similar to: [PATCH] doc: specify that quantized LPC shift must be non-negative

Tristan Matthews wrote: > Refs http://sourceforge.net/p/flac/bugs/424/ > - Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement). > + Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement and must be non-negative). Maybe it's better to explain why? At least

>From the FLAC format description: <5> Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement). What happens when the shift is negative? Problem is that libFLAC decoder (and possibly all other decoders) uses always '>>' operator and this operator shifts only to right. (int)x >> -1 is the same as (int)x

Hi all, When trying to better understand the way LPC exactly works, I stumbled upon something which, after some digging, was already reported and (partly) fixed: https://sourceforge.net/p/flac/bugs/424/ Apparently, the FLAC specification has a LPC shift that can be both positive and negative, but the encoder specifically makes sure that only positive shifts are encoded and the decoder only

I am also philosophically opposed to changing the specification. That said, there's nothing wrong with adding a note to the specification about the common implementations, particularly the reference library. Then, future developers will know both the precise specification and still have the warning that they risk incompatibility by deviating from the reference implementation. I own devices

Yes, this is such a case. However, implementing this in a future encoder/decoder would break compatibility with most (likely all) existing decoders, and only in some very, very rare cases where the material is such that the encoder chooses to use negative shifts, which makes it even harder to troubleshoot. Furthermore, as this can only be used in very rare cases, there is no benefit from allowing

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 just noticed how the LPC prediction is quantized by browsing the source code of the java port jFlac: : int sum = 0; for (int j = 0; j < order; j++) sum += qlpCoeff[j] * data[startAt+i-j-1]; data[startAt + i] = residual[i] + (sum >> lpQuantization); : I checked the C version (libFLAC) and it matches. However, it was a surprise because this way of quantizing the

I have some data measured with a coarsely-quantized clock. Let's say the real data are q<- sort(rexp(100,.5)) The quantized form is floor(q), so a simple quantile plot of one against the other can be calculated using: plot(q,type="l"); points(floor(q),col="red") which of course shows the characteristic stair-step. I would like to smooth the quantized

Is this a case where something allowed by the specification isn't implemented by the reference encoder/decoder (such as 25-32 bits per sample) but could be in a different implementation? If so, I am not sure whether it makes sense to change the specification based on the reference implementation. Stephen On Wed, Jun 17, 2020 at 3:22 PM Martijn van Beurden <mvanb1 at gmail.com> wrote:

To me the real question is not whether that portion of the spec has been implemented by any existing encoders/decoders but whether the spec is broken (i.e. cannot be implemented as written). I don't know the rationale for making the LPC shift explicitly signed. In C a negative shift is undefined and it does seem in FLAC__lpc_restore_signal() for example that the LPC shift is used as the

Dear all, I have been trying to understand what runif() is telling me. I am generating lots of numbers (billions and billions (wow, I''ve dreamed about saying that for many years... :-) ), for a distribution that has the following quantile function: 1 / (2 * sqrt(1 - p)) (that is, the distribution has a lower cutoff) As you can imagine, this has rather heavy upper tail. I was

I'm copying the flac-dev list to see if anyone has any feedback also... --- Juhana Sadeharju <kouhia@nic.funet.fi> wrote: > Hello again. I had time to check the paper out. I have filled the > steps given in the paper with formulae, and then written a piece of > C code. It is not complete code, but could be a reasonable start. > Maybe there is one typo in the paper -- I have

Good day! I'm creating HDL IP CORE (for using in FPGA) for theora encoder (now only I-frames). I don't undestand one moment. Now i develop such stages: 1. From RBG(byer) to YCbCr converter 2. DCT processing (8x8 pixels blocks) 3. Quantizator of DCT coeff. 4. Zig-Zag of quantized DCT coeff. and now i have uresolved last stage of compression - how i must send 8x8 blocks to huffman

Hello, I wanted to know if there was way to convert a histogram of a data-set to a kernel density estimate directly in R ? Specifically, I have a histogram [bins, counts] of samples {X1 ... XN} of a quantized variable X where there is one bin for each level of X, and I'ld like to directly get a kde estimate of the pdf of X from the histogram. Therefore, there is no additional quantization of

Hi radeonhd, nouveau, mesa3d developers, Firstly, thank you all very much for all the important work you do. I've been working as a part-time developer on the "Spring RTS" project (open-source game engine) which runs on linux (and other os). Some time ago I tried the engine on the open-source ATI radeonhd driver, which I understand to be partly based on mesa 3d, and all textures

Hello. Earlier I asked if it would be possible to use one common large codebook for all audiofiles. Can you estimate how the following system would work? 1. The audio would be divided to 16 samples blocks. 2. For each block, the first sample would be stored without any encoding. 3. Linear interpolations between the stored samples are computed. 4. The residues (length of 16 samples) between the

Hi, can someone help me understand and experiment with Speex in a better way than my knowledge. I want to change the LPC filter order of the Speex algorithm. I think I need to change the LPC order( or lpcSize) in modes.c file. but to make it work I need to modify the codebook also, can someone tell me how to go about changing the codebook entries to support the changed order of prediction.

Thread-split from the vorbis-mailing list ("Vorbis determined to be as good as MPC at 128 kbps!") <p>On Sun, 30 May 2004, Segher Boessenkool wrote: [Steven So] SS>> If iTunes AAC can encode castanets with much less pre-echo at SS>> ABR 128 kbps, then hopefully there will be an imaginative SS>> (and non-patented) way of doing this in Vorbis without the SS>>

On Thu, Jan 16, 2014 at 5:32 PM, Tobias Grosser <tobias at grosser.es> wrote: > On 01/17/2014 02:17 AM, David Blaikie wrote: > >> Right - you usually won't see a normal distribution in the noise of test >> results. You'll see results clustered around the lower bound with a long >> tail of slower and slower results. Depending on how many samples you do it

I've made a block diagram of the encoder because I tried to find out, how it works http://stoffke.freeshell.65535.net/ogg/block.html Although there are specifiation docs, that give very detailed information about single aspects of the encoding (or decoding) , I'm missing documenations that give a more general overview, about how the encoder works. (Vorbis Illuminated seems a bit