I just watched the 6 different levels on the Tek scope, and they are all clean as a whistle, but you can definately hear it when it's at 0db, so it's not bad enough for the scope to pick up, but the ears can, and all 6 were set at the same listening level, anyone got some ideas?
Cheers George
Cheers George
The transport is a Teac VRDS-T1 with Tent XO3, using the Tent spidf and powered by the Tent XO supply.
How do you check a transport using a computer?
The overloading that I hear at 0db level with the 1k sine wave is a very slight overtone sound sitting on the 1k sinewave sound only at 0db it's not there and the minus -10,-20,-40 -60db levels.
Cheers George
How do you check a transport using a computer?
The overloading that I hear at 0db level with the 1k sine wave is a very slight overtone sound sitting on the 1k sinewave sound only at 0db it's not there and the minus -10,-20,-40 -60db levels.
Cheers George
The DAC itself will/should never 'overload'. The designer knows EXACTLY what the input range is and should have designed it to cope with every single level between zero and full scale.
If something in the digital chain cannot cope with the full scale range of values the designer should be dragged out into the street, pointed and laughed at and promptly shot. There is no excuse for it.
If you've simply bought yourself an external DAC that puts out an analogue voltage too large for your amp then thats a whole different thing.
If something in the digital chain cannot cope with the full scale range of values the designer should be dragged out into the street, pointed and laughed at and promptly shot. There is no excuse for it.
If you've simply bought yourself an external DAC that puts out an analogue voltage too large for your amp then thats a whole different thing.
BlackCatSound said:
If you've simply bought yourself an external DAC that puts out an analogue voltage too large for your amp then thats a whole different thing.
As said, it happens at the same played level, so it's not the amp or speakers or anything after the dac, and it cannot be seen on the scope, only very slightly detected by ear at 0db digital, I wonder how many others have this problem and don't know it?
The only way to find out is to play the different db levels at the same volume either through the system or good headphones at the same analog level.
Cheers George
Hi George,
Run that into your sound card and look at an FFT at 0 dB. It is probably in the analog stage. My normal suggestion would be that your muting transistors are breaking down, but your schematic doesn't show any.
I actually use the 0 dB tone to check muting transistors in CD players and DACs. I've caught a lot of these problems by doing that.
-Chris
Run that into your sound card and look at an FFT at 0 dB. It is probably in the analog stage. My normal suggestion would be that your muting transistors are breaking down, but your schematic doesn't show any.
I actually use the 0 dB tone to check muting transistors in CD players and DACs. I've caught a lot of these problems by doing that.
-Chris
georgehifi said:
As said, it happens at the same played level, so it's not the amp or speakers or anything after the dac
Same digital level or same analogue level?
anatech said:Hi George,
Run that into your sound card and look at an FFT at 0 dB. It is probably in the analog stage. My normal suggestion would be that your muting transistors are breaking down, but your schematic doesn't show any.
I actually use the 0 dB tone to check muting transistors in CD players and DACs. I've caught a lot of these problems by doing that.
-Chris
Good try Chris , but no, if you have a look at the muting circuit in my last circuit, it's a substantial relay, controled from the dac itself.
Yes, loud cd's can overload tha dac !
I just stumbled on this thread, and i have read an interesting article showing that 'loud' cd's might overload the dac.
Remember that even though the sampled value cannot exceed 0 dB FS, the recontructed analogue signal might.
Direct link to the article (pdf-format): 0 dBFS+ Levels in Digital Mastering
zip-file with sound samples to illustrate the findings: Programmed for Distortion
Many other interesting articles can be found in the tech library of tc-electronics: http://www.tcelectronic.com/TechLibrary
How I wish all sound engineers had read these articles
Cheers,
Niels
I just stumbled on this thread, and i have read an interesting article showing that 'loud' cd's might overload the dac.
Remember that even though the sampled value cannot exceed 0 dB FS, the recontructed analogue signal might.
Direct link to the article (pdf-format): 0 dBFS+ Levels in Digital Mastering
zip-file with sound samples to illustrate the findings: Programmed for Distortion
Many other interesting articles can be found in the tech library of tc-electronics: http://www.tcelectronic.com/TechLibrary
How I wish all sound engineers had read these articles
Cheers,
Niels
Interesting (but not for the faint hearted)read Niels, so it seems that 0db signals that go into a dac chip can actually come out at +6db level and clip the inbuilt filters of moden upsampled chips, where the older non upsamlped 16 bit dac chips that have outboard analog filters are less likely to clip, is this how I read it?
Can the digital level as a whole be reduced 6db before it enters the dac chip to fix this problem?
Cheers George
Can the digital level as a whole be reduced 6db before it enters the dac chip to fix this problem?
Cheers George
Hi George,
I think you read the same from the article as me.
I don't know how to manipulate the bitstream that enters the dac in order to reduce the level by 6 dB, but it probably could be done.
To test the effect you could try this cumbersome method (requires a computer with cd-burner): grab the data from an audio cd to wav-files. Use a good wave-editor to attenuate the signal 6dB, and create an audio cd from the edited wav-files.
Of course this method only works on a cd that is loud but not clipped.
Note that you are actually loosing the information in the least bit (LSB), but the result may still be better sounding than the source cd .
And, yes, I have tried this
Cheers, Niels
I think you read the same from the article as me.
I don't know how to manipulate the bitstream that enters the dac in order to reduce the level by 6 dB, but it probably could be done.
