto:martin.pe...@sympatico.ca wrote: On 2013-12-31 19:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! A DAC can only go to 0dBFS by definition. If it appears to go beyond that then something is scaling the input to be less than full scale at full scale. For instance a 24-bit DAC could be sent 16 16-bit full-scale streams and not clip. Only if 16-bits is considered full scale does that make it +12dBFS. Martin _ Pd-list@iem.at mailto:Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/__listinfo/pd-list http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
Yes, of course the signal out of the DAC is purely analog. The signal is referenced to 0dBFS on the digital side and also something like dBu or dBv on the analog side (although it varies from part to part). I should have been clearer in stating this. :) But the main point is that there are signals beyond all 1s in the data sent to the DAC if clipping exists. This is confusing to many because the digital math as commonly taught typically only refers to 'inside the box'. I leave this exception out of any workshop or uni course because of the confusion, but it should be known more widely. Sounds like the zexy [limiter~] is better than many commercial plugins that don't upsample! As we know, the hard clip can create some very nasty aliasing artifacts and when done in the box there is no way to undo that damage. Best practice is still to keep the peaks away from full scale even if using a very good DAC that can del with clipping well. On Wed, Jan 1, 2014 at 2:03 PM, IOhannes m zmölnig zmoel...@iem.at wrote: On 2014-01-01 19:50, Chris Clepper wrote: Nope, the DAC can freely construct intersample peaks as it sees fit and those can easily exceed 0 dBFS. It has been common practice in the industry for more than a decade to reconstruct clipped samples well above 0 dBFS - partially to make up for shitty mixing and mastering prevalent in music, and also because it's the right way to do it. +1 nevertheless you cannot send digital values to the DAC that exceed 0dBFS. the intersample peaks are *purely* analog. FS stands for full scale and refers to the full range of the digianl fix-point values. thus - by definition - 0dBFS only refers to digital values, and can never be exceeded. however on the analog side, the nominal 0dB can easily be exceeded in the reconstruction. btw, zexy's [limiter~] tries to take intersample peaks into account by upsampling the signal prior to limiting,... so if you use it to limit between -1..+1, then the reconstructed analog signal should not exceed the nominal analog output range. (for practical reasons, upsampling is limited, so in some borderline cases you could still construct a signal that exceeds the 0dB analog) gfmadsr IOhannes ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On Wed, Jan 1, 2014 at 2:03 PM, Martin Peach martin.pe...@sympatico.cawrote: I can see how a filter circuit following a DAC can swing more than the DAC for example if two successive samples are full-scale, but there's no way a DAC can output beyond its own full scale except momentarily while it's settling to a value inside its range. The scaling has to be done before the DAC. You just can't reconstruct a clipped signal unless the clipping is very mild or the signal is very simple, like a sine wave. What if the signal is +12dBFS white noise? The DAC can't really discriminate based on the aural complexity of a signal. A peak with N successive clipped samples has a reconstructed value above full scale and the DAC will attempt to produce that signal. Since all of the info above full scale is lost, the reconstruction is just a guess whether the input is a sine or kick drum or random noise. The sine wave has a pretty clearly correct reconstruction while a percussive sound might not. A common side effect of doing extreme declipping using something like iZotope in the box is a softening of transients. Not surprising. I haven't looked too deeply into how various DACs go about their process, but some of the higher end parts have pretty decent DSP on board or even allow external SHARC chips to do the work. I do know people that make high end mastering converters who know quite a lot about this. There are limits - The Stooges 'Raw Power' CD is so clipped that even high end DACs with lots of DSP have no hope there! I meant that if you take 16 bits to be full-scale but you have a 24-bit DAC you _could_ use the 16 LSBs of the DAC as full scale, then you have a lot of headroom but your signal to noise ratio is not as good, and maybe something like this is happening in the default MacOS headphone driver. It's not clear what is happening in the Apple codec driver, but the Core Audio doc linked earlier gives a pretty good idea that they are doing a 'soft clipping' algorithm perhaps much like zexy's [limiter~]. I have seen some Wolfson codec datasheets that mention doing the soft clip in the hardware and those parts were aimed at mobile phones and the like. The intent is to prevent hard clipping from reaching the puny internal speakers in such devices and not an aesthetic choice. Martin On 2014-01-01 13:50, Chris Clepper wrote: Nope, the DAC can freely construct intersample peaks as it sees fit and those can easily exceed 0 dBFS. It has been common practice in the industry for more than a decade to reconstruct clipped samples well above 0 dBFS - partially to make up for shitty mixing and mastering prevalent in music, and also because it's the right way to do it. 16 bits full scale and 24 bits full scale are the same 0dBFS signal. The bits are added at the bottom not the top. On Wed, Jan 1, 2014 at 1:34 PM, Martin Peach martin.pe...