Thanks for that Ross. Have been spending much of the day reading up on PIC programming so i can play with stuff that the GUI doesn't permit. I have a background in Java and C++ so it shouldn't take too long.
Will be interesting to see if your thoughts concur with Russ'. Are you still using a Lundahl transformer connected directly to the outputs of the Sabre? How are you doing volume control?
Cheers.
Will be interesting to see if your thoughts concur with Russ'. Are you still using a Lundahl transformer connected directly to the outputs of the Sabre? How are you doing volume control?
Cheers.
Spartacus said:
So far, the sound is very detailed. I like the 9 bit mode. I don't perceive that there is more THD this way. It's very difficult to do an A/B test when I have to swap out processor chips.
The test rig has a Jensen 4:1 transformer directly on the outputs with no op-amps. It lowers the volume a lot so I have to turn up the gain on my preamp. My PCB that I am designing will use op-amp IV conversion and then a Jensen 1:1 line output transformer that is capable of very low distortion when driven with a low impedance source.
I have no plans on doing the volume control in the Sabre. I plan on using a preamp.
BRN said:
Hi Brad,
Right now I have it connected as a passive transformer output. The 4:1 transformer is connected to the DAC outputs and a short cable to a high-impedance preamp input.
An op-amp IV circuit will likely have a little less distortion and will be able to drive longer cables, headphones and the like.
Still, The passive transformer on the Sabre8 sounds very good.
Transformer output stages give DC isolation between the power supplies of the DAC and the pre/power amps. Since I am using optical and coax-transformer SPDIF inputs, all inputs and outputs are isolated from the other equipment to prevent unwanted noise currents.
I live in a 50 year old house. The electric was upgraded in the 1980s but there is still a lot of noise on the power lines. I go to great lengths to get rid of pops, clicks and hum
Hey guys,
I have a couple DACs running now (and 4 different I/V stages so far), so I can A/B pretty easily.
I can't really tell much difference sound wise between 6-bit/true differential, and 9-bit pseudo differential. That is to say they both sound great, but that's not the end of the story. The HF noise coming out of the DAC in 6bit mode is noticeably more than it is in 9 bit mode. And this does not seem like a small amount either.
One other note. In 6 bit mode to get the best performance you should use the n/64 notch delay.
What I have noticed, is that in the 9bit(pseudo differential) mode that DAC output seems to require less filtering while still sounding very sweet. The amount of digital garbage(quantization noise I guess) that makes it out of the analog inputs seems to be much reduced.
Running the analog section with a less aggressive filter has some nice benefits (better slew rate etc) to the final audio.
If you plan a little you can use the output impedance of the Sabre to your advantage. You can create a low pass RC filter to the input of a fully differential output stage. I have had my best sounding results so far using this scheme. It results in a not exactly orthodox I/V scheme, but it is a very effective one and it keeps the components in the signal path to a minimum.
One thing is for sure, you must filter out the quantization noise. This DAC requires more attention at the I/V stage than a PCM1794A to be sure. This is especially true in the stereo mode!!! 4 times the noise per channel, and unfortunately its not all common mode.
Cheers!
Russ
I have a couple DACs running now (and 4 different I/V stages so far), so I can A/B pretty easily.
I can't really tell much difference sound wise between 6-bit/true differential, and 9-bit pseudo differential. That is to say they both sound great, but that's not the end of the story. The HF noise coming out of the DAC in 6bit mode is noticeably more than it is in 9 bit mode. And this does not seem like a small amount either.
One other note. In 6 bit mode to get the best performance you should use the n/64 notch delay.
What I have noticed, is that in the 9bit(pseudo differential) mode that DAC output seems to require less filtering while still sounding very sweet. The amount of digital garbage(quantization noise I guess) that makes it out of the analog inputs seems to be much reduced.
Running the analog section with a less aggressive filter has some nice benefits (better slew rate etc) to the final audio.
If you plan a little you can use the output impedance of the Sabre to your advantage. You can create a low pass RC filter to the input of a fully differential output stage. I have had my best sounding results so far using this scheme. It results in a not exactly orthodox I/V scheme, but it is a very effective one and it keeps the components in the signal path to a minimum.
One thing is for sure, you must filter out the quantization noise. This DAC requires more attention at the I/V stage than a PCM1794A to be sure. This is especially true in the stereo mode!!! 4 times the noise per channel, and unfortunately its not all common mode.
Cheers!
