| georgehifi |
I have a very loud cd and I'm sure it's either overloading the output of the Tent XO3 on the transport or overloading the input of my da converter. Can any of you digital heads give any comment on this? (Jocko your the input output king)?
Cheers George |
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| davidsrsb |
CDs data cannot exceed the range 0000 to FFFF.
Many newer CDs are heavily compressed and have many samples at the limit values.
Some DAC digital filters cannot cope with signals too close to full volume as they produce interpolated samples beyond the limits.
Some badly designed filters even go unstable when thet get internal overflows in the maths.
You could rip the cd to a PC and using software like goldwave or audacity reduce the volume by 3dB say and burn a CDR with the quieter version to try. |
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| rfbrw |
| quote: | Originally posted by georgehifi
I have a very loud cd and I'm sure it's either overloading the output of the Tent XO3 on the transport or overloading the input of my da converter. Can any of you digital heads give any comment on this? (Jocko your the input output king)?
Cheers George |
The value of the sample in the digital domain has no bearing on the amplitude of its physical representation. |
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| Elso Kwak |
| The digital filter from Sony; CXD1244 clips on the ringing of a 0dB square wave....... |
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| Samuel Jayaraj |
| I cannot recall the month and year of publication, but Elektor had published a clipping circuit which indicated that many CDs produced exceeded the input limit of DACs. |
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| I_Forgot |
| quote: | Originally posted by Elso Kwak
The digital filter from Sony; CXD1244 clips on the ringing of a 0dB square wave....... |
Your ears would, too, if you listened to that sort of thing. Fortunately there aren't that many 0 dB square waves in music.
I_F |
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| phase_accurate |
| quote: | | Fortunately there aren't that many 0 dB square waves in music. |
No - but many that hit 0000 and FFFF quite often.
I even know one that is regarded as audiophile by some demonstarators !! :eek:
Regards
Charles |
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| jean-paul |
| quote: | Originally posted by I_Forgot
Your ears would, too, if you listened to that sort of thing. Fortunately there aren't that many 0 dB square waves in music.
I_F |
:nod: I am glad musicians do not create such sounds. |
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| Nordic |
| quote: | Originally posted by jean-paul
:nod: I am glad musicians do not create such sounds. |
Come on, have you heard Brittney sing? |
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| davidsrsb |
| quote: | Originally posted by Samuel Jayaraj
I cannot recall the month and year of publication, but Elektor had published a clipping circuit which indicated that many CDs produced exceeded the input limit of DACs. |
I have never seen a DAC chip that cannot handle all posssible 16 bit data codes. If there is sucha thing it is seriously flawed.
There are some "audiophile" commercial CD players with outputs much higher than the red book standard 2V rms. Magazine reviewers always seem to fall for "louder is better"
A preamplifier with +5V powered cmos analogue switches is going to clip at these high levels. Some are high enough to overload switches running on +/- 5V |
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| rdf |
| 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. |
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| georgehifi |
Let me shed a bit more light on this, the cd that i'm talking about is the new Jackson Brown (The Naked Ride Home) it is by far the loudest cd in the 1000 odd that I have.
Now it seems that it only sounds distorted on my separate transport (Tent XO clocked) and separate D/A converter A3.24 Musical Fidelity. For some reason the same cd does not sound this distorted on stand alone players, this led me to belive that the sending/recieving components of separate transport/ da converters may have a limit to what they can handle compared to stand alone players.
Cheers George |
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| rfbrw |
| The tx/rx section is digital. It's levels are fixed and cannot vary in response to the value of the transmitted data. You need to look to the I/V stage and beyond. |
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| davidsrsb |
| quote: | Originally posted by georgehifi
Let me shed a bit more light on this, the cd that i'm talking about is the new Jackson Brown (The Naked Ride Home) it is by far the loudest cd in the 1000 odd that I have.
Cheers George |
Has this got copy protection? This can cause excessive jitter which may make your DAC lose lock
It could also be transport read errors, try EAC on a PC and see if it complains about reading tyhe disk. |
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| georgehifi |
It's not one of those Sony copy protected cd's they sound ugly as well, this this a Warner/Elektra disc but it does have a strange imprint in the centre that I've never seen (D.A.T.A. IFPL 311) whatever that means.
