similar to: Stupid question about recording prompts

(Disclaimer: this is not an LPC vs. LSP question) After looking at the Vorbis code I was wondering why you were using LSP to quantize the spectral envelope instead of simply quantizing the cepstrum (DCT(log(envelope))) or modified cepstrum (DCT(envelope.^alpha)). To me it seems like when the information is already in the frequency domain, there's no need to go back to LPC. Also, I think a DCT

Hi, I have a RH9 system with an onboard VIA sound chip. According to the archives, VIA won't work for asterisk. So, I disabled the VIA and I purchased a Creative Labs "Soundblaster PCI 128-Voice" soundcard ($13). This card is on the "approved" RedHat list. However, the documentation inside the package says "VIBRA 128". Anyhow, kudzu doesn't see then

Anyone else seen this eth-device and what brings it up? [root at oh1mrr etc]# ifconfig __tmp1192348344 Link encap:Ethernet HWaddr 52:54:AB:DD:2E:D1 UