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Re: Actually, no, i don't consider myself a Guru at all ...

195.86.126.19

You're point misses the mark.
It's simply not an issue.

Any half decent record engineer adjust his levels for a maximum input to capture the best possible dynamic range. That was true for tape and is still true for digital recording. Tape saturation was, and still is, used for effect.

Simply ask a drummer, singer or any other participant to make a loud noise and adjust your levels accordingly and add some headroom to compensate for the effect that live performances tend to get louder during the session.

Add to that the knowledge and experience and you know how to compensate for sufficient headroom.

This is just good practice and hold true for any format.

You still refer to 'waisted bits'. Which shows a lack of understanding the basic pcm theory and why it works so well for audio recording.

If you adjust your equipent for lets say a maximum loudness level of 100dB with 6db headroom. You use 23 bits dynamic range but you still encode the sample value with 24 bit precision so no bits are waisted!
Each 6dB headroom 'costs' only 1 bit.

Now imagine a soft sound at 40dB during the performance. This sounds dynamic range is captured within 13 bits. (Sample values are still encoded with 24 bit precision!) Any distortion errors due to a lower signal level within the dynamic range window of your recording setup fall far below the hearing threshold.

You're point isn't valid because the end users volume level is usually adjusted in accordance with the highest loudless levels.
Suppose the users listens to this recording at 90dB loudness.
Any artifacts already below hearing threshold will get quiter another 16 dB.


Frank


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