How can 1 cd stand out as twice the output level as anything else I have. I'd understand if it were Deep Purple, or something like that, but Jackson Browne, give me a break.
Cheers George |
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| Netlist |
You would be surprised of what modern industry is capable off. I wouldn’t blame the artists. Try importing some tracks from different cd's in a waveform viewer like Cooledit or Audition. Far too many are compressed and hard limited and make all the dynamics disappear. Look at post #7 here which is not at all an exception these days. It's a big shame.
/Hugo |
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| georgehifi |
An interesting thing is that this disc is the only one that will be read by a cd player that I have here that has a very very weak laser, it will not read the TOC never mind play any of the discs I have except for this very loud one. What do you guys make of that.
Cheers George |
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| danb1974 |
Probably good reflective layer.
Pits and lands on the cd don't look different (bigger) if the encoded audio is "louder". |
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| davidsrsb |
| Too much signal from the read diode array could cause the servo to misbehave, causing a high read error rate |
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| Nordic |
I have a cd called Zucchero and Co. The original plays just fine in my pc, but on the DVD player I use to play my CD's with the main amp, some tracks goes loud and soft alternatively (almost unlistenably bad), after reading the why does copied CD's thread, I tried it, and it solved the problem.
Stream tracks live |
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| georgehifi |
Nordic, what did you try? that thread is 11 pages long, what did you try that fixed your problem, burn a copy from the original? Did that fix the problem?
Cheers George |
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| Nordic |
I ripped the audio with EAC, and then used it to burn the CD.
It came out as one long track like an LP...
Still want to play around with it and see if is better with using the auto leveling feature or without. And to see if tracks can be made seperate from each other. |
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| Netlist |
This is a 32bit recording of the third song. ‘Blue with Sheryl Crow’.
How could you possibly expect this is going to sound right? It’s clipping all over and compressed to the absolute limit.
I can’t believe that a copy will fix that problem.
/Hugo |
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| Netlist |
Note that the 100% limit is not even reached.
Here are a few samples of the many that are clipped.
/Hugo |
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| davidsrsb |
| quote: | Originally posted by Netlist
This is a 32bit recording of the third song. ‘Blue with Sheryl Crow’.
How could you possibly expect this is going to sound right? It’s clipping all over and compressed to the absolute limit.
I can’t believe that a copy will fix that problem.
/Hugo |
Ouch - Average rms power -4.41dB.
This tracks gone through a fuzz box. |
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| Nordic |
Take my word for it, after listening to the original for a while the nomalized copy sounds alot more pleasing.
But theanks for the input, this shows exactly why the CD behaves the way it does. When it does play corrected it is very good though. |
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| Netlist |
It would be nice if you could mail me a piece of the original and the copied audio. Say five minutes of each. Track #3 would be nice.
hugorx with the hotmail thing behind.
/Hugo :) |
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| Nordic |
| Soon as I figure out how to do one track at a time... |
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| Netlist |
EAC should do that be default IMO.
Within EAC, select a track and press F5.
When the extraction process started, after about half a minute you press cancel. This will result in a +/-500Kb file which can be played.
/Hugo |
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| Nordic |
| please try it quickly, all I get is a 44kb file, I tried waiting longer too... |
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| Netlist |
Strange. Some settings in EAC could be wrong. Try running the wizard again.
/Hugo |
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| Wombat |
| btw. A recording already hitting 0dB doesn´t change a bit when normalized if you doesn´t give a % value. |
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| Nordic |
I'll reinstall later and see if it restores defaults...
Nevertheless I was quite interested by your findings and downloaded Sigview to look at that track...
VERY WEIRD is all I can say. If you divide the song in 5 parts it seems only the first 3 has the high levels.... (off the chart high)
Strangley enough on my computer it plays just fine...
There seems no correlation between the graph I got and the music, I first thought maybe it has a hard time measuring because of the brutal bass and drum and piano sound, but the sounds are still there in the "low level" parts.
Wombat, I only saw your post now...
I will confirm that I can't see a diff between the original track and the copied one, yet they clearly sound diffirent. |
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| davidsrsb |
| This looks more and more like a DRM schem messed up |
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| davidsrsb |
Another thought George. Is your player HDCD compatible?
HDCD has a Peak Extend option, if a HDCD player mistakenly tries to expand a disk without the peak compression, you will get the overload distortion |
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| georgehifi |
No, has no HDCD.
Cheers George |
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| georgehifi |
Ok I managed to get a couple of test discs with varying degrees of 1001hz test tones and as I thought all set at the same volume:
-60db clean
-40db clean
-20db clean
-15db clean
-10db clean
0db has a slight edgy ring to the tone.
This was the Pierre Varny test dics, and it states if you can hear a change at the 0db level your da convertor chip is overloading, so can anyone give some solution to lowering the overhaul level so I don't overload my dac?, it's the Burr Brown PCM1738E.
Cheers George |
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| georgehifi |
| That didn't work out too good here are two halves. |
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| georgehifi |
| And the other half |
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| georgehifi |
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 |
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| peufeu |
What transport do you use ?
Have you checked it for bit correctness using a computer soundcard digital input ? |
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| georgehifi |
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 |
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| BlackCatSound |
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. |
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| georgehifi |
| quote: | Originally posted by BlackCatSound
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 |
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| anatech |
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 |
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| BlackCatSound |
| quote: | Originally posted by georgehifi
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? |
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| georgehifi |
| quote: | Originally posted by anatech
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. |
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| nilaus |
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 |
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| georgehifi |
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 |
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| nilaus |
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 :cool:
Cheers, Niels |
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| anatech |
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.
| quote: | | 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 |
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.
-Chris |
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| georgehifi |
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/n...00_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/n...03_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 |
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| anatech |
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 |
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| -_nando-_ |
| quote: | Originally posted by rdf
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 ! :dead: |
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| rfbrw |
| Anyone with half a brain will set the clip indicator to indicate clipping at some point before 0dBFS leaving a safety margin. |
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| georgehifi |
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 |
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| anatech |
Hi George,
My understanding from some of the guys that did mastering was that they did leave some headroom. They can't speak for everyone.
Time to get an old DAC or CDP for those oldies.
-Chris |
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| georgehifi |
| quote: | Originally posted by rfbrw
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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| georgehifi |
I've just listened again on my Martin Logan ESL's and they give a very pure undistorted sine wave sound at -10db or -5db digital, no matter how loud I play it (to almost ear splitting level).
Then at 0db digital no matter how quite I set the volume the 1k sine wave has a slight ringing overtone to it that should not be there. It is obvious to me that the Burr Brown PCM1738 d/a chip is just, and I mean just, going into overload.
I then did the same on a pair of Stenhiser non-electrostatic headphones and the same thing is heard although not as cleanly, but it's there.
Cheers George |
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| anatech |
Hi George,
With digital electronics, when you run out of 1's or 0's, that's called clipping. Sorry to sound glib, but that's the truth of it. They would have to add a couple more positions at the MSB end of the word just to accomodate this. The true problem as I see it are the results of the math with the filter. You are going to get these effects, therefore you would need a longer word, but at the MSB end as I mentioned.
Of course they could always shift everything sideways so we lose a couple bits of resolution. Guess which way they would go.
-Chris |
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| georgehifi |
| quote: | Originally posted by anatech
Hi George,
Of course they could always shift everything sideways so we lose a couple bits of resolution. Guess which way they would go.
-Chris |
So correct me if I'm wrong but most of us are using 24bit dacs these days (except for the TDA crew) couldn't we afford to loose a couple of bits off the 24bits and still have 16bit resolution? There must be a solution to this inherent problem with the new world d/a chips.
Or must we resign to the fact that when a digital stream hits 0db we are going to hear slight distortion, and that we should live with it.
I say not and we must revolt against it, in the most loudest way possible
Cheers George |
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| rfbrw |
| quote: | Originally posted by georgehifi
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 |
It is obvious where a digital device overloads. A 16 bit digital audio device overloads at 0111 1111 1111 1111 for positive going values and for negative values it is 1000 0000 0000 0000. It has nothing to do with the manufacturer.The overload point is known to all. That others choose not to take this into account is not the fault of the manufacturer. |
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| rfbrw |
| quote: | Originally posted by georgehifi
So correct me if I'm wrong but most of us are using 24bit dacs these days (except for the TDA crew) couldn't we afford to loose a couple of bits off the 24bits and still have 16bit resolution? There must be a solution to this inherent problem with the new world d/a chips.