Russ
Russ,
Forgive the rudimentary understanding; are you suggesting to use the ~700 ohm output impedance (from memory) with a cap to make an RC low pass filter, used in conjunction with differential outputs? If so, to filter out a little quantization noise from the 192 khz output, what freq would we need to aim for? (around 96khz?)
BTW, received all my goods safe n sound. Wow, they are so much smaller than I imagined! Thanks!
Forgive the rudimentary understanding; are you suggesting to use the ~700 ohm output impedance (from memory) with a cap to make an RC low pass filter, used in conjunction with differential outputs? If so, to filter out a little quantization noise from the 192 khz output, what freq would we need to aim for? (around 96khz?)
BTW, received all my goods safe n sound. Wow, they are so much smaller than I imagined! Thanks!
risingtenpi said:
Hello, and your welcome.
Its a little more complex than just adding caps. You have to add a bit of series resistance right after the filter caps. Tis make the IVY stage not very close to zero impedance. This would normally be a bit strange. But with a fully differential output stage, it actually make a lot of sense. What more, it sounds excellent.
I will give more details later, I have not settled on exactly the best way to do it yet and I am still testing things.
Cheers!
Russ
I'm not sure if it had come up already, but I just saw that ESS has put some more info on their download page. Specially this document should be usefull to the DIY community: http://www.esstech.com/techsupp/Application_Note_Component_Selection_and_PCB_Layout.pdf
Denon DVD-1930CI is similar to the 1920
The one obvious difference is that the 1930 has an RS232 port while the 1920does not. It's connections originate in the region of the HDMI plug but I haven't investigated further. The manual describes no function for this port - it only mentions something lame about 'future compatability'. I bet they weren't planning for it to be hijacked by DIYers!
BTW, those vias with the DSD are both small diameter and closely spaced.
Frank in Mpls.
The one obvious difference is that the 1930 has an RS232 port while the 1920does not. It's connections originate in the region of the HDMI plug but I haven't investigated further. The manual describes no function for this port - it only mentions something lame about 'future compatability'. I bet they weren't planning for it to be hijacked by DIYers!
BTW, those vias with the DSD are both small diameter and closely spaced.
Frank in Mpls.
Attachments
Internal digital volume control
Hi all,
I have looked, perhaps I just missed it, but I have not found any detail info on how the internal Sabre volume control operates in the digital domain. Can someone please help me on this and perhaps pin point possible positive and negative effects using the internal digital volume control as opposed to an external analog one?
Does the bit width of the source come into play here? E.g. does the volume control perform better if I have a 24-bit source than if I have a 16-bit source?
Regards,
Dex
Hi all,
I have looked, perhaps I just missed it, but I have not found any detail info on how the internal Sabre volume control operates in the digital domain. Can someone please help me on this and perhaps pin point possible positive and negative effects using the internal digital volume control as opposed to an external analog one?
Does the bit width of the source come into play here? E.g. does the volume control perform better if I have a 24-bit source than if I have a 16-bit source?
Regards,
Dex
Re: Internal digital volume control
Hi Dex,
Dustin wrote about the volume control in post 456 in this thread:
http://www.diyaudio.com/forums/showthread.php?postid=1488733#post1488733
Volume control is after the digital filter. The 16 or 24 bit source shouldn't matter.
DexterMorgan said:
Hi Dex,
Dustin wrote about the volume control in post 456 in this thread:
http://www.diyaudio.com/forums/showthread.php?postid=1488733#post1488733
Volume control is after the digital filter. The 16 or 24 bit source shouldn't matter.
Is2
Hi,
I have been using the Buffalo for a couple of weeks. It sounds very good. This is my 4th dac and it is the first dac I have heard were the cymbles sound right. They are not too clangy, bright, or hard on the top range. The even, balanced sound from the Buffalo is amazing. I am using it SPDIF in with the Ivy with single ended outputs. Great job guys.
This dac also sounded the same when using the sony es cdp or the modified toshiba 3950 as a source. That never happened before.
I read about Is2 and DSD sources and google ect but still am not sure about them. It seems they are SACD generated digital sources. I am only using Redbook CD and vinyl as sources for now, with a computer source in the works. Would I be able to use Is2 or DSD with whose types of sources? Thanks.
PS: 97% of this I can understand. You guys did deep into this stuff!
Hi,
I have been using the Buffalo for a couple of weeks. It sounds very good. This is my 4th dac and it is the first dac I have heard were the cymbles sound right. They are not too clangy, bright, or hard on the top range. The even, balanced sound from the Buffalo is amazing. I am using it SPDIF in with the Ivy with single ended outputs. Great job guys.