@sympatico.ca mailto:martin.pe...@sympatico.ca wrote: On 2013-12-31 19:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! A DAC can only go to 0dBFS by definition. If it appears to go beyond that then something is scaling the input to be less than full scale at full scale. For instance a 24-bit DAC could be sent 16 16-bit full-scale streams and not clip. Only if 16-bits is considered full scale does that make it +12dBFS. Martin _ Pd-list@iem.at mailto:Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/__listinfo/pd-list http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/ listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 31/12/13 08:30, Dan Wilcox wrote: Ouch. I guess alot of us don't have serious projects :D (Out of curiosity, does Max do soft clipping also?) the point was that OSX was messing with the sound between the software, presumably any software, and the audio output ... which may perhaps be called a feature while listening to songs in iTunes but is a big worry if you are trying to use the system seriously as a musician. It is a problem that comes up on this list from time to time, I don't recall any reply saying it could be bypassed or turned off. Simon ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! On Tue, Dec 31, 2013 at 6:50 PM, Simon Wise simonzw...@gmail.com wrote: On 31/12/13 08:30, Dan Wilcox wrote: Ouch. I guess alot of us don't have serious projects :D (Out of curiosity, does Max do soft clipping also?) the point was that OSX was messing with the sound between the software, presumably any software, and the audio output ... which may perhaps be called a feature while listening to songs in iTunes but is a big worry if you are trying to use the system seriously as a musician. It is a problem that comes up on this list from time to time, I don't recall any reply saying it could be bypassed or turned off. Simon ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/ listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 01/01/14 11:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! So it is the driver for the built-in audio output that is adding the feature/problem? I rarely use OSX these days, just an old setup with a G4 mac mini and a MOTU that hasn't been updated for years but still does its job nicely. Simon ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
It's not even clear if there is some sort of soft clipping at play in Alexandre's case, but some of the Apple hardware has used such either in the CoreAudio driver or in the hardware codec itself. As I recall the main reasoning was to prevent hard clipping from damaging the tiny laptop speakers and some models it was bypassed when headphones are plugged in. Personally, I test all mixes and masters in iTunes using the headphone out of a Mac Mini. If it sounds bad there I don't blame the driver though. On Tue, Dec 31, 2013 at 8:00 PM, Simon Wise simonzw...@gmail.com wrote: On 01/01/14 11:32, Chris Clepper wrote: It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! So it is the driver for the built-in audio output that is adding the feature/problem? I rarely use OSX these days, just an old setup with a G4 mac mini and a MOTU that hasn't been updated for years but still does its job nicely. Simon ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
I have rem's multiface ii by the way, it says it handles a headrom of 13 db or something, don't really know what it means. It'llbe a while 'til I check anyway, I'm at a very nice beach in Brasil andI just dipped myself into the atlantic ocean LoveTo You All!!! 2013/12/31 Chris Clepper cgclep...@gmail.com It's very, very easy to avoid any sort of clipping processing by using hardware with drivers that don't have any! Avid, Apogee, MOTU, RME, and many others have bit transparent OSX CoreAudio drivers. Also, any DAC worth it's using can reconstruct far beyond 0dBFS without distortion, so hearing volume increase past -1..1 in software is not surprising. I recall the ADI 1955 and equivalent TI part putting out +12dBFS or something ridiculous, but those ain't Wolfson low power headphone codecs neither! On Tue, Dec 31, 2013 at 6:50 PM, Simon Wise simonzw...@gmail.com wrote: On 31/12/13 08:30, Dan Wilcox wrote: Ouch. I guess alot of us don't have serious projects :D (Out of curiosity, does Max do soft clipping also?) the point was that OSX was messing with the sound between the software, presumably any software, and the audio output ... which may perhaps be called a feature while listening to songs in iTunes but is a big worry if you are trying to use the system seriously as a musician. It is a problem that comes up on this list from time to time, I don't recall any reply saying it could be bypassed or turned off. Simon ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/ listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
Ouch. I guess alot of us don't have serious projects :D (Out of curiosity, does Max do soft clipping also?) On Dec 29, 2013, at 10:20 PM, pd-list-requ...@iem.at wrote: From: Miller Puckette m...@ucsd.edu Subject: Re: [PD] headroom in Pd Date: December 29, 2013 at 6:42:00 PM GMT+1 To: Martin Peach martin.pe...@sympatico.ca Cc: IOhannes m zmölnig zmoel...@iem.at, pd-list@iem.at This is frightening - if I were a musician reading this I'd be frightened to ever use Appe software in a serious project. Dan Wilcox @danomatika danomatika.com robotcowboy.com ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
