darrenyeats
pfm Member
IMO the DAC1's true potential is realised when the extras (XLR output attenuators, physical volume pot) are switched off, and it is fed a digital signal that is attenuated, upsampled well (e.g. SoX with -v) to 110.6kHz then dithered at 24 bits. If used via XLR, also the XLR calibration trimmers (NOT to be confused with output attenuators) must be used.
XLR is a balanced connection, this offers a high degree of common mode noise rejection, which in some systems has sonic benefits.
(1) to (7) took me to "now the sound is not annoying".
Finally (8) took me to "now it sounds good".
1. Use XLR but disable XLR output attenuators by setting to 0db (factory is -20db).
2. Use 'calibrated' mode which bypasses the volume pot.
3. Use the internal 'calibration trimmers' to reduce the XLR output levels. It should be possible to reduce output by at least 15dB, I got more like 20dB. See notes *.
4. Use digital attenuation in the transport. It should be dithered at 24 bits. The ideal way to do this is discussed in point 8!
Why? "DSP headroom"; see pictures in post 11 later in thread; see Figs 1, 2 and 13 at https://benchmarkmedia.com/blogs/ap...chmark-dac2-vs-dac1-side-by-side-measurements.
Why dithered? Dither makes sure the errors take the form of noise instead of distortion.
Why dither at 24 bits? The DAC1 accepts input, and operates, at 24 bits.
Make day-to-day use of the digital volume control in the transport and never go above a certain level of attenuation****
If you own a Squeezebox Touch see http://forums.slimdevices.com/showthread.php?104629-Dithered-volume-control-for-Squeezebox and http://forums.slimdevices.com/showthread.php?105197-All-day-battery-power-for-Touch-%A324.
5. Pay attention to cabling and power supply. See here: http://www.pinkfishmedia.net/forum/showpost.php?p=2456579&postcount=126.
6. I found S/PDIF TOSLINK sounds better than coax with Squeezebox Touch into the DAC1 - maybe due to the combination of galvanic isolation of TOSLINK and jitter reduction of the DAC1. I would not use USB: http://forums.slimdevices.com/showt...nly)-exist-now&p=787260&viewfull=1#post787260
7. Let it warm up to a stable temperature, half an hour should be sufficient. See http://www.soundonsound.com/sos/jul05/articles/benchmark.htm (sidebar 'A Warm Feeling'). There is some evidence that leaving a DAC switched on longer can measurably help clock stability, I think this will be inaudible but for reference:
http://www.audioquest.com/wp-content/uploads/2014/04/Phase-Noise-Jitter-Report-0317-14.pdf.
8. If you have a Squeezebox Touch with Linux server**, upsample to 110.6kHz using SoX prior to feeding files into the DAC1. The sox command should include attenuation (e.g. vol 0.5amplitude which to SoX is a 1 bit shift) followed by 'rate -v -b 89 110600' (this will use the default linear phase filter, -v means very high quality, -b 89 gives an optimally smooth drop off between 19kHz and 22kHz assuming a 44.1kHz rate - for an informative thread see http://www.pinkfishmedia.net/forum/showthread.php?p=3040013#post3040013). Ensure the sox command ends with 'dither' directive. This can be done to music files beforehand. If you have a Squeezebox Touch: you will have to install EDO to support this rate; you could also in convert.conf set up the sox call to convert on the fly all streamed music; ideally combine with (4) so bit shift, upsampling, volume attenuation and dither are done in one SoX call for maximum precision - recommended! PM me for detailed instructions!
The DAC1 can accept arbitrary rates (between 28kHz and 195kHz) because of its ASRC. The manual states the TOSLINK works up to 96kHz, but it seems to work at 110,120,130kHz - however not 140kHz***!
You can also try a similar filter but flat to 20kHz (for 16/44) using "-b 92.5" instead.
--
Steps 1-3 is how you can get an XLR output level of around 2VRMS and output impedance of 60 ohms *at the same time*. You can't achieve this any other way than steps 1-3 with a DAC1, and you can't achieve it at all with a DAC2/3 (because there is no step 3 possible with a DAC2/3). Reasonable output level and good output impedance are relevant if you're using your DAC as a DAC/pre. Output impedance is less important if using a separate pre or integrated amp.
Other options that give an XLR output level of <=2VRMS and output impedance of <=60ohms *at the same time*:
Various Weiss DACs
Various dCS DACs
...!
* (Point 3) Benchmark confirmed to me by email that the 'calibration trimmers' don't raise output impedance. (By contrast the 'output attenuators' do raise output impedance, for me a sign they are in the "wrong" place i.e. after the preamp output stage.) To achieve point 3 you will need test tones, a DMM and a precision screwdriver set - and possibly an amp - to calibrate left and right channels. If you need help then PM me.
(Tip: considering output levels of modern sources, it's good FIRST to use any input sensitivity settings on your power amp or actives to minimise loudness. Then, set the DAC1's output levels. That's the order of priority. But you could well find it appropriate to use quietest settings at both stages, like I did. Also see note "****" below!)
** (Point 8) not tried Windows server; also all this might be possible with Signalyst HQ Player, but I'm not sure if it supports non-standard rates or if PC audio cards generally support non-standard rates.