Or must we resign to the fact that when a digital stream hits 0db we are going to hear slight distortion, and that we should live with it.
I say not and we must revolt against it, in the most loudest way possible
Cheers George |
There is no problem as far as the D/A is concerned. It simply reproduces what it is sent. If the dac hits 0dBFS it is because the data sent is at 0dBFS. |
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| georgehifi |
Yes but what they're saying is the 0db going into the d/a chip can become +6db http://www.tcelectronic.com/media/n...00_0dbfs_le.pdf
when converted to analog before it comes out and distorts the internal analog filters of the d/a chip.
So if it's happening inside the d/a chip then it is the responsiblity of the d/a chip manufacters to have their internal filters so they don't overload. It has to be their problem and not the recording industries because they are playing by the rules and not giving any more than 0db to the dac.
Cheers George |
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| rfbrw |
| quote: | Originally posted by georgehifi
Yes but what they're saying is the 0db going into the d/a chip can become +6db http://www.tcelectronic.com/media/n...00_0dbfs_le.pdf
when converted to analog before it comes out and distorts the internal analog filters of the d/a chip.
So if it's happening inside the d/a chip then it is the responsiblity of the d/a chip manufacters to have their internal filters so they don't overload. It has to be their problem and not the recording industries because they are playing by the rules and not giving any more than 0db to the dac.
Cheers George |
A D/A converter does not have internal analogue filters, that is an external circuit.
The paper refers to intersample peaks. If you are inept enough to set your recording levels so high that the samples either side of a peak are at 0dBFS, then it stands to reason that the D/A will hard clip when it attempts recreate the peak as it simply cannot exceed 0dBFS. It is a simple matter to set your levels to avoid this. This may be new to TC Electronics but it is well known in the video world. At the post-production facility where I worked we routinely set our A/D's to indicate clip at 18dB below 0dBFS. The paper refers to some engineers opting for 3dB below 0dBFS (see section 5.2).
A lot of that paper reads as a case of giving a dog a bad name and kicking it. If you are hard clipping heavily enough to produces square waves and all that that entails, and not notice it, you must be deaf. All of the sources of distortion given can be avoided simply by being competent. |
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| georgehifi |
rfbrw, I said, and a few times now that I hear a SLIGHT overtone at 0db digital, I said nothing about hard clipping, this SLIGHT overtone that can be heard is not great enough to be picked up by a good scope, yet it can JUST be heard, and is not present at lower than 0db digital levels no matter how loud it is played.
I'm tending to believe the guys who have vast experience in building and designing the T.C. Electronic equiptment.
I hope others who have test cd's in their kit with varying levels of 1k sine waves up to 0db can try this, because you would have to be deaf not to hear it.
Cheers George
And here's a block diagram of my D/A chip, I'm sure it says digital filter in there. |
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| georgehifi |
And yes rfbrw before you jump on me it is a digital filter and I said analog, slip of the tongue.
What is more interesting is the conclusion to the T.C. Electronic A.E.S. report posted below.
Cheers George |
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| Werner |
"All of the sources of distortion given can be avoided simply by being competent."
Delete the 'simply'.
It is a sad fact today that many productions hit 0dBFS routinely,
and that even a lot of productions just slam through it, clipping hard for tens of consecutive samples.
The most gruesome I have encountered so far (I have not been looking) is Depeche Mode's The Singles 81-85, which looks and sounds like a chainsaw.
Another interesting case is Simple Minds' New Gold Dream. Having the LP, I got the CD as a kid when it was released in 1983 or whatever (and then had to wait another 6 years to play it ;-).
Much as the LP peaked once massively close to the end of side two, this CD too hit 0dBFS once, in the same spot.
Last month I obtained a recent pressing of the same album. It sounds louder, brighter, and each track now hits 0dBFS 100-200 times.
So 'simply' and 'competently' are not exactly verbs in the music production industry. |
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| georgehifi |
WernerI have a cd in my collection that has to be turned up twice as loud as the average cd has to (about 3o;clock), it called the "Tony Dagcadi Trio" it is the cleanest most explosive "dynamicly" cd in my collection yet it is the quietest volume setting wise, at twice the volume setting for normal listening level.
