Audacity Forum

ok... I am new here. I plugged in my microphone and went to edit and set the preference to 2 channel stereo. When i record, there are 2 channels, but the botton channel has a straight line. what am i doing wrong? Thx chris

That's out of my league, but here's a little snip from the Audacity Wiki's Trouble shooting Guide:

http://audacityteam.org/wiki/index.php? ... Recordings

Unbalanced stereo recording

If you are recording a stereo source and the left or right channel is weak or non-existent, first make sure you have Audacity set to record in stereo. Do this by setting "Recording Channels" to "2 (stereo)" on the Audio I/O tab of Preferences. You recorded track will then say "Stereo". Make sure your physical recording connections are tight and that any balance controls on for example your mixer or amplifier are centrally placed. Make sure the left and right channels are balanced in the system mixer. On Windows this is the recording section of Sound, Sounds and Audio Devices or similar in the Windows Control Panel (access it by hitting the Windows key on your keyboard), or on Mac OS X 10.2 or later this is Mac hard disk > Applications > Utilities > Audio-MIDI Setup: Audio Devices.

If you cannot correct the imbalance at the input stage, then in Audacity, click on the Track Panel above the mute/solo buttons to select all the audio of the track, and click Effect > Normalize. This will scale the volume of each channel independently to -3 dB (3 dB less than the distortion level), so they are at the same volume level. If you then want to adjust the tracks to a higher or lower level (keeping the newly adjusted balance between the channels), use Effect > Amplify. By default (simply click OK in the effect) this amplifies to -0 dB, so making both tracks as loud as they can be without distortion. Note: in the Beta 1.3.x version of Audacity, Effect > Normalize allows you to choose a custom level to normalise to, so you don't need to use Amplify afterwards if want a final level other than -3dB.

If for any reason the result of normalisation does not sound as you would like it, you can alternatively click on the downward pointing arrow in the Track Panel to reveal the Track Pop-Down Menu and and choose Split Stereo Track from the menu. This enables you to separately edit and control the left and right channels. Press the green Play button or spacebar and use the -....+ gain sliders on the Track Panels of the left and right channels to adjust the gain on each to your satisfaction as the track plays. Note this is not an edit of the waveform so it won't change appearance, although the gain changes are respected when you export the audio. To edit the waveform itself so you can see what the exported waveform will look like, use Project > Quick Mix (or Tracks > Mix and Render in the Beta version).

If you are changing the gain on the tracks with the gain sliders, make sure before you export the audio that you don't apply too much gain so that the result distorts. You can check this by looking at the green VU Playback Meter in the Meter Toolbar . Enable this Toolbar on the Interface tab of Preferences if it is not visible: we always recommend using this Toolbar to test the recording level before you begin any recording, so that the overall input level is not too loud or too soft.

If you prefer to adjust the gain on each track to a precise level avoiding any risk of output distortion, click in the Track Panel of the left channel to select it, Effect > Amplify it to your chosen "New Peak Amplitude" (the box underneath the slider), then repeat for the right channel and amplify that to your chosen New Peak Amplitude. You must amplify both channels separately if you want to change the balance between them.
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If that doesn't help, maybe this tutorial might contain something that will:

http://www.guidesandtutorials.com/audac ... orial.html

It's much more simple than that. Most Mic Inputs are mono, you won't get a stereo signal out of one.

alatham wrote:It's much more simple than that. Most Mic Inputs are mono, you won't get a stereo signal out of one.

I was wondering about that, but I tried it on mine and I got a dual signal for both channels after setting it to stereo. I use a headset that has two plugs, one for the input & the other for output. Because I don't like to have to crawl behind my desk to plug & unplug things, I haven't plugged in the speaker portion of it as I want to be able to listen to the speakers. It's not the ideal situation, but it works reasonably well.

I only have one plug going in & I assume the microphone is mono, but I was getting dual outputs when Audacity was set for stereo. Maybe it is a stereo input, however, even though it only has one input plug. Anyway, here's a screenshot of a very short test clip:

I bet if you were to split those into two mono track, panned center, and invert one of them, then mix them together they'd cancel each other out.

Audacity sometimes will record a Mic In signal to two different tracks, but those two tracks are usually (always?) identical. The signal is effectively mono.

Some drivers handle this differently and will send nothing to the other tracks, your drivers apparently send the same thing to each track.

Yeah, I figured they were probably identical. I didn't realize the different drivers might handle it differently like that. Thanks for the info.

<<<they were probably identical.>>>

And you don't have to go to extremes to figure it out, either. Just select some of the audio and keep blowing it up (zoom in) until you can see the individual humps and valleys in each wave. They should match. "Stereo" tracks don't.

As far as the mono/stereo thing goes, this is how a stereo headphone is wired:



That's not how a microphone is wired although it uses the same plug. The shield is the same, "R" is now computer battery voltage going up to run the microphone electronics, and "L" is the single microphone performance going back down to the computer sound card.

There is only one analog performance voltage, not two. If you get two tracks, it means some "tricks" were done in software to make copies.

Koz

Hi,

I use a USB microphone that already has an A/D converter that automatically samples to 16 Bit. I didn't read the specification of the microphone before doing my recording, and I used 32 Bit sampling in Audicity. Did that affect my recording for linguistic purposes? How can something that's already been digitalized be further digitalized? How do my microphone and the Audicity interfere? Does Audicity have an anti-aliasing filter? How do the Audicity and the sound card compression interfere?

Sorry, there just seems to be too many devices involved in recording a sound file and I don't have a clue how they interfere.

Stefan

stefanst wrote:I use a USB microphone that already has an A/D converter that automatically samples to 16 Bit. I didn't read the specification of the microphone before doing my recording, and I used 32 Bit sampling in Audicity. Did that affect my recording for linguistic purposes?

Linguistic purposes? I'm not sure what that means. But I can tell you that your recording is fine.

How can something that's already been digitalized be further digitalized?

It's a simple matter of using more bits to represent the same data. A very poor analogy: Lets say you're trying to represent the number 3. You can either use a set of integers to represent it, or you can use a floating point number (decimal point). So you've either got 3, or 3.0. Both are the same number, the second one takes more space to store (but is also more accurate).

The advantage of moving a 16-bit piece of data to 32-bits is that you reduce any rounding errors you might have received if you do any editing to the file. If all you're going to do is export the file to 16-bit, then there's no improvement (but also no added problems).

I would recommend just leaving Audacity at that setting.

How do my microphone and the Audicity interfere?

The word you're looking for is "interface." Basically, Audacity doesn't interface with your microphone directly. Audacity has to go through Windows' driver software first. So Audacity tells Windows "I want to start recording" and Windows tells the microphone "Start recording." Then Windows has to relay all the into to Audacity. It's an inefficient way of doing things, but it's the only practical way of making a machine capable of multi-tasking.

What the microphone is doing is taking an analog signal, then converting it to a 16-bit digital signal (at the right sample rate) and sending that through the USB port.

Does Audicity have an anti-aliasing filter?

Yes, but only when re-sampling, not when changing the bit depth of a signal.

How do the Audicity and the sound card compression interfere?

Sound card compression? Do you mean Data Compression (mp3, FLAC, ogg, etc), or Dynamic Compression (the audio effect)?

Data Compression has nothing to do with your sound card at all. It's entirely up to software playing those types of files to convert the audio to a PCM (wav-type) signal so the sound card can play it. Further, Audacity doesn't talk to your sound card at all when creating compressed files.

Dynamic Compression also has nothing to do with your sound card. All Audacity is doing is manipulating the data.