Is my interface stupid, or is it just me?

Y

yourbestfriend

New member
I hope someone can help me out with this. I recently did a mixdown of a track from my Tascam 488 to my Tascam DA-30 MKII. I used the preset calibration on the DA-30 MKII and ran through the song a few times before I got the perfect monitor and headphone mix. Everything was super breezy and it sounded great on the final mixdown, so I figured I would try to transfer to my computer to post some MP3s online.

Now, I don't have a very high end interface or digital recording program (I'm using a Behringer UCA202 and Audacity because CHEAP) but I didn't figure I would have this problem. The frequencies are being automatically compressed/limited. Whole frequency range and nuance is just...gone. It's nothing to do with the DAT deck, because the signal sounds fine when I'm actively monitoring FROM THE INTERFACE, but it's obvious that the waveform is being compressed. I can't tell if it's the interface doing it (since the signal sounds fine during monitoring on the UCA202) or if Audacity is just too lame to take the full girth of this track.

Is it my soundcard? What is it? I hope someone out there has an answer for me.
 
yourbestfriend said:
I hope someone can help me out with this. I recently did a mixdown of a track from my Tascam 488 to my Tascam DA-30 MKII. I used the preset calibration on the DA-30 MKII and ran through the song a few times before I got the perfect monitor and headphone mix. Everything was super breezy and it sounded great on the final mixdown, so I figured I would try to transfer to my computer to post some MP3s online.

Now, I don't have a very high end interface or digital recording program (I'm using a Behringer UCA202 and Audacity because CHEAP) but I didn't figure I would have this problem. The frequencies are being automatically compressed/limited. Whole frequency range and nuance is just...gone. It's nothing to do with the DAT deck, because the signal sounds fine when I'm actively monitoring FROM THE INTERFACE, but it's obvious that the waveform is being compressed. I can't tell if it's the interface doing it (since the signal sounds fine during monitoring on the UCA202) or if Audacity is just too lame to take the full girth of this track.

Is it my soundcard? What is it? I hope someone out there has an answer for me.
Click to expand...

I'm not real familiar with Audacity but I assume like most DAWs there are different options for exporting a project. It might just be the way you're saving it. Is the issue after you've saved the file as a mp3 or while monitoring inside of the DAW before exporting? Also mp3 is by nature a compressed format so it's to be expected to an extent. If it's while listening inside Audacity that you're hearing the difference then maybe it has a limiter or something on the master bus by default that you want to remove...like I said I don't know audacity, but I know in many DAWs the default template has some form of compression or limiter on the master bus unless you change it.
 
I haven't even tried exporting to an actual MP3 file yet. this is all happening while monitoring within the DAW. Unfortunately, this is not a real hardware/software DAW. Audacity is a free program and the UCA202 is just an RCA line interface. No balanced line inputs or options for input level or any real interaction between the Audacity software and the UCA202 interface other than Audacity taking the signal from the interface.

I tried changing some default options in Audacity, like changing the bit rate from 44.1 kHz to 48 kHz and expanded the waveform range, but neither of those options changed the compression happening. Maybe I should try to find a guide on Audacity? I know there's a wiki somewhere.

It sucks because I'm loosing a lot of high frequencies, loosing my drums and some guitar parts. Strangely enough, my vocals sound crisp, but those are all high end too. I really would prefer to have an accurate representation of the songs on the internet, but that might mean looking into more hardware I can't afford right now. This is something I'd like to avoid.
 
maybe I should post a screen shot? the same thing's happening with my first run with reaper, tho I haven't adjusted the input level on it yet...because I can't find it. It's a more sophisticated DAW than I'm used to.
 
I'm recording on to a Tascam 488 cassette 8-track recorder. From there I was mixing down to a Tascam DA-30 MKII. Everything sounded fine at that point. The analog outputs on the DA-30 MKII are balanced line for XLR and stereo RCA jacks. I have to use the stereo RCA jacks since that's the only input on the UCA202 USB interface.

Now, the UCA202 fits in my USB port in the back of my tower. Then it's all up to my sound card and DAW to process that signal. I can do line level monitoring from the interface, with headphones and stereo. Everything sounds fine from here! The signal is sounding exactly like the mixdown on the DA-30 MKII. However, it's obvious while watching the signal being captured in the DAW that the waveform is being compressed. There are no peaks are valleys, it's just...a solid block.

The thing is, it's not like normal clipping because adjusting the input level, either from the DAW or from the control panel for the UCA202, does nothing to change this compressed block of sound. On playback it sounds dull, with all extreme frequencies diminished. I've changed the playback rate in both DAWs with no change in what's happening in the recording.

Should I just buy a new interface, or is it my sound card that just can't handle the awesomeness of my recordings?
 
uh...never mind. I just tried mixing down from the Tascam 488 directly to the DAW and adjusting the master volume on the 488...well, there's no more clipping, and I get all my frequencies.

It sucks there's no output volume control on the DA-30 MKII. Looks like I'll just have to have two separate mixes for every single track. Hurray.
 
The UCA 202 is a useful device (had two) but it does, not suprisingly, have its limitations!

The lack of any input controls is a big PITA but is easily overcome if you have some modest soldering skills. What is needed is a 22k+22kOhm log stereo pot in a tin and suitable input and output connectors. Jacks in, RCA out?

This will allow you to keep the level below -12dBfs most of the time and NEVER go above -6dBFS! The box does not overload gracefully! Tho' it is only a 16bit device both of mine had a noise floor of better than -80dBFS so with almost any music source, near CD dynamic range.

Then the 202 needs setting up in Win7 as "not" a microphone i.e. the level needs to be set way down to 2 or 3 (from max).

Note: Audacity does not save as .wav instead "export" as .wav then MP3 it.
Better, download Magix, Samplitude Silver. Proper DAW and a very good built in MP3 encoder.

Dave.
 
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