Have we become obsessed with having volume setting at 7o'clock for ear splitting level and the distortion they produce because they hit the 0db point, and by the way this is with a passive preamp, god knows what the poor souls do with 50-100 times gain active preamps, they must be just of the zero bump stop on their volume controls.
Cheers George |
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| rfbrw |
| As I said it is all about competence. Note the d/a converter used by Sony. |
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| georgehifi |
| quote: | Originally posted by rfbrw
As I said it is all about competence. Note the d/a converter used by Sony. |
rfbrw Then if the Sony 777 has the same d/a chip as my MF Dac the PCM1738, then the problem has to be after the d/a chip, in either the I/V opamp which I've changed from the NE5532 to 4 x AD825 which should be far better. Also I've done a dc coupling mod with offset adjustment 5k mulit turn pots.
Can anyone see what could be limiting in this I/V or output stage at 0db digital, remembering it cannot be seen on the scope.
Cheers George |
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| anatech |
Hi George,
I can't read the resistor values to figure out how much gain that circuit has. For it to clip, you'd be putting out over 20 Vp-p. The darlington output transistors might be a bit slow, but I doubt they would cause your troubles. You need a 'scope right about now.
I do have a test CD that outputs 0 dB levels. I'll have to check a new CD player next time I see one. I have a feeling that your troubles are in the digital domain still.
-Chris |
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| peufeu |
| Can you plug an oscilloscope in the analog output of the CD player to see what's actually there ? |
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| georgehifi |
| quote: | Originally posted by anatech
Hi George,
I can't read the resistor values to figure out how much gain that circuit has. For it to clip, you'd be putting out over 20 Vp-p. The darlington output transistors might be a bit slow, but I doubt they would cause your troubles. You need a 'scope right about now.
I do have a test CD that outputs 0 dB levels. I'll have to check a new CD player next time I see one. I have a feeling that your troubles are in the digital domain still.
-Chris |
I managed a bit better rez on this one, try it, thanks Chris.
Cheers George |
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| rfbrw |
| You are assuming the 0db track was not recorded already clipped, however slight. Then again, the opamps may be driving the differential pair into slight clip. |
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| georgehifi |
| quote: | Originally posted by rfbrw
You are assuming the 0db track was not recorded already clipped, however slight. Then again, the opamps may be driving the differential pair into slight clip. |
rfbrw The test discs i have and both do it are the Denon Audio Technical (not for sale to the public). and the Pier Varny classic from years gone by.
And Chris as I said in a previous post I've looked at the output loaded with 22k with a Tektronics scope and the 1k looks perfect as if it was my Teck generator producing the 1k sine wave and not the transport/dac combo, and it looks perfect whether it's -20 -10 or 0db tracks on both the test discs, but it definately can just be heard at the 0db level.
I'm at a wits end with this, I hope someone can see a problem with headroom in the I/V or output stage as that would be a better fix for me as I'm no good with the digital side of things.
Cheers George |
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| georgehifi |
Here is a pic of the front end of my dac, just in case you guys can see anything wrong there.
Cheers George |
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| georgehifi |
While I'm at it here the power supply to.
Cheers George |
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| georgehifi |
Just found the Piere Varnay Lab Discs have 3 x 1k sinewave tracks at 0db +3db and +6db all with de-emphisis off here are the pics, remember that the slight distortion can just be heard but not seen at 0db but boy can you hear it at +3db and +6 db
Cheers George
0db |
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| georgehifi |
| and the third at +6db |
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| georgehifi |
Now with the aid of the scope and the +3db I should be able to see it at the different points, after the D/A then after the I/V and finaly the output to see which one goes into cliping.
My aim is that (hopefully it's not the D/A) is to get the +3db as clean as the 0db at least that should stop the slight over tone at 0db.
Any one got any other ideas?
Cheers George |
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| mrshow4u |
| ....These last scope shots show ADC overload. ...and the performance is good. No wrap-around etc. The DAC get data only. The only valid data is 1 or 0. 16-bit (or 18-bit, 20-bit, 24-bit, etc) is serially fed to the DAC (most cases) and latched in. Think of the DAC as a fast, programmable power supply. Based on the input word value, a programmed output current is generated. You can only feed a min value 0x000000...... to 0xFFFFFFFF..... The DAC cannot latch in a larger word size, nor can it accept other than binary values. You can overload Digital Acquisition. Hopefully it will have smooth overload performance. You cannot overload a DAC. |
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| georgehifi |
mrshow4u, you kind of lost me a bit there, what your saying I believe is that the clipping is the D/A converter itself and that it looks good to you?