To test the effect you could try this cumbersome method (requires a computer with cd-burner): grab the data from an audio cd to wav-files. Use a good wave-editor to attenuate the signal 6dB, and create an audio cd from the edited wav-files.
Of course this method only works on a cd that is loud but not clipped.
Note that you are actually loosing the information in the least bit (LSB), but the result may still be better sounding than the source cd .
And, yes, I have tried this
Cheers, Niels
Hi George,
No, I didn't miss that you used a relay. I mentioned that if you re-read my post. My suggestion was to point you to the analog section and use a scope, or better yet and THD meter and 'scope the residual. If you can use a sound card and run FFT as you go back in stages it would be easier to see.
-Chris
No, I didn't miss that you used a relay. I mentioned that if you re-read my post. My suggestion was to point you to the analog section and use a scope, or better yet and THD meter and 'scope the residual. If you can use a sound card and run FFT as you go back in stages it would be easier to see.
I hadn't thought of that, but of course this might be possible. Interesting as you would think the lab rats would have modeled this.so it seems that 0db signals that go into a dac chip can actually come out at +6db level and clip the inbuilt filters of moden upsampled chips
-Chris
I wish I had a disortion analyser so then at least I could see it, I think that the links that Neils posted
http://www.tcelectronic.com/media/nielsen_lund_2000_0dbfs_le.pdf
could be right on the money, as these guys, (T.C. Electronic) http://www.tcelectronic.com/TechLibrary
build all the gear in the recording stages, and if they are saying it's now a problem with the new upsampled dac chips with inbuilt filters, what can be done about it
(the digital input has to be lowered),
if you read some of their remarks it's only stated to be a problem after 1998 as before that the digital recordings weren't that loud, this is another interesting read
http://www.tcelectronic.com/media/nielsen_lund_2003_overload.pdf
on how it got louder and louder but the dac chips with their inbuilt filter have less overhead and can clip at 0db.
Cheers George
http://www.tcelectronic.com/media/nielsen_lund_2000_0dbfs_le.pdf
could be right on the money, as these guys, (T.C. Electronic) http://www.tcelectronic.com/TechLibrary
build all the gear in the recording stages, and if they are saying it's now a problem with the new upsampled dac chips with inbuilt filters, what can be done about it
(the digital input has to be lowered),
if you read some of their remarks it's only stated to be a problem after 1998 as before that the digital recordings weren't that loud, this is another interesting read
http://www.tcelectronic.com/media/nielsen_lund_2003_overload.pdf
on how it got louder and louder but the dac chips with their inbuilt filter have less overhead and can clip at 0db.
Cheers George
Hi George,
Do you have a sound card in your computer?
If so, download and install http://audio.rightmark.org/index_new.shtml . There are also some programs that turn your sound card into an oscillator, or an oscilloscope. For the measuring functions you may need to provide your own signal conditioning / attenuator.
-Chris
Do you have a sound card in your computer?
If so, download and install http://audio.rightmark.org/index_new.shtml . There are also some programs that turn your sound card into an oscillator, or an oscilloscope. For the measuring functions you may need to provide your own signal conditioning / attenuator.
-Chris
rdf said:When we were setting up our first all-digital domain production suite a CD certain to have a high average level - something by Slipknot - caused panic when the board meters gave all indications of input overload. They locked at one value and didn't budge. The sound was awful, from an analogue mindset it gave every indication of clipping. Checked the analogue output of the digital console: yep, clipped waveform. The CD player fed the board AES-EBU, so furiously re-read the console manual. Nothing wrong. Checked the analogue output of the CD player: clipped waveforms. Second player, same thing. Eventually we came to realize the CD was mastered clipped. This wasn't a clipper/processor, this was straight-out back-to-back zener consistently shear the tops off dead flat. Welcome to the new artistic sensibilites.
Oh my god, we're lost !
Unfortunately Chris I have only onboard sound and it doesn't have a digital input.
This overloading seems to be wide spread, reading the T.C. Electronic articals has made me realize that todays recordings and modern dac chips have a major problem and the only way to fix it as I see it, is to lower the overhaul recording level, or to lower it before it gets to the dac chips.
I have no experience in this, and I wish some of the digital heads on this forum would step in and give some of their knowledge on how to get around this problem, as it seems to be a major fault in the modern day digital chain that has been kept quite or not realized until now.
Cheers George
This overloading seems to be wide spread, reading the T.C. Electronic articals has made me realize that todays recordings and modern dac chips have a major problem and the only way to fix it as I see it, is to lower the overhaul recording level, or to lower it before it gets to the dac chips.
I have no experience in this, and I wish some of the digital heads on this forum would step in and give some of their knowledge on how to get around this problem, as it seems to be a major fault in the modern day digital chain that has been kept quite or not realized until now.
Cheers George
rfbrw said:Anyone with half a brain will set the clip indicator to indicate clipping at some point before 0dBFS leaving a safety margin.
I've look at the data sheets of my Burr Brown convertor and also of AD convertors and nowhere can I find the spec of the overload point, whether it's at +10db, +6db or 0db. Wouldn't it be nice if a d/a manufcturer stated that their d/a chip has an overload margine of +20db above 0db
It would be nice if this spec was given with all d/a chips so we know, we all assume it must be fine at 0db, but what happens at +6db!!!!!!
Cheers George
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