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Audio Discuss, How to standardize input level when recording (audio streams, etc.) at International Chat: Software related forum; The only way I know is the manual (trial/error) method. Is there anyway to do this automatically through the recording software? I'm thinking that the software would have to override the (pc's and media player's) volume levels and set them accordingly when recording. But then not


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Old 21-07-2006   #1 (permalink)
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How to standardize input level when recording (audio streams, etc.)

The only way I know is the manual (trial/error) method.

Is there anyway to do this automatically through the recording software?

I'm thinking that the software would have to override the (pc's and media player's) volume levels and set them accordingly when recording. But then not all sources are the same level so there still has to be some flexibility.

Keeping all that in mind, trial/error seems like the best solution, but there's got to be a way to have the software perform this task. Once the audio is playing through the sound card, is there any software that can automatically optimize the gain/input settings for recording?

Please help.

If you can point me in the direction of a software that could perform this task, or have any constructive advice...would be very nice !
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Old 12-08-2006   #2 (permalink)
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Re: How to standardize input level when recording (audio streams, etc.)

Good that you haven't asked Bob Katz. He would have shaken his head because of such an intension.

Of course there could be software available to do this job like the automatic algorithms built into tape decks or videorecorders. But there is a fundamental issue when it comes to automize recording levels:

The program would have to raise the signal as much as possible to keep the signal to noise ratio and the dynamic range high enough while avoiding any clipping which would distord the audio. If the incoming signal would be always at the same level, this process could be easily made in practice. But because it is not, the maximum level which occurs in a given signal has to be known *before* recording. How should a program know that in advance? Another problem raises because some soundcards (like the Creative Labs Soundblaster Live! Value) already clip before (!) reaching 0dBFS - probably because of poorly designed analog stages. Thus it would be pretty tricky for an algorithm to recognize such distortion because it is within the allowed level range. Even when exceeding it while recording, it is hard for an analog to digial converter to tell whether or not clipping occured because in the digital world there is nothing beyond 0dBFS (the 0dBFS+ phenomenon disregarded).
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Old 12-08-2006   #3 (permalink)
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Re: How to standardize input level when recording (audio streams, etc.)

Thanks for the reply little-endian. I expected nothing more > way too many variables.

But I just thought of something...

All of what you said makes sense and seems to be impossible for the software to make such decisions. However, you mention the software would have to know the incoming level *before* recording.

What if the software can be manually setup so it knows what to look for and adjust levels within user specified margins?

In other words, you're not expecting the software to *find* the clipping point, you're *defining* it manually.

And I don't know who Bob Katz is
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Old 13-08-2006   #4 (permalink)
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Re: How to standardize input level when recording (audio streams, etc.)

My remarks assume you are recording an analog source. Gain riding is what studio engineers sometimes do on recordings. They manually turn the gain up on quiet parts of the music so the average level is higher -- but they know what is coming up and are prepared. If you are recording an analog source onto a CD or digitally to your computer you will probably have some clipping unless you record at a low level. A decent ADC will clip softly. You must monitor the source and do the best you can with your levels. If you record the whole recording at a very low level so all the recorded levels are below "zero dB" it is possible with an editing program to "normalize" the levels which boosts the volume so the highest peak is at "zero dB". These programs also do can do compression which causes the average volume level to be higher without clipping the signal. But recording at low levels causes loss of definition due to the way the encoding algorithms work -- may or may not be important depending on the source. There may be some recording programs that will automatically gain ride while recording in a manner similar to a some monaural cassette decks that are used to record conversations -- these recorders probably just clamp the loud parts to avoid gross distortions. So your trial-and-error approach is probably your best bet. Clipping is inevitable if you want a reasonable average level -- try to get clipping but very little of it. And use a good sound card. m-audio 2496 or USB/Firewire Audiophile or something better than the mass-market SB and SB clones. After the fact you can adjust. And I see I have reiterated much of what little-endian already said.....regards, gamma1

Last edited by gamma1; 13-08-2006 at 19:55.
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