Then why if you can explain simply to a digital pleb like me, can I hear a slight distortion on the esl's speakers at 0db digital but not at -3db digital or -10db digital all being played back at the same analog volume?
It seems to me that the scope cannot pick up the slight distortion that I hear, but my ears can at 0db digital.
Cheers George
Here are the notes of those three tracks |
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| mrshow4u |
| quote: | | what your saying I believe is that the clipping is the D/A converter itself and that it looks good to you? |
Sort of. The current that the DAC is putting out is programmed, but not distorted. Yes, the waveform is a distorted audio waveform, but the DAC is not clipping. In digital, you can program a word value between all bits off to all bits on. Say you have a scale from zero to one. The range is divided up this evenly by the word length (16-bit, etc) for linear PCM. So that flat-top overload that you see will happen on every DAC. That's what the data is "telling" the DAC to output. That flat-topped waveform that you see was either: generated at the conversion Analog overloading a ADC, or the overload occured in DSP through addition or multiplication of non-overloading data. If your finding overload occuring and dependent on the DAC circuit, it could be from the I/V converter with too much gain or too small rail voltage.
......I just saw the added info about 0dB FS with audible distortion but not -3dB FS.. Make sure when making this comparison, that the volume control is knocked down 3 dB when playing the -0dB FS track. Maybe the panel is distorting a little bit with the increased level? Kepping the same acoustic level will determine that. |
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| georgehifi |
Yes the same acoustic level was carfully matched.
I posted the circuit back one page of the DAC.
mrshow4u, I'm using these AD825 http://www.analog.com/UploadedFiles...1852AD825_f.pdf
for the I/V with 15+ and 15- they are fet input, the originals were NE5532's and they are bi-polar input, do you think the problem lays there? As the Ad825's sound so much better than the NE5532's and the specs kill them also.
Cheers George |
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| clem_o |
Hi!
Why not settle the controversy by sticking a distortion analyzer at the output of the CD player? It's extermely difficult to see "slight" distortion of sine waves.
You may be better off digitally synthesizing a triangle wave that peaks at the maximum possible digital values and looking at that on the tek scope - that "may" be more visible. I can easily syth that with a program and send it to you as a wav file if you wish - then just burn it into a cd-r.
Could the 0dBFS be overloading something else in the playback chain?
Cheers!
Clem |
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| georgehifi |
I wish I could afford a distortion analyzer, and yes the slight distortion is still there with headphone into the dac as well.
I hope someone can see aproblem by me using the Fet input AD825's instead of the Ne5532's as the I/V opamps
Cheers George |
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| mrshow4u |
Hi Georgehifi, Does the problem lie in the NE5532??? Hmmm. Well I can't say. I looked at the -0 dB FS (full-scale) and did not see overload. It "looks" good. If you play the -0 dB FS 1 kHz tone and turn down the volume a little, does it still sound distorted? From the scope shot, it looks all good. Are you scoping at the line out? I think the distortion you might be hearing may not be coming from the DAC. ...just a guess. You can check the P-to-P voltage (at 0 dB FS) and see if the voltage swing is a couple of volts away from the rail voltage. The AD devices should saturate pretty close to the rail, but safety margin is always good. You might want totry with good headphones (with a volume control!!) and if you hear the distortion at all play level, reverse the headphones (L>R, R>L) and see if the distortion is really in the content.
Really, All DACs and I/V converters will/must show clipping above -0 dB FS. The distortion is in the data. |
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| georgehifi |
| quote: | Originally posted by mrshow4u
Hi Georgehifi,
If you play the -0 dB FS 1 kHz tone and turn down the volume a little, does it still sound distorted?
From the scope shot, it looks all good. Are you scoping at the line out?
You might want to try with good headphones (with a volume control!!) and if you hear the distortion at all play level, reverse the headphones (L>R, R>L) and see if the distortion is really in the content.
