Discussion in 'Audio Hardware' started by Phono Groove, Dec 5, 2016.
It doesn't matter what level your amp is set at, you can adjust its volume without affect to your recording. In Audacity, there is a control with the icon of a microphone and a slider - use that to control recording volume. Start recording and look at the waveform. If the waveform is really small, you may increase the recording volume, but do not increase it so much that the peaks are shaved off. The important thing is that you get the full range of dynamics in your recording.
I tried my first needledrop and it came out as a success, I used audacity then cleaned the noise from the track with clickrepair so far so good! One question; howcome the left channel seems to have slightly more volume than the right. Im seeing this in audacity the left always reaches higher than the right!?!
Could be the record, or it could be your setup. Might want to eliminate the record as a factor first.
Good test: if you've got a mono LP, try to record a song from that. If the levels are still off then at least you know that it's a hardware issue of some kind.
After that I would carefully re-check cartridge alignment and antiskate setting. If they are really properly set, could be a problem with the cartridge.
I used to do this, but with a Mac Mini he won't have a CD-Rom drive, and if you're going to buy one of those, you might as well buy a A/D converter instead.
I've routinely over the years recorded at 96khz, then downsampled for Redbook files. At least when Barry Diament was around here, the attitude was to capture as much as possible, then downsample if preferred. So you don't dither, just truncate from 24->16-bit?
You'd be surprised how many lp's have channel imbalance, the mono test is a good one Gas, what I use
I think Audacity has a channel balance one can use, I can't remember...
Not sure how much I'd trust it. There's a normalize function, but if you're handling channels interdependently it will re-amplify each to the same level.
But what if one channel is purposely lower in volume, for example some of those early 60's stereo LP's? Might be better to confirm if the setup has a constant channel imbalance and, if unfixable by hardware, one could at least fix in post-production. If the imbalance is constant, of course.
Yes my understanding is dithering is only for the rate - like if you would go from 96KHz to 44.1 it would need to re-sample/ dither to the new sample rate. There are many algorithms to do this some better than others - remember all the CD recorders and Minidisk recorders always bragged about having a "Sample rate converter" for this.
Bit depth from 24 bit to 16 bit just chops off the least significant bits and truncates. Since the difference between 16 and 24 bit is the space between the 16 bit theoretical minimum and the 24 bit theoretical minimum that means what I lose is almost completely irrelevant (assuming I captured at 24 bit and normalized before the truncation).
It's the same reason that it makes sense to record at 24 bit in a music studio environment when your final file is intended to be a 16 bit CD. If you think about it, with needle dropping you really are creating a two track recording studio in your computer.
Dither is really for the resolution, not the sample rate, though the processes tend to be connected. By adding dither noise, it allows you to encode a little extra resolution from that 24-bit file into the 16-bit file. If you start with a 24-96 file, you don't really need dither when going to 24-44.1, though you would probably want dither if going to 16-44.1 to preserve some of the lost resolution.
Thanks for the clarification. It's all pretty confusing. Since I'm using the Sony Professional Sound Forge / CD Architect software I trust that all these functions are pretty well implemented in my setup. I'm very happy with my needle drop quality.
You must dither when going down from 24bit to 16bit or you will introduce distortion. Dither also has to be the last thing you do in the processing chain.
This is what I do in Audition 3.0: - record 24/96, amplify/normalize as I see fit, save separate tracks after any de-clicking, then for each track resample to 44.1 and dither to 16 bit using iZotope.
If you're going to resample to 44.1 kHz, then record at 88.2 kHz. Since it's an even multiple, there won't be any extra bits floating about.
I use a Sony CD recorder with a CDRW. The only trick was that the CDRW had to be a 'music' cd. The 'non-music' CDRW would not work. I use Sound Forge to rip the CDRW and cut the songs into individual tracks.
I haven't yet found a digital click and pop remover that works well enough for me. They all take out too much good sound for my ears. I just live with the vinyl imperfections in the digital tracks. I figure if I'm going to hear them when I play the record, what's the difference?
That's an early model Spin-Clean........
Reverse the L and R cables (at TT input to amp) and see if the symptom does not go to other channel.
Yeah, it's easy to get it to work, but I've never really been fully satisfied with the results I've gotten. There's just something missing. I'll hear professional quality rips from guys online and then be disappointed with the thin, cold rips I'm able to muster up with my equipment.
In an attempt to get better results, I bought a Tascam recorder, but the tape out signal on my receiver was too hot for the recorder. I tried taming it with some volume attenuators, but that somehow introduced a whole other distortion issue. I finally got frustrated and just sold the damn thing.
I don't understand it - how can a signal from line-out be too hot for a line-in? Unless the Tascam was expecting mic-level signal..?
What model was that?
Beats me. All I know is even after adjusting the recording volume, I couldn't get the signal out of the red.
That Tascam seems to only have a 3.5mm input - maybe it expects to be fed mic-level signal?
I know I mentioned Zoom H1 in an earlier post as an example but I actually meant to say H4. It has two 1/4"/XLR combo inputs that accept both mic-level signal (with the preamps turned on) and line-level (with preamps off, obviously). There are plenty of digital handheld recorders that work like this. That Tascam you had probably isn't one of them; something like the DR-100MKIII on the other hand certainly is.
DR-07MKII manual states:
"When an external audio device with a fixed output line level is connected, it would be impossible to control the
gain level appropriately because of excessively large
input signals. In such cases, use the headphone jack or
other level-controllable output for connection to the unit."
From what I've read on pro audio forums and I think what Barry Diament said here a few years ago, that this shouldn't be a worry (i.e., downsampling 96-->44.1 will not be a problem). "bits" doesn't apply here, as far as I understand, since this is sample rate.
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