*** (Point 8) this is evidence to me that the Squeezebox Touch is really outputting non-standard rates. It may be that Squeezebox Touch and DAC1 are a "peaches and cream" combo because this works!
**** Actually, digital -25dB seems to result in better sound than -5dB despite the technical loss of SNR - at least this is the state of affairs after doing steps 1-8 in my case. The sound is effortless, I don't know why. One should be flexible for quieter recordings. Personally, I most often range between -30dB and -20dB, sometimes higher or lower - but I never go higher than -17dB. If you think this is foo, then at least stay below about -5dB since that at least gives necessary DSP headroom!
--
Benchmark say the volume pot is transparent, the output attenuators are transparent, the digital interfaces all sound the same etc ... but the DAC1 measurements on Stereophile are taken in its stripped back state with volume pot DISABLED and output attenuators DISABLED. And also, it seems, whilst using its TOSLINK input. This might explain why it measures so well and yet audiophiles complain about the sound (when listening with attenuators in circuit, volume pot in circuit and using USB, for example).
See section "Addendum to Benchmark DAC1 Review" here https://positive-feedback.com/Issue26/benchmark_dac1.htm and section "The Jumpers" here http://www.positive-feedback.com/Issue48/benchmark_usb.htm I dismissed it as audiophile rubbish for one or two years but, since I experimented, I believe it really isn't.
It is true that each digital interface is resistant to input jitter, but it's not true that each interface exhibits exactly the same jitter (Ken Rockwell has measured the TOSLINK as slightly better than the coax and far better than USB).
Original post:
Issue 1. Channel balance.
Ken Rockwell reviewed the HDR some time ago and observed that channel imbalance using the potentiometer peaks at 1.25db around the low extremity of volume. The HDR has a good custom-made ALPS potentiometer: I suspect these comments apply to many other devices (although some stepped attenuators might be better for channel balance).
http://kenrockwell.com/audio/benchmark/dac1-hdr.htm#meas - Channel Tracking
Issue 2. DSP Headroom
According to Ken Rockwell, many DACs (NOT just the DAC1) can be tripped up by extreme waveforms.
http://kenrockwell.com/audio/benchmark/dac1-hdr.htm#meas - Square Wave Spectra (overshoot handling)
Note this has been remedied in the DAC2 HGC
http://www.benchmarkmedia.com/dac/dac2-hgc - High Headroom DSP - with 3.5 dB "Excess" Digital Headroom
Note the 3.5db figure is not related to anything specific about the DAC1, it is a generic figure based on the amount of headroom needed for almost any extreme (e.g. hyper-compressed) recording. Natural music will most likely not be affected. This figure is applicable to many other upsampling DACs which have the same DSP headroom situation.
XLR is a balanced connection, this offers a high degree of common mode noise rejection, which in some systems has sonic benefits.
(1) to (7) took me to "now the sound is not annoying".
Finally (8) took me to "now it sounds good".
1. Use XLR but disable XLR output attenuators by setting to 0db (factory is -20db).
2. Use 'calibrated' mode which bypasses the volume pot.
3. Use the internal 'calibration trimmers' to reduce the XLR output levels. It should be possible to reduce output by at least 15dB, I got more like 20dB. See notes *.
4. Use digital attenuation in the transport. It should be dithered at 24 bits. The ideal way to do this is discussed in point 8!
Why? "DSP headroom"; see pictures in post 11 later in thread; see Figs 1, 2 and 13 at https://benchmarkmedia.com/blogs/ap...chmark-dac2-vs-dac1-side-by-side-measurements.
Why dithered? Dither makes sure the errors take the form of noise instead of distortion.
Why dither at 24 bits? The DAC1 accepts input, and operates, at 24 bits.
Make day-to-day use of the digital volume control in the transport and never go above a certain level of attenuation****
If you own a Squeezebox Touch see http://forums.slimdevices.com/showthread.php?104629-Dithered-volume-control-for-Squeezebox and http://forums.slimdevices.com/showthread.php?105197-All-day-battery-power-for-Touch-%A324.
5. Pay attention to cabling and power supply. See here: http://www.pinkfishmedia.net/forum/showpost.php?p=2456579&postcount=126.
6. I found S/PDIF TOSLINK sounds better than coax with Squeezebox Touch into the DAC1 - maybe due to the combination of galvanic isolation of TOSLINK and jitter reduction of the DAC1. I would not use USB: http://forums.slimdevices.com/showt...nly)-exist-now&p=787260&viewfull=1#post787260
7. Let it warm up to a stable temperature, half an hour should be sufficient. See http://www.soundonsound.com/sos/jul05/articles/benchmark.htm (sidebar 'A Warm Feeling'). There is some evidence that leaving a DAC switched on longer can measurably help clock stability, I think this will be inaudible but for reference:
http://www.audioquest.com/wp-content/uploads/2014/04/Phase-Noise-Jitter-Report-0317-14.pdf.
