Re: [music-dsp] Inquiry: new systems for Live DSP
On 15/03/2014 1:46 AM, Richard Dobson wrote: But portaudio only states the software i/o buffer latency, it knows nothing directly of internal codec latencies. You would need to subtract the (two-way?) buffer latency portaudio reports, and then measure or compute how much of the remainder is down to the dac filtering. I am not entirely sure how accurate that portaudio estimate is anyway - it's an area of portaudio that may not be entirely bug-free. It has been discussed on the portaudio list fairly recently IIRC. Possible bugs not withstanding, there are indeed limits to what PortAudio can report with respect to latency. However the following is not strictly correct: portaudio only states the software i/o buffer latency In summary, PortAudio reports the sum of: - any buffering latency introduced by PortAudio (often this is zero) AND - any buffering latency that can be inferred to be introduced by the native API. AND/OR - any latency (of any kind, not just buffering) that is reported by the native API and/or the driver. Some native APIs (CoreAudio, ASIO) provide mechanisms for the audio driver to report detailed latency information for components below the client software interface. In general PortAudio aims to surface that information to the PortAudio client. Many of the bugs in PA were fixed (hence discussions), and those that remain I think relate to imperfect inference when the native APIs that don't explicitly report latency information, or with obscure edge cases (multiple-device full duplex I/O on OS X comes to mind). As has been pointed out elsewhere, if an audio interface has a digital interface, its driver can't report the latency introduced by arbitrary ADC/DACs connected to the digital interface. As far as I know, it is largely undocumented/undefined whether the latency reported by an ASIO driver includes the hardware ADC/DAC delays for hardware that has analog IO. CoreAudio reports multiple latency components, and may well go all the way to reporting converter latency (I don't remember). I'm not sure about ALSA. Ross. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
I've ordered these for my musical digital processing activities: http://www.diyinhk.com/shop/audio-kits/49-isolated-xmos-384khz-high-quality-usb-to-i2s-pcb-with-ultralow-noise-65uv-regulator.html http://www.diyinhk.com/shop/audio-kits/31-384khz32bit-pcm5102a-dac-i2s-input-ultra-low-noise-regulator.html So I can use 384kHz sample frequency on any PC, with a very good spec DA convertor (32 bit, whatever that will mean), and a ground lift for the amplifiers from the computer ground. I've seen there are recent Asus offerings which an do some of this too, but these two products I feel are interesting compared with what's out there. For the people with the interest in playing with DSP n FPGAs (programmable logic), which IMO is pretty interesting, it is possible to use the $29.50 DA convertor board with an fpga using 4 signals, to make a good sounding synthesizer or even to plug the FPGA in between the USB-to-I2S connector and the convertor, to test effects. No more news at the moment, but I think, if it all arrives and works properly, may be a nice answer to some of the requests here lately. T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
On Fri, Mar 14, 2014 at 4:46 PM, Richard Dobson richarddob...@blueyonder.co.uk wrote: On 14/03/2014 14:27, Olli Niemitalo wrote: http://yehar.com/Fast%20Track%20Ultra%2048%20kHz%20output-input%20ir.jpg It looks more like a minimum-phase lowpass filter. The marker at sample #29 indicates what PortAudio thinks the latency is: 29 samples. But portaudio only states the software i/o buffer latency, it knows nothing directly of internal codec latencies. OK, my misunderstanding, I simply used paStreamInfo-inputLatency + paStreamInfo-outputLatency. -olli -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
On 2014-03-14, Olli Niemitalo wrote: Not sure what's going on at the lowest bass frequencies, as the peak there could be a measurement artifact. I'm guessing that's just the minimum phaseness of the kernel intruding. Remember, for a minimum phase filter the phase response equals the log Hilbert transform of the magnitude one. The Hilbert transform is then essentially a derivative plus a linear phase low pass to equalize the magnitude. From that perspective, rapid rolloff at band edge will give precisely the kinds of peaks in the group delay you're seeing in the picture. Or you could take the Hilbert transform pair for the characteristic function of an interval [a,b], which is 1/pi log |(x-a)/(x-b)|; that has a pair of singularities at the endpoints. I'm betting if you plot the group delay on a log frequency scale (or better yet constant ERB), you'll see that they distributed the group delay variation so that it doesn't lead to significant envelope dispersion over any given critical band. Plus of course if you did a full roundtrip through two such converters, any deviation from the ideal brickwall response near the band edges would be squared, making the plot somewhat more difficult to interpret. Why you'd be seeing a lower band edge is beyond me, however. It isn't as though you can't drive delta-sigma converters right down to DC. Maybe there's a capacitive coupling there somewhere, or something... Ardatech calls such a thing a lowest group delay filter and TI calls it a low latency filter. They are useful when running a real-time input--output effect like Guitar Rig, or for software monitoring of inputs, but will hinder some more analytical uses of a sound card. Peter Craven once speculated in the JAES (I think) that such filters might actually lead to more accurate reconstruction of spatial and envelopment cues because they don't exhibit preringing. Another way to go about the same thing is to use higher sampling rates and something like a Bessel filter instead of a sinc, leading to no ringing at all but a much gentler magnitude rolloff. The call's out on such things, nobody really knows if there's any point to them. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
