search for: fmarengorodriguez

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 fo...

...ctually 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 it. > > > On Mon, Mar 18, 2013 at 11:08 AM, Fernando Alberto Marengo Rodriguez < > fmarengorodriguez at yahoo.com.ar> wrote: > >> 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 v...

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 Castells <canistells77 at hotmail.com> Enviado: martes, 19 de marzo de 2013 5:09 Asunto: Re: [ogg-dev] Min and max cutoff frequency Maybe Monty will make a video about it one day and we will all under...

...e about Ogg Vorbis. Could you please tell me which Ogg audio player hardware is available in the market? Thank you very much and bestregards, Fernando ________________________________ De: Benjamin Schwartz <benjamin.m.schwartz at gmail.com> Para: Fernando Alberto Marengo Rodriguez <fmarengorodriguez at yahoo.com.ar> CC: Monty Montgomery <monty at xiph.org>; Silvia Pfeiffer <silvia at silvia-pfeiffer.de>; "ogg-dev at xiph.org" <ogg-dev at xiph.org>; Sergio Castells <canistells77 at hotmail.com>; Lisandro Conde <lisfconde at yahoo.com.ar>; Muro Gustavo...

> 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

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

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

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.

...C 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 it. On Mon, Mar 18, 2013 at 11:08 AM, Fernando Alberto Marengo Rodriguez < fmarengorodriguez at yahoo.com.ar> wrote: > 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. &gt...

...e more interested in decoding these files, since OGG is open source. Do you happen to know any Ogg Vorbis audio player? Thank you very much and best regards, Fernando ________________________________ De: Monty Montgomery <monty at xiph.org> Para: Fernando Alberto Marengo Rodriguez <fmarengorodriguez at yahoo.com.ar> CC: Silvia Pfeiffer <silvia at silvia-pfeiffer.de>; Benjamin Schwartz <ben at bemasc.net>; "ogg-dev at xiph.org" <ogg-dev at xiph.org>; Sergio Castells <canistells77 at hotmail.com> Enviado: jueves, 21 de marzo de 2013 12:08 Asunto: Re: [ogg-d...

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

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, 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

Dear list, I am currently investigating about the FLAC format and one thing I can't understand is the "WASTED BITS PER SAMPLE" flag. I've seen an explanation saying: "After decoding subframe decoder should bitwise shift all samples to left." Could anyone explain to me what this flag is for? Many thanks in advance. Regards, Fernando -------------- next

Dear list, We are investigating about some state-of-the-art lossless audio codecs and their performance in terms of? rate and compression ratio. Therefore, it is very important to us to know whether a codec is hardware independent, i.e. if it produces the same output file regardless of the hardware. Could you please tell me whether FLAC is hardware independent? Thank you very much in advance.

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

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