here's the deal, if I have a square wave in Pd running at 1 -1 peak to peak, then you say that should be my maximum output, right? Thing is that if I give it an extra boost (say, multiply it by 2) I can clearly listen an increase in loudness. Hence, something in my system is allowing some headroom to be output. I got a macbook air from 2010 running 10.7.5... if Pd is not responsible for this, maybe my hardware + Mac OS is? here's the patch, try yourselves and tell me what you get please. Cheers #N canvas 653 26 257 182 10; #X obj 79 97 dac~; #X obj 85 41 square~ 440; #X floatatom 125 72 5 0 0 0 - - -; #X obj 85 70 *~; #X connect 1 0 3 0; #X connect 2 0 3 1; #X connect 3 0 0 0; #X connect 3 0 0 1; 2013/12/21, IOhannes m zmölnig zmoel...@iem.at: On 2013-12-21 14:58, peiman khosravi wrote: However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons! this really depends...a clipping sine will have loads of high frequencies that might be equally damaging to your audience. if you want to be safe, use math to make sure that your signal won't exceed -1..+1 before sending to the [dac~]. or use a limiter (zexy has a handy one). fgmrdsa IOhannes ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2013-12-29 10:08, Alexandre Torres Porres wrote: here's the deal, if I have a square wave in Pd running at 1 -1 peak to peak, then you say that should be my maximum output, right? Thing is that if I give it an extra boost (say, multiply it by 2) I can clearly listen an increase in loudness. Hence, something in my system is allowing some headroom to be output. I got a macbook air from 2010 running 10.7.5... if Pd is not responsible for this, maybe my hardware + Mac OS is? Yes it's a Mac-specific issue. On Win7 I get no difference above 1.0. The Apple audio driver is responsible for clipping values outside of [-1.0..1.0] as they arrive from possibly multiple applications. The docs state that clipping can be done in a soft way, so I suspect that the default driver (for the headphone output) is doing some sort of compression. Possibly if you use an external interface this won't happen (?). (see for example https://developer.apple.com/library/mac/documentation/DeviceDrivers/Conceptual/WritingAudioDrivers/ImplementDriver/ImplementDriver.html#//apple_ref/doc/uid/TP3732-BAJCBIAF ) Martin here's the patch, try yourselves and tell me what you get please. Cheers #N canvas 653 26 257 182 10; #X obj 79 97 dac~; #X obj 85 41 square~ 440; #X floatatom 125 72 5 0 0 0 - - -; #X obj 85 70 *~; #X connect 1 0 3 0; #X connect 2 0 3 1; #X connect 3 0 0 0; #X connect 3 0 0 1; 2013/12/21, IOhannes m zmölnig zmoel...@iem.at: On 2013-12-21 14:58, peiman khosravi wrote: However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons! this really depends...a clipping sine will have loads of high frequencies that might be equally damaging to your audience. if you want to be safe, use math to make sure that your signal won't exceed -1..+1 before sending to the [dac~]. or use a limiter (zexy has a handy one). fgmrdsa IOhannes ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
This is frightening - if I were a musician reading this I'd be frightened to ever use Appe software in a serious project. (Of course, we do't know what happens in Windows under the hood either. The only way you can truly know what you're getting is to use an open-source OS. cheers Miller On Sun, Dec 29, 2013 at 12:12:04PM -0500, Martin Peach wrote: On 2013-12-29 10:08, Alexandre Torres Porres wrote: here's the deal, if I have a square wave in Pd running at 1 -1 peak to peak, then you say that should be my maximum output, right? Thing is that if I give it an extra boost (say, multiply it by 2) I can clearly listen an increase in loudness. Hence, something in my system is allowing some headroom to be output. I got a macbook air from 2010 running 10.7.5... if Pd is not responsible for this, maybe my hardware + Mac OS is? Yes it's a Mac-specific issue. On Win7 I get no difference above 1.0. The Apple audio driver is responsible for clipping values outside of [-1.0..1.0] as they arrive from possibly multiple applications. The docs state that clipping can be done in a soft way, so I suspect that the default driver (for the headphone output) is doing some sort of compression. Possibly if you use an external interface this won't happen (?). (see for example https://developer.apple.com/library/mac/documentation/DeviceDrivers/Conceptual/WritingAudioDrivers/ImplementDriver/ImplementDriver.html#//apple_ref/doc/uid/TP3732-BAJCBIAF ) Martin here's the patch, try yourselves and tell me what you get please. Cheers #N canvas 653 26 257 182 10; #X obj 79 97 dac~; #X obj 85 41 square~ 440; #X floatatom 125 72 5 0 0 0 - - -; #X obj 85 70 *~; #X connect 1 0 3 0; #X connect 2 0 3 1; #X connect 3 0 0 0; #X connect 3 0 0 1; 2013/12/21, IOhannes m zmölnig zmoel...@iem.at: On 2013-12-21 14:58, peiman khosravi wrote: However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons! this really depends...a clipping sine will have loads of high frequencies that might be equally damaging to your audience. if you want to be safe, use math to make sure that your signal won't exceed -1..