similar to: Flac-dev Digest, Vol 67, Issue 3

2010/4/27 Brian Willoughby <brianw at sounds.wa.com> > i.e. I wouldn't be surprised if there are no FLAC files with this flag > set. > If you look for lossyWAV you'll find that it is actually being used -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac-dev/attachments/20100427/12821b2a/attachment.htm

I'm guessing here, but it seems that some encoders might only be able to process samples in 4-bit or 8-bit groups, maybe even 16-bit multiples, so this flag would allow the decoder to remove any padding that might have been necessary on the encoder side. For example, if someone were to encode 12-bit samples, but the encoder was forced to use 16-bit registers for calculations, then

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 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

> 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

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, 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.

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, > 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 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.

I can't get it to run since v1.1 (lossywav) with latest dev wine or the stable builds. Last version of lossywav wich worked was 1.0. My PC : Piii-550, mga400 vga, intel bx chipset, 512mb, debian lenny/sid 2.6.18, wine 1.13 Here is the crash log: @p550:~/tmp$ wine lossywav fixme:ntoskrnl:KeInitializeSpinLock stub: 0x4577a4 wine: Unhandled exception 0x0eedfade at address 0x0000:0x7b845c60

Hi, I have a two Colum matrix ( I named it holder) as shown bellow. It seems to be three columns, but the first column contains the row names. I am trying to fill all NAs by linear interpolation the known values in the matrix. I tried using the interpNA function, but I get an error message: > interpNA(holder, method = "linear") Error in approx(idx, x[idx], 1:n, method = method, f

Hello, I've set up and have been reading through the FLAC reference implementation source code on Windows and stepping through it in the debugger.   I've been trying to understand how the protocol knows where the subframe and frame boundaries are. Is there a good tutorial that discusses the ins and outs of the flac protocol? Also, is there a piece of the reference code that shows how

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 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

On 05-06-13 00:27, Erik de Castro Lopo wrote: > Martijn van Beurden wrote: >> Considering flac.sourceforge.net, is this ever going to be updated? In >> case it should be redirected, I checked on my own sourceforge project >> webpage, adding the following two lines to .htaccess should redirect >> traffic to any resource on flac.sourceforge.net to xiph.org/flac >>

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