While tracking or mixing, what happens when you raise the fader over 0 on the channels?
Does it add more 1's and 0's to the bit streem? Does it change the sound other than DB level? Is it something to be avoided? Also, when backing up automation,, and recalling it , at what point should the scene be recalled in memory 00? Do you have to go to the song memory and hit read, then recall the automation? I sometimes do a syx dump of both the scene and automation from 00 so that when i recall it , the scene goes rite back to 00.

I might be wrong on parts of this explanation. If I am, I apologize, anyone is free to correct me.

If you have a signal coming into the DA7 on the analog mic/line input, you typically adjust the MIC/LINE input knob with the fader at 0dB so that you get a max. signal without clipping or distorting (same way you set the gain trim for most consoles). In some cases a certain amount of occasional clipping is tolerable if short enough duration. When you then push the fader above 0dB you start to distort the signal because the fader is adding level to something which is already peaking at 0dB (if you you have adjusted the channel trim correctly).

Same goes for digital tape returns from MDMs. You have recorded a signal to your ADAT and the signal peaks at 0dB, so you have the hottest possible level to tape. When that tape channel comes into the DA7 through lightpipe, you will have the identical level as recorded on your ADAT with the fader at 0dB (you can see this indicated on the meterbridge). If the fader is pushed above 0dB, you will add level to the signal and start to distort the signal.

What happens when you push the fader higher? The audio gets digitally scaled up. The details of the whole operation are probably a lot more complicated than I am explaining, but technically you are correct it does add more 1s and 0s to the bit stream in a way, it makes the value of the bit word higher as long as it does not pass max. value (I don't think it can add bits onto the bit word length).

Each sample has a value either above 0 for the positive half of the waveform or below 0 for the negative half of the waveform. With 16 bit signals for instance there are 65536 total values (2 to the 16th power), the max. positive value is 32767 (0x7fff) and the max. negative value is -32768 (0x8000), these are constant. When a signal is at "hottest" level, the maximum positive half of the waveform does not go above 32767 and the max. negative half of the waveform does not go below -32768.

A -3dB 16 bit digital signal peaks at a certain value below max. On the positive side of the waveform I think it's about 23197 (using the dB = 20log(value/32767) formula). When you add 6dB to the signal using the fader, it pushes that level above the 32767 max. positive limit of 16 bit and since there are no values allowed in 16-bit above 32767, unfortunately is clipped to the max. positive value of 32767. Same thing goes for the negative half of the waveform.

If you look at clipped signals in a waveform editor, it looks like you gave the waveform a crew cut.

Depending on how severe the clipping is, you may or may not hear it. Short instances of 10mS or so may not be noticeable. If the audio is pushed way over the threshold, you will definitely hear it and it will change the sound.

The short of it is, you should monitor each channel on the meter bridge to make sure you are not adding level to it and causing the channel to clip.

I believe scene memory 00 is filled with the current automation file start scene when you recall it. Save the automation files to guarantee each start scene is saved along with automation data.

Once again, sorry if any of this is incorrect. You should go to a book on digital audio for a more complete explanation.

[This message has been edited by gregk (edited 07-10-99).]

BTW, I use mix subgroups 1-8 and I monitor their max. level using the meter bridge. This seems to prevent signal from multiple channels from adding up and clipping at the subgroup stage.

Thanks for the explanation. It was good to be reminded about monitoring post fader , out of habbit I was pre fader. The way I was told by Panasonic tech support was you adjust the mic pre reguardless of where the fader is set, then if you need more signal to tape , turn the fader up to adjust the level on tape. I think the mic pre's work independently of the fader, at least at setting levels coming into the board. If i'm wrong , please let me know..

Jeremy, I think you are right. Make the mic pre work as hard as possible and let the fader stay at 0dB if recording. I do not know why anyone would recommend otherwise since this seems to be the recommendation for almost every console I've seen. If you have adjusted the signal and you need to attenuate it, that's a different story, you could use the fader to scale down but I wouldn't add gain using the fader.

As far as the fader working independently of the mic pre I was under the impression that the A/D conversion happens way before the fader in the signal path. I might be wrong, though.

BTW, once in a while I see a certain kind of clipping that is very short (1 sample long) but sounds very damaging, it is when a wave peaks at the positive or negative 1/2 and for some reason one of the samples at the peak of the wave makes its way all the way to the other extreme just for that sample. Usually makes a very nasty audible click. It can also be fixed, that's the good news. I don't know what causes that kind of clipping though. Perhaps numerical error.

[This message has been edited by gregk (edited 07-10-99).]

Hey Gregk, why wouldn't you use the fader to add gain? Most of the time I need to add gain to tape after the pre is set at a safe level. I would have to print a lower signal to tape not to add gain. That's why I asked the question before because I wasn't sure if it altered the sound doing that, adding harshness...

If your mic pre/trim on the DA7 is set right to the point where it peaks once in a while (as it should) then with the fader set at 0dB you should also see the channel peak at 0dB on the meter bridge (providing you are not subtracting any level from the signal via use of EQ). If you are subtracting level via use of EQ, then you probably have to make up the difference with the fader to get the signal back up to 0dB for a tape feed.

If you mean you have to back off the trim a little because it's distorting otherwise and you find yourself adding a tiny bit of fader to compensate, I don't think that would affect the sound any at all.

As far as not adding gain w/fader I was pretty much talking about severe cases, where you might have the mic trim inappropriately too low and have to push the fader all the way up to make up the difference.

There can possibly be a big difference in sound between a digital level scaling operation adding for instance 10dB to a signal in the digital world vs. just having the optimal analog leve entering into the A/D converter in the first place. Technically I guess the signal to error ratio would go down if you simply scaled a signal up (for instance "normalizing" a signal or boosting a digital fader) instead of having the hottest possible signal into an A/D converter. If the input level into the A/D converter is too low only a portion of the available bit resolution is used. Boosting that low digital signal in the digital world just boosts the quantization noise as well.

[This message has been edited by gregk (edited 07-12-99).]