Re: [MP3 ENCODER] Resampling at wrong pitch/speed: bug?
Hi Mark, you wrote on Wed, Oct 27 1999: lame -b 128 -X5 -v -V 4 -h -k -d --resample 48 in.wav out48.mp3 The bug is that the error message "Error: resample code not yet written!" was not being printed :-) LOL! Thanks for the clarification. I think the upsample to 48kHz at 320kbs because (IIRC) a 44.1kHz and lower 320kbs frame violates some ISO restrictions on max buffer sizes. Fortunatly most decoders allow for larger buffer sizes. Other than that, is there any reason to upsample? You probably know about that Microsoft/Intel "Standard", the AC97/PC97 according to which all sound-processing has to take place at 48 kHz, regardless of input. So most cards resample everything at 48 kHz. I came to Linux because I was sick and tired of MS/Intel and would want my sound-card to output the digital stream exactly as it is. So I assumed that the resampling to 48 kHz would degrade quality, to confirm my assumption I tried to encode the same wav (from Audio-CD) once with the usual 44.1 kHz and then with 48 kHz. I wanted to compare these outputs (assuming, the soundcard does not do any more "clever" resamplings at 48 kHz). This darn MS/Intel "standard" is very annoying, if you want bit-exact outputs, like e.g. AC-3 or DTS-Streams (DVDs). Kind regardsFrederick -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
Re: [MP3 ENCODER] Resampling at wrong pitch/speed: bug?
Hi all, just gave it a try: lame -b 128 -X5 -v -V 4 -h -k -d --resample 48 in.wav out48.mp3 The .wav file was grabbed from an audio-CD (44.1 kHz). mp3 sounds horrible, way too fast. Is this a feature or should the resampled file sound like the original? The FhG resamples to 48 kHz without any noticeable speed-up. FhG does the resampling even by default, if you choose 320 kBit. Ok, it's lame 3.35, haven't tried out the 3.36 but since this is not mentioned in the history-file, I suppose 3.36 behaves the same. Kind regardsFrederick The bug is that the error message "Error: resample code not yet written!" was not being printed :-) I think the upsample to 48kHz at 320kbs because (IIRC) a 44.1kHz and lower 320kbs frame violates some ISO restrictions on max buffer sizes. Fortunatly most decoders allow for larger buffer sizes. Other than that, is there any reason to upsample? I have been thinking about the downsampling: I think we should just add a simple linear downsampler, with the tapered lowpass sox type filter applied directly to our MDCT coefficients. This would require very little extra work, and my guess is the aliasing from linear downsampling is insignificant compared to what MPEG2 compression does to the high frequencies. Has anyone who uses sox/MPEG2 noticed a difference between the sox 'resample' option and 'polyphase'? Mark -- MP3 ENCODER mailing list ( http://geek.rcc.se/mp3encoder/ )