8. If you have a Squeezebox Touch with Linux server**, upsample to 110.6kHz using SoX prior to feeding files into the DAC1. The sox command should include attenuation (e.g. vol 0.5amplitude which to SoX is a 1 bit shift) followed by 'rate -v -b 89 110600' (this will use the default linear phase filter, -v means very high quality, -b 89 gives an optimally smooth drop off between 19kHz and 22kHz assuming a 44.1kHz rate - for an informative thread see http://www.pinkfishmedia.net/forum/showthread.php?p=3040013#post3040013). Ensure the sox command ends with 'dither' directive. This can be done to music files beforehand. If you have a Squeezebox Touch: you will have to install EDO to support this rate; you could also in convert.conf set up the sox call to convert on the fly all streamed music; ideally combine with (4) so bit shift, upsampling, volume attenuation and dither are done in one SoX call for maximum precision - recommended! PM me for detailed instructions!
The DAC1 can accept arbitrary rates (between 28kHz and 195kHz) because of its ASRC. The manual states the TOSLINK works up to 96kHz, but it seems to work at 110,120,130kHz - however not 140kHz***!
You can also try a similar filter but flat to 20kHz (for 16/44) using "-b 92.5" instead.
--
Steps 1-3 is how you can get an XLR output level of around 2VRMS and output impedance of 60 ohms *at the same time*. You can't achieve this any other way than steps 1-3 with a DAC1, and you can't achieve it at all with a DAC2/3 (because there is no step 3 possible with a DAC2/3). Reasonable output level and good output impedance are relevant if you're using your DAC as a DAC/pre. Output impedance is less important if using a separate pre or integrated amp.
Other options that give an XLR output level of <=2VRMS and output impedance of <=60ohms *at the same time*:
Various Weiss DACs
Various dCS DACs
...!
* (Point 3) Benchmark confirmed to me by email that the 'calibration trimmers' don't raise output impedance. (By contrast the 'output attenuators' do raise output impedance, for me a sign they are in the "wrong" place i.e. after the preamp output stage.) To achieve point 3 you will need test tones, a DMM and a precision screwdriver set - and possibly an amp - to calibrate left and right channels. If you need help then PM me.
(Tip: considering output levels of modern sources, it's good FIRST to use any input sensitivity settings on your power amp or actives to minimise loudness. Then, set the DAC1's output levels. That's the order of priority. But you could well find it appropriate to use quietest settings at both stages, like I did. Also see note "****" below!)
** (Point 8) not tried Windows server; also all this might be possible with Signalyst HQ Player, but I'm not sure if it supports non-standard rates or if PC audio cards generally support non-standard rates.
*** (Point 8) this is evidence to me that the Squeezebox Touch is really outputting non-standard rates. It may be that Squeezebox Touch and DAC1 are a "peaches and cream" combo because this works!
**** Actually, digital -25dB seems to result in better sound than -5dB despite the technical loss of SNR - at least this is the state of affairs after doing steps 1-8 in my case. The sound is effortless, I don't know why. One should be flexible for quieter recordings. Personally, I most often range between -30dB and -20dB, sometimes higher or lower - but I never go higher than -17dB. If you think this is foo, then at least stay below about -5dB since that at least gives necessary DSP headroom!
--
Benchmark say the volume pot is transparent, the output attenuators are transparent, the digital interfaces all sound the same etc ... but the DAC1 measurements on Stereophile are taken in its stripped back state with volume pot DISABLED and output attenuators DISABLED. And also, it seems, whilst using its TOSLINK input. This might explain why it measures so well and yet audiophiles complain about the sound (when listening with attenuators in circuit, volume pot in circuit and using USB, for example).
See section "Addendum to Benchmark DAC1 Review" here https://positive-feedback.com/Issue26/benchmark_dac1.htm and section "The Jumpers" here http://www.positive-feedback.com/Issue48/benchmark_usb.htm I dismissed it as audiophile rubbish for one or two years but, since I experimented, I believe it really isn't.
It is true that each digital interface is resistant to input jitter, but it's not true that each interface exhibits exactly the same jitter (Ken Rockwell has measured the TOSLINK as slightly better than the coax and far better than USB).
Original post:
Issue 1. Channel balance.
Ken Rockwell reviewed the HDR some time ago and observed that channel imbalance using the potentiometer peaks at 1.25db around the low extremity of volume. The HDR has a good custom-made ALPS potentiometer: I suspect these comments apply to many other devices (although some stepped attenuators might be better for channel balance).
http://kenrockwell.com/audio/benchmark/dac1-hdr.htm#meas - Channel Tracking
Issue 2. DSP Headroom
According to Ken Rockwell, many DACs (NOT just the DAC1) can be tripped up by extreme waveforms.
http://kenrockwell.com/audio/benchmark/dac1-hdr.htm#meas - Square Wave Spectra (overshoot handling)
Note this has been remedied in the DAC2 HGC
http://www.benchmarkmedia.com/dac/dac2-hgc - High Headroom DSP - with 3.5 dB "Excess" Digital Headroom
Note the 3.5db figure is not related to anything specific about the DAC1, it is a generic figure based on the amount of headroom needed for almost any extreme (e.g. hyper-compressed) recording. Natural music will most likely not be affected. This figure is applicable to many other upsampling DACs which have the same DSP headroom situation.
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