I asked this in HA, but no one answered ... Better luck here?
Here is the situation: I created a file in audacity via generate -> tone -> 440 Hz, Amplitude 1 at 16bit 44.1 KHz I feed the digital output from my soundcard directly into a Behringer DEQ2496 digital equalizer -- A nice toy that happens to have a digital peak/rms meter (0.1 db accuracy)
The following happens:
1. If I play the file directly from audacity I get peak= 0dB and RMS = -3 dB -- no surprises here.
2. If I save the file as wav and play it in foobar2000 or in WMP I get peak = clip , and -3db RMS
3. If I play the wav file in dBpowerAMP I get peak = -6 dB and RMS = -9 dB.
OK, so what goes on? Clearly the digital output is different in the 3 cases above, but why and how?
It seems that dba chops one bit, true? But why is 1 and 2 different? Is there a simple way to look directly at the samples? Maybe capture on a different PC?
Thanks! -- Ham
Here is the situation: I created a file in audacity via generate -> tone -> 440 Hz, Amplitude 1 at 16bit 44.1 KHz I feed the digital output from my soundcard directly into a Behringer DEQ2496 digital equalizer -- A nice toy that happens to have a digital peak/rms meter (0.1 db accuracy)
The following happens:
1. If I play the file directly from audacity I get peak= 0dB and RMS = -3 dB -- no surprises here.
2. If I save the file as wav and play it in foobar2000 or in WMP I get peak = clip , and -3db RMS
3. If I play the wav file in dBpowerAMP I get peak = -6 dB and RMS = -9 dB.
OK, so what goes on? Clearly the digital output is different in the 3 cases above, but why and how?
It seems that dba chops one bit, true? But why is 1 and 2 different? Is there a simple way to look directly at the samples? Maybe capture on a different PC?
Thanks! -- Ham
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