similar to: Question about residue and pdf

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.

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

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

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, 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 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

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 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

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

Hi, Please brace yourselves for some possibly stupid questions. Why are there only 2 codebooks for floors and many more for residue values? What makes the encoding of floor values fundamentally different from encoding residue values? I have a basic understanding of VQ, and looking at the floor decode function, it all makes sense 1) Read the codebook index from the stream 2) Do a

Hi I am trying to build a new decoder in Java, I know there are a couple of java decoders already but they don't implement the bit of the spec I am having questions on Question 1) In section 8.6.2 it mentions "However, it is also possible to set [residue_begin] and [residue_end] to specify a range partially or wholly beyond the maximum vector size. Before beginning residue decode,

Hello, I'm trying to modify the residue vectors in Vorbis audio files at encoding time. For testing, I'm just adding +1 to each residue vector's element, so e.g., if I have an original residue vector of [0,1,2,3] I modify it during the encoding to be [1,2,3,4], and at the decoder's side I should receive the [1,2,3,4] vector. The problem is that, after adding the +1, I still

I found this sentence in the Ogg format specs: http://www.xiph.org/ogg/vorbis/doc/vorbis-spec-res.html "A residue vector may represent spectral lines, spectral magnitude, spectral phase or hybrids as mixed by channel coupling." But where does the spectral phase information come from ? AFAIK MDCT doesn't provide any phase information. And in OGG-encoding, MDCT is taking place a few