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it has way to much bass and hardly any highs or mid's needs lot of work still
yup you are right, it process even heavy distortion at some point...
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Gunther,
I'm intrigued by this statement and it's one I've heard here and there.. My question is, don't most modern encoders have their own internal bandpass filtering? Meaning, you could send completely raw audio, with no filtering whatsoever, and the codec should be able to filter out what it needs to, based on it's settings and it's capabilities. If I artificially limit the bandwidth, I could be missing something that the codec would normally be ok with and pass through.
Or... Is it a matter of the above may be true, but the internal filtering is crap and we shouldn't trust it and ought to filter the audio ourselves? When I said modern encoders, I'm speaking of MP3 and HE-AAC, mainly.
Thanks.
lossless Encoders do compress data and they have indeed internal filters, but these arent brickwall bandpass filters,
Leif explained the technical side about how encoders work a few years ago and the deviation in bandwith with the encoding, I can't find the article back unfortunately.
Bottom line is mp3 encoders used at their limits produce pretty fast distortion, especially siblings. That's why I filter my former streams from 30- 16Khz still pretty high for a stream with attenuation from 1.5db in the end signal path. I used Edcast (both plugin and standalone version) and Samcast and all they have the same "issue" if you call it a issue
You can see the deviation of MP3 below
16 kbps, 11 khz, Mono, MP3
24 kbps, 22 khz, Mono, MP3
32 kbps, 22 khz, Mono, MP3
48 kbps, 22 khz, Mono, MP3
56 kbps, 22 khz, Stereo, MP3
64 kbps, 22 khz, Stereo, MP3
96 kbps, 44 khz, Stereo, MP3
128 kbps, 44 khz, Stereo, MP3
160 kbps, 44 khz, Stereo, MP3
192 kbps, 44 khz, Stereo, MP3