For my own interest, it suffices to start with this: Opamp(--filter)--AD--Fast-low-latency(+big-memory)-DSP--DA--buffer/filter( withnot too much effect in 20Hz-20kHz) as a generality, and such high frequency, and (like I proposed) such accurate convertors that I'd be going from 5xCD sampling quality=192kHz to 5x5xCD sampling quality ~= 1MHz. It interests me that such chain with outstanding enough specs for serious audio (20dB rustling leaves level to fully over the top 120dB rock levels, with still relevant bits which properly average static, leaving a nice noise floor that pretty perfectly retains small information that could be filtered out when needed) and measurement (bit accuracy is pretty amazing, up to a few parts in a million all the way through, and the AD IIRC can actually measure at 0dBFS up to a MHz, with much higher internal sampling frequency) is available without a stellar budget. For testing out PWM stuff, or rendering mathematical waves or simulated schematics (with good accuracy, maybe I'll need to find a nice 64 bit SPICE simulator), performing live DSP based audio feedback through amps and speakers, and even for testing out what filters can be used to properly work with very high sampling frequency, and short internal delay (and how many bits are needed for that). I have good hope from working a lot at 192kHz that 1 or 2 MHz should be fine for most things. T.V. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
On 12 Mar 2014, at 15:53, Theo Verelst theo...@theover.org wrote: A DA converter with 20 bits (actual) accuracy, pretty low noise, and a settling time of 1 microsecond, driven by more or less standard 3 wire interface: http://www.analog.com/en/digital-to-analog-converters/da-converters/ad5791/products/product.html (eval board: http://www.analog.com/en/evaluation/eval-ad5791/eb.html ) That doesn’t mean it has a usable sample rate up to 1 MHz. After 1 us, the glitch is over and you can turn on your S/H stage to take over the value. Without a S/H stage, you have a nice noise floor from the glitches. IMO, for 100 or 200 kHz, it might be workable, but not 1 MHz. And an AD converter with 2.5 MS/S and 24 bits accuracy sigma delta which should be great for measurements and undisputably great sampling behavior for perfect delay lines, etc.: http://www.analog.com/en/analog-to-digital-converters/ad-converters/ad7760/products/product.html (eval board: http://www.analog.com/en/evaluation/eval-ad7760/eb.html ) Sure, but no antialiasing filters up front. How about clocking the whole show, etc. Take this one: http://www.analog.com/en/codec-afe/audio-codecs/ad1939/products/product.html Group delay can go down to 0.5 ms.That’s about 17 cm audio thru air …. Best, Steffan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Inquiry: new systems for Live DSP
And an AD converter with 2.5 MS/S and 24 bits accuracy sigma delta which should be great for measurements and undisputably great sampling behavior for perfect delay lines, etc.: http://www.analog.com/en/analog-to-digital-converters/ad-converters/ad7760/products/product.html (eval board: http://www.analog.com/en/evaluation/eval-ad7760/eb.html ) Sure, but no antialiasing filters up front. How about clocking the whole show, etc. Also notice that that one doesn't claim 24 bit accuracy. The SNR/dynamic range specs work out to about 16 bits accuracy at the highest sampling rate, and can go up to 18-20 bits accuracy if you knock the sampling rate down to 77kHz. It's a 24-bit sigma-delta modulator, not 24 bits of actual output performance. E On Thu, Mar 13, 2014 at 6:55 AM, STEFFAN DIEDRICHSEN sdiedrich...@me.comwrote: On 12 Mar 2014, at 15:53, Theo Verelst theo...@theover.org wrote: A DA converter with 20 bits (actual) accuracy, pretty low noise, and a settling time of 1 microsecond, driven by more or less standard 3 wire interface: http://www.analog.com/en/digital-to-analog-converters/da-converters/ad5791/products/product.html (eval board: http://www.analog.com/en/evaluation/eval-ad5791/eb.html ) That doesn't mean it has a usable sample rate up to 1 MHz. After 1 us, the glitch is over and you can turn on your S/H stage to take over the value. Without a S/H stage, you have a nice noise floor from the glitches. IMO, for 100 or 200 kHz, it might be workable, but not 1 MHz. And an AD converter with 2.5 MS/S and 24 bits accuracy sigma delta which should be great for measurements and undisputably great sampling behavior for perfect delay lines, etc.: http://www.analog.com/en/analog-to-digital-converters/ad-converters/ad7760/products/product.html (eval board: http://www.analog.com/en/evaluation/eval-ad7760/eb.html ) Sure, but no antialiasing filters up front. How about clocking the whole show, etc. Take this one: http://www.analog.com/en/codec-afe/audio-codecs/ad1939/products/product.html Group delay can go down to 0.5 ms.That's about 17 cm audio thru air Best, Steffan -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
