similar to: Is FLAC hardware independent?

Dear list, Could you please tell me the values of the minimum and maximum cutoff frequencies for each coding version of the 44.1 kHz sampled data? For instance, are the values fmin=100 Hz and fmax=12 kHz valid? Thank you very much in advance. Kind regards, ? Fernando A. Marengo Rodriguez, PhD Post-doctoral fellow on Acoustics and Beamforming -- Laboratory of Noise and Vibration (LVA) Federal

Maybe Monty will make a video about it one day and we will all understand it. ;-) Silvia. On Tue, Mar 19, 2013 at 3:22 PM, Benjamin Schwartz <ben at bemasc.net> wrote: > Presuming that you are asking regarding the Ogg Vorbis audio format, the > correct answer is: there is no minimum or maximum cutoff frequency. Vorbis > can code all frequencies from DC to Nyquist. What Vorbis

Dear list, I am looking for different window functions used in perceptual audio coding. In the mp3 format, these windows overlap with each other, but I haven't found any information about their mathematical expression, nor their numerical values (see attached document). Which temporal windows are used in ogg audio coding? Thank you very much in advance. Kind regards, ------ Fernando A.

Dear list, > Which "output file" are you referring to?? Also, your question is incompletely specified, because you do not qualify whether the input is the same when you expect the output to be the same. My question is the following: For any encoding option (e.g. -5, default), does the flac encoder produce the same byte-for-byte output regardless of the CPU? Regards, Fernando

Hello Rene, If you want to be sure that you get no information loss, I suggest a very simple test. Recover your WAV file from any of the FLAC files you mentioned in your e-mail. If this WAV file is bit-by-bit identical to the input WAV file, then you have no information loss. Also, it is important to take into account that the compression ratio is highly dependent on the encoded wav file. If you

Oh, I will be very happy if I could see this video! Thank you very much Silvia. Kind regards, Fernando ________________________________ De: Silvia Pfeiffer <silvia at silvia-pfeiffer.de> Para: Benjamin Schwartz <ben at bemasc.net> CC: Fernando Alberto Marengo Rodriguez <fmarengorodriguez at yahoo.com.ar>; "ogg-dev at xiph.org" <ogg-dev at xiph.org>; Sergio

Dear list, I' m studying FLAC performance, and I'd like to know how much compression can be achieved for different audio files. 1) It seems that for nontonal sound (wideband noise), the compression factor is better than for compound sound (tones + nontonal components), which is typically 2. The reason for this result could be the following: the LPC filter is more suitable for

Dear list, I wonder if the flac encoded file has a great amount of bits due to the residue. I mean, what percentage of the flac file has information of the residue? Is this sequence represented by more than 80% of the flac file? On the other hand, which is the value typically adopted for the parameter "n" in the Rice coder? I know these results depend on the input wav file, but I

Presuming that you are asking regarding the Ogg Vorbis audio format, the correct answer is: there is no minimum or maximum cutoff frequency. Vorbis can code all frequencies from DC to Nyquist. What Vorbis will actually do is extremely complex, extremely nonlinear, and highly dependent on bitrate. If you are in the mentality of linear time-invariant filters, you will never be able to understand

Dear Stefan, In the paper "Improved Forward-Adaptive Prediction for MPEG-4 Audio Lossless Coding", a non-linear compander is applied to the parcor coefficients prior to quantization. This compander is designed in order to minimize quantization error, especially for magnitudes close to unity. If you determine the typical distribution of magnitudes of the LPC coefficients, you could

Dear list, Following Victor Westmann's idea, I'd like to translate FLAC's site to Spanish. I'd be very grateful if anyone offers to give me a hand in this. Best regards, Fernando A. Marengo Rodriguez Universidad Nacional de Rosario Yahoo! Cocina Encontra las mejores recetas con Yahoo! Cocina. http://ar.mujer.yahoo.com/cocina/ -------------- next part -------------- An

On Thu, Mar 21, 2013 at 9:54 AM, Fernando Alberto Marengo Rodriguez < fmarengorodriguez at yahoo.com.ar> wrote: > I am asking this because we are studying the OGG Vorbis format and its > applications. We are very interested in constructing an audio player > hardware, based on DSP or DSPic, and the audio files are stored in an SD > card. > What is your motivation for

Dear Benjamin, > What is your motivation for constructing this hardware? ?Is your goal to learn/teach from the experience, or are you trying to build something that you cannot buy? Yes, I am interested in learning how Ogg Vorbis works, and what its performance is in audio players in comparison with MP3 and AAC players. It is said that Ogg encoded files have higher quality than MP3 encoded

> Presuming that you are asking regarding the Ogg Vorbis audio format, the > correct answer is: there is no minimum or maximum cutoff frequency. Vorbis > can code all frequencies from DC to Nyquist. What Vorbis will actually do > is extremely complex, extremely nonlinear, and highly dependent on bitrate. > If you are in the mentality of linear time-invariant filters, you will

> PS: Here is the answer of David Bryant, the developer of WavPack: "The WavPack codec is hardware independent; the standard "C" encoder will produce the same byte-for-byte output regardless of the CPU it is running on (assuming a properly working C compiler, of course)." The same is true of flac, as long as you're using the same version of the encoder, right? The same

Which "output file" are you referring to? Also, your question is incompletely specified, because you do not qualify whether the input is the same when you expect the output to be the same. When uncompressing, the final results are lossless, therefore it is hardware independent when you consider the total process. When compressing, the results depend upon the implementation. In

On Aug 8, 2009, at 23:11, Didier Dambrin wrote: > Electronic music quite often doesn't leave a computer these days. > And it > mainly consists of drums, synths & vocals/effects. Drums are often > samples > sequenced at sample (not sub-sample) accuracy, thus repeated (of > course if > the song was post-resampled, there will be sub-sample times). Good point. I

This patch to opus-tools adds optional support to WavPack lossless format as input to opusenc. Like support to FLAC, it depends on an external library, libwavpack, and may be disabled on configure. Lucas Clemente Vella (1): Reading input from WavPack files. Makefile.am | 7 +- configure.ac | 37 ++++++++ src/audio-in.c | 71 ++++++++------- src/opusenc.c | 19 +++- src/opusenc.h

Hello FLAC list. As far as I know 24 bit FLAC support is broken. It often doesn't compress the audio at all, but instead stores the chunks as verbatim type (although the FLAC format supports 24 bit). Perhaps this is fixed? If so, do let me know. I agree that perhaps 32 bit float/pcm isn't entirely necessary when it comes to storing different qualities. But when wanting to preserve

On Fri, Aug 14, 2009 at 5:05 PM, Josh Coalson<xflac at yahoo.com> wrote: > it's unlikely flac will ever support floating-point samples natively. ?the main application for it is audio engineering, which demands easy editing and very high speed for both encoding and decoding above everything else. thats not why floating point is used. the highest current feasible bit resolution for