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The slight distortion is there at all levels, even a whisper when playing the 0db digital, and not there at all levels at -3db digital.
Yes scoping at the line out with 22k load, also listening from the line out with headphones, and it is equally there in both channels so reversing does nix
Cheers George |
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| clem_o |
Up the supply voltages to +-20V and see if it clears up!! :-)
The 5532s are fine with +-20V. I'd guess that the discrete output stages will be as well. |
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| mrshow4u |
| Clem_o's idea is good. You can lower the Vref voltage on your DAC to and see if that reduces the distortion. |
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| clem_o |
Could the DAC be 'running out of steam'? I.e. the capacitive loading of the i-v converter, which combines the function of initial LPF...
Cheers
(though this seems perfectly acceptable as per Burr-brown's app circuit)... |
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| anatech |
Hi Clem,
You suggested a very good idea with the triangular waveform. One we normally reach for on the bench for checking linearity. Good application. An FFT would show the slightest issue with a sine wave. An alternative would be to run the DAC output into a distortion analyzer. Connect your scope on the residual output and look at that.
Hi mrshow4u,
I agree with your points. The D / A can not clip, but it can be overloaded on it's output. Shouldn't be too much of a problem on a current output D/ A convertor. Is that ref pin accessible? What happens if it's loaded down?
Hi George,
I normally stay with the same family when changing op amps. So to replace a NE5532, stay with another bipolar type. Op amps are optimized for specific applications. There is no one "best" device for that reason. The NE5532 isn't that bad a chip either.
-Chris |
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| clem_o |
Hi Chris,
Thanks - I remember my brother telling me about the triangle wave and it makes sense...
Wouldn't a computer with FFT be able to pick up the distortion georgehifi is hearing? To be able to hear overtones would mean significant harmonics well visible above even a mediocre ADC in a sound card I think - specially at 1KHz..
But can a DAC run out of current? In what would appear to be the most 'logical' topology, where various resistors are used to produce the current, no, but that's not how DACs are built anyway, given it's not easy to balance currents over that wide a span.
Cheers!
Clem |
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| anatech |
Hi Clem,| quote: | | Wouldn't a computer with FFT be able to pick up the distortion georgehifi is hearing? |
Yes. It would stand out like a sore thumb. Unfortunately his computer has no line in jack.
George, it may be labeled as a mic jack. Please have another look. Soundcards are not expensive either. The Soundblaster Live! 24 bit is very good sounding but limited to about 19 KHz. Enough for you. They are around $35 CDN around here.
| quote: | | But can a DAC run out of current? In what would appear to be the most 'logical' topology, where various resistors are used to produce the current, no, but that's not how DACs are built anyway, given it's not easy to balance currents over that wide a span. |
D/A chips are amazing these days. Such high accuracy. Current or voltage out is about the same work. Most people go with current out so they can use whatever op amp they want, or go with a resistor and tube. Whatever they want. On board op amps have not received good reviews.
-Chris |
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| clem_o |
Ok, I have an extra tek distortion analyzer module - designed to plug into one of those tek housings with a programmable power supply. Problem with it the pcb that extends into the housing cracked off, which is why I got it cheap. It still works, I ran it off an external, fixed voltage(s) supply. Anyone want it? You pay for S/H...
Getting to swap it for a dead TDS-210 would make me even happier... (I have a tds210 with a bad LCD)...
Cheers!
ps: It requires something like +5, -12 and +12 if I recall.. Easily achieved from a PC power supply... |
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| georgehifi |
| quote: | Originally posted by clem_o
Could the DAC be 'running out of steam'? I.e. the capacitive loading of the i-v converter, which combines the function of initial LPF...
Cheers
(though this seems perfectly acceptable as per Burr-brown's app circuit)... |
I'm back, soooo much info, i'm getting cranial overload.
Clem_o that thought crossed my mind also, fets are harder to drive because of their input capacitance, I wonder what the output stage of the D/A is like, if it's weedy it may have trouble, in the next couple of days I'll try the NE5532's back in and see what happens, if it works (bummer) there goes my direct coupling as they have 20 x the offset dc of the AD825's. not easy to trim out.
Time to go for a surf and forget about 1's and 0's for a while, and think about 6ft glassy barrels.
Cheers George |
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