This dac also sounded the same when using the sony es cdp or the modified toshiba 3950 as a source. That never happened before.
I read about Is2 and DSD sources and google ect but still am not sure about them. It seems they are SACD generated digital sources. I am only using Redbook CD and vinyl as sources for now, with a computer source in the works. Would I be able to use Is2 or DSD with whose types of sources? Thanks.
PS: 97% of this I can understand. You guys did deep into this stuff!
Digital volume control
Thanks for the post info with the link to the patent associated with the digital volume control in the Sabre chip.
So far, has anyone of you guys done any measuring, and/or subjective testing for that matter, regarding using the internal volume control or not. I take it the internal volume control can be bypassed if one would want to?
Regards,
Dex
Thanks for the post info with the link to the patent associated with the digital volume control in the Sabre chip.
So far, has anyone of you guys done any measuring, and/or subjective testing for that matter, regarding using the internal volume control or not. I take it the internal volume control can be bypassed if one would want to?
Regards,
Dex
Hey, I just noticed something:
The datasheet says to use 386*Fs for SPDIF input, the half for i2s (values may not be correct, that's just what I quickly remember). And that the clock should not be more than +10% to keep jitter reduction.
The point is: what Fs do we talk about? I mean do we take 192kHz even if the incoming data will run at 44.1kHz? Wouldn't we then have some advantage taking a programmable clock that would output the right clock, or as there's oversampling the internal clock rate will remain at 192k...
Thank you,
Nicolas
The datasheet says to use 386*Fs for SPDIF input, the half for i2s (values may not be correct, that's just what I quickly remember). And that the clock should not be more than +10% to keep jitter reduction.
The point is: what Fs do we talk about? I mean do we take 192kHz even if the incoming data will run at 44.1kHz? Wouldn't we then have some advantage taking a programmable clock that would output the right clock, or as there's oversampling the internal clock rate will remain at 192k...
Thank you,
Nicolas
Edit:
Sorry, I promise you I learn to reed and drink less whisky...
The paper said
Master Clock
As detailed in the datasheet, the MCLK must be at least 386*Fs for SPDIF inputs and 192*Fs for
Serial/DSD inputs. For optimum jitter tolerance the MCLK should be at least 10% higher than this
theoretical MCLK minimum, this ensures audio clarity free from input clock jitter.
If operating with a synchronous MCLK, it is recommended to use an inverted MCLK. The inverted
synchronous MCLK ensures that the Sabre noise is as low as possible.
AT LEAST, not "less than".
I'm really stupid sometimes.
Ok, so we're cool.
I'm just waiting for the quote from Vectron for their TCXO C2310. Should be around 50$ - a bit expensive but when you look at the phase noise figure it's not the one that displays the absolute lowest jitter, but it's jitter is more out-of band and less in the audio band
Sorry, I promise you I learn to reed and drink less whisky...
The paper said
Master Clock
As detailed in the datasheet, the MCLK must be at least 386*Fs for SPDIF inputs and 192*Fs for
Serial/DSD inputs. For optimum jitter tolerance the MCLK should be at least 10% higher than this
theoretical MCLK minimum, this ensures audio clarity free from input clock jitter.
If operating with a synchronous MCLK, it is recommended to use an inverted MCLK. The inverted
synchronous MCLK ensures that the Sabre noise is as low as possible.
AT LEAST, not "less than".
I'm really stupid sometimes.
Ok, so we're cool.
I'm just waiting for the quote from Vectron for their TCXO C2310. Should be around 50$ - a bit expensive but when you look at the phase noise figure
it's not the one that displays the absolute lowest jitter, but it's jitter is more out-of band and less in the audio band
Oops, my edit came too late... sorry.
Well, to feed the thread:
Someone here says he got the best results using a WM8804 before the DAC.
Despite the Wolfson unit has a worse spdif receiver than the ESS, and given the ESS does jitter reducing on its i2s inputs too, that may be explained by the use of a 40Mhz XO instead of a 80Mhz one - thus greatly reducing jitter/phase noise.
No?
Well, to feed the thread:
Someone here says he got the best results using a WM8804 before the DAC.
Despite the Wolfson unit has a worse spdif receiver than the ESS, and given the ESS does jitter reducing on its i2s inputs too, that may be explained by the use of a 40Mhz XO instead of a 80Mhz one - thus greatly reducing jitter/phase noise.
No?