+1 before sending to the [dac~]. or use a limiter (zexy has a handy one). fgmrdsa IOhannes ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2013-12-20 23:34, Martin Peach wrote: On 2013-12-20 16:55, Alexandre Torres Porres wrote: Hi there, where can I find info about headroom and clipping on Pd. Or can anyone tell me quickly how it goes? Does it always really clip over a maximum of 1, or is there some headroom? Does it depend on the audiocard or something? The soundcard will always clip above +1 and below -1, and sometimes even yes, but the output of Pd (what you send to [dac~]) is not necessarily sent directly to the soundcard: e.g when using jack, the samples will be passed as floating point values to the next client: so samples exceeding -1..+1 need not clip at all. you can confirm this by connecting the output of Pd to the input of Pd via jack, and send a [osc~ 440]*10 ... but when you connect this output to the system output, you will get clipping. afaik, you get something similar on OSX, where the (portaudio) API will take floating point samples, and the signal gets sent through a limiter to prevent obvious clipping. so the bottom line is: hardware always has a physical range limit (that is mapped to -1..+1). but Pd is software and some audio APIs can handle sample values in floating point format just fine. with these, you most likely don't get any *immediate* problems if the range exceeds -1..+1. however, how excessive samples are handled is highly depending on the audio API and eventually other components in the signal chain. so if you do want to output signals that are outside -1..+1, you are on your own (from Pd's perspective). within those limits (if the interpolated waveform between samples goes over the limit). well, the reconstruction filters in the DAC won't necessarily clip in this case. gfamdsr IOhannes signature.asc Description: OpenPGP digital signature ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons! *www.peimankhosravi.co.uk http://www.peimankhosravi.co.uk || RSS Feed http://peimankhosravi.co.uk/miscposts.rss || Concert News http://spectralkimia.wordpress.com/* On 21 December 2013 13:48, IOhannes m zmölnig zmoel...@iem.at wrote: On 2013-12-20 23:34, Martin Peach wrote: On 2013-12-20 16:55, Alexandre Torres Porres wrote: Hi there, where can I find info about headroom and clipping on Pd. Or can anyone tell me quickly how it goes? Does it always really clip over a maximum of 1, or is there some headroom? Does it depend on the audiocard or something? The soundcard will always clip above +1 and below -1, and sometimes even yes, but the output of Pd (what you send to [dac~]) is not necessarily sent directly to the soundcard: e.g when using jack, the samples will be passed as floating point values to the next client: so samples exceeding -1..+1 need not clip at all. you can confirm this by connecting the output of Pd to the input of Pd via jack, and send a [osc~ 440]*10 ... but when you connect this output to the system output, you will get clipping. afaik, you get something similar on OSX, where the (portaudio) API will take floating point samples, and the signal gets sent through a limiter to prevent obvious clipping. so the bottom line is: hardware always has a physical range limit (that is mapped to -1..+1). but Pd is software and some audio APIs can handle sample values in floating point format just fine. with these, you most likely don't get any *immediate* problems if the range exceeds -1..+1. however, how excessive samples are handled is highly depending on the audio API and eventually other components in the signal chain. so if you do want to output signals that are outside -1..+1, you are on your own (from Pd's perspective). within those limits (if the interpolated waveform between samples goes over the limit). well, the reconstruction filters in the DAC won't necessarily clip in this case. gfamdsr IOhannes ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2013-12-21 14:58, peiman khosravi wrote: However, it's probably wise to clip the signal before sending it to dac~. Entirely for health and safety reasons! this really depends...a clipping sine will have loads of high frequencies that might be equally damaging to your audience. if you want to be safe, use math to make sure that your signal won't exceed -1..+1 before sending to the [dac~]. or use a limiter (zexy has a handy one). fgmrdsa IOhannes signature.asc Description: OpenPGP digital signature ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
[PD] headroom in Pd
Hi there, where can I find info about headroom and clipping on Pd. Or can anyone tell me quickly how it goes? Does it always really clip over a maximum of 1, or is there some headroom? Does it depend on the audiocard or something? thanks ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
Re: [PD] headroom in Pd
On 2013-12-20 16:55, Alexandre Torres Porres wrote: Hi there, where can I find info about headroom and clipping on Pd. Or can anyone tell me quickly how it goes? Does it always really clip over a maximum of 1, or is there some headroom? Does it depend on the audiocard or something? The soundcard will always clip above +1 and below -1, and sometimes even within those limits (if the interpolated waveform between samples goes over the limit). Pd will not clip internally, so you can calculate with larger numbers as long as you scale them back down before listening to them. Martin ___ Pd-list@iem.at mailing list UNSUBSCRIBE and account-management - http://lists.puredata.info/listinfo/pd-list
