I am dubbing my animation movie. I create very small wav files, 2-3 words each. My old cheap and simple mic became unusable. It worked with my onboard AC97 card. So I upgraded to a Saffire 6 usb card with an SX-8 mic. I am using the gain control button to make my voice as loud as the card clipping indicator will not turn red.
The problem is that the maximum amplitude that appears in my wave form is not at the edge of the edited wave form truck, as shown in the attached jpg. It happeded with another usb card that I checked (Lexicon). It is more difficult to see if there was a clipping.
With the old and cheap onboard card the amplitude reached the edge and I could see the clipping. Now the clipping is somewhere around the line of the envelope tool. If I miss the red light while recording I cannot see the location of the clipping. If I turn the gain lower my wave has small amplitude. How can I use my new usb card and yet let the wave form reach the edge of the truck as it was with my old equipment?
As I reconstruct this in my head, you stopped using your old microphone and plugged this external soundcard into the computer in place of it. You can’t do that. You are overloading the computer’s audio input – the new soundcard’s signal is much, much louder than the old microphone.
You are intended to connect the soundcard’s USB digital connection to the computer instead of the analog connection.
Alternately, you can connect the soundcard’s analog outputs (the RCA connectors) to the computer using the computer’s Stereo Line-In connection – if you have one. Windows Laptops tend not to feature a Stereo Line-In. You didn’t tell us what kind of computer you had.
Hi,
Thanks for replying.
I have read your posts. I did not explain myself good enough.
My old setup that worked very good:
Using the on board AC97 sound card with a simple mic.
Operating System: Windows XP SP3 Professional
Computer:Gigabyte P4 Titan 845PE with Pentium4 2.4GHz.
Attached (with this post) is the good waveform of the above setup
My new setup that has a problem:
The same computer, but instead of using the on board sound card,
I now connected my new external “Saffire 6 USB” with SX-8 mic.
The picture of the bad waveform is the one I sent before.
I want my waveforms to be like the one attached with this post.
It is easy for me to see where are the bad parts of the wave and avoid using them.
You need to reduce the gain on the Saphire 6 (the physical “gain” knobs), and reduce the recording level for the device in the Windows Control Panel. (You can also access the Windows setting and adjust it by right-clicking the speaker icon in the system tray (by the system clock) and choosing “Open Volume Mixer”).
Hi,
Thank you for replying.
This works fine with the old and simple card, not with the usb card.
I want to work with waveforms which occupy the full range of the audacity waveform truck (like in picture2).
I did it with the old card and i want to do it with my new setup. So i can easily see where the wave might be clipped.
If i reduce the gain i get a waveform with a small amplitude and its clipping area is not clear to me. it is somewhere around the place where the envelope tool draws its line, but just around.
Why cant waves that were recorded with my SX-8/Saffire 6 usb configuration, occupy a full range of the audacity waveform truck?
Because the waveforms are being clipped before they get to Audacity. The most likely place that they are being clipped is at the pre-amp stage in the Saffire 6.
If audacity could draw a pair of lines (clip limit up and clip limit down) , where the clipping occurs, it would be close to what worked before (with the old setup). In picture 2 there are places which i see immediatly that i should avoid using (those which were clipped by the waveform truck or too close to its edges). Also in picture 2 the amplitude is high and i did not need to amplify the wave. In the first picture, the wave never occupies the whole truck. Therefore i have to amplify it. And i do not realy know where i was close to be clipped.
As far as Audacity is concerned, clipping occurs at +/- 1.0
If clipping has occurred somewhere in the audio chain before the data reaches Audacity then Audacity does not know about it. For example, if you have a really sensitive microphone and you record a rocket engine at a distance of 5 cm, then obviously the microphone will not cope and the signal from the microphone will be heavily distorted before it even reaches the pre-amp. Audacity has no way of knowing that has happened. As long as the signal that Audacity receives is in the range of +/- 1.0, then as far as Audacity is concerned it is valid data.
When recording it is necessary to have the correct recording levels throughout the signal chain. You were obviously lucky with your previous set-up and did not need to be concerned about anything other than the level in Audacity, but in your current set-up you have clipping distortion occurring before the signal reaches Audacity, so now you need to be concerned about it. If the distortion is occurring at the pre-amp (in the USB sound card) because you have the gain turned up too high, there should be some visual indication on the pre-amp that this is happening (probably an LED lighting up - check the manual for details).
You said you do not get the red overload lights on the sound card. Try pressing the PAD button and try again.
The SX-8 microphone is a moving coil or “Dynamic” type. You can generate very high signals with a dynamic microphone, way higher than you could with the simple computer microphone you were using before.
The clipping is happening at the analog to digital conversion step – inside the sound card. That’s why you can’t affect it with any of the level controls inside the computer – the Windows Control Panel or Audacity.
The 48V button should not be pushed.
Make sure the Mixer control is all the way toward Input.
Is the sound crunchy and distorted when you plug your headphones into the sound card?
Are you using an XLR cable between the microphone and sound card?
I have read your posts.
I attach two pics: Picture3 and Picture4.
My new temporary technique:
I shout a short and loud garbage word so it will be clipped,
and after it i say my dubbing.
Picture3 shows it.
After it I normalize the waveform with Audacity “normalize”.
Picture4 shows it.
I think that waveforms should be shown in audacity that way (picture4 manner).
But without me having to shout.
The -0.5 to 0.5 of the usb card signals should be -1 to +1.
(It is not exactly plus-minus 0.5)
At least there should be lines at -0.5 and 0.5.
Maybe it can be done by paramters of audacity.
Or maybe something have to be fixed between Audacity and the card.
Giving a good loud yell can be a good way to set the recording levels. The recording levels need to be set so that the equipment can handle the loudest sounds that you are going to make and still leave a little headroom for safety.
The issue about it only showing +/- 0.5 in Audacity is as you suggest, something that needs to be fixed between Audacity and the sound card.
You should be able to fix that in the Windows Control Panel.
Open the Windows Control Panel and have a good look to see if there is an icon for your sound card. If there is, then the necessary record level setting is probably in there. If there is no icon specifically for your sound card, then you will need to go into the standard “Sounds” settings. There’s some info here about the Sounds settings in the Windows Control Panel: http://wiki.audacityteam.org/wiki/Mixer_Toolbar_Issues#Using_the_Control_Panel
I have just installed a new PC and my problem is not solved. (As i expected. I tested it on another PC previously).
Focusrite tech support team responded to my emails. They sent me a link to the “Repair” software. The card works fine with “Repair”. I want to work with Audacity.
It looks like Focusrite have taken the trouble to write decent ASIO drivers for the sound card but have neglected open source software that is not allowed to be distributed with ASIO support. If the sound card does not work properly with Windows drivers (as opposed to ASIO drivers) then unfortunately your options are limited to either (a) using non-open source software that has ASIO support (such as Reaper or Wavosaur), or (b) building Audacity from the source code and enabling ASIO support. Unfortunately option (b) is not at all easy unless you are familiar with Windows programming, though there is information on this page if you want to try.
