similar to: Min and max cutoff frequency

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

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

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

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

> 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, 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 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, 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, > 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 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

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 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, 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.

Dear Monty, > You can sometimes make a prediction based on the encoder and specific encoding mode... How can I make such prediction for different encoding modes? > so, for what reason are you actually asking 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

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

hello, i am a newbie to this list and i don't know if this has been posted before (if so, plzz accept my apologies) i just noticed a nice mini-howto on ext2-ext3 conversion at http://www.symonds.net/~rajesh/howto/ext3/index.html i think that maybe a link can be added to this after the download section... cheers Sayamindu *********************************************************************

Dear Erik, >Its not that we need space for 7616 years, its that if we only use >32 bit offsets, then we would be limited to files of 2 Gigabytes >(signed 32 bit integer) is simply not enough. > >For instance, at 96kHz/24 bits, recording 8 channels would chew up >the 2Gigabytes in about 15 minutes. Some songs are longer than that, > >If 32 bits is not enough, the next

Hello my name is Piercarlo and I've discovered a bug in smbd_audit when I trace a Samba 2.2.7 or 2.2.7a on red hat 7.3 -the error exist on the redhat 2.4.18-17kernel ,on the original rh73 2.4.18-3 kernel and on the standard linus kernel (I tried on 2.4.20)-->these kernels are all the kernels that I tried. This error appear when samba is configured as pdc with the roaming of the users

Dear Ulrich, Thanks for your answer. >Well, today 4 GiB is about half an hour of 8-channel, 96 kHz, 24-bit >uncompressed audio, or about 0.9 % of the capacity of a modest 2 TB >HDD. Not much, in other words, and who hasn't cursed yet at artificial >4 GiB (or even 2 GiB) limitations? So I wouldn't be too sure about the >"ever", even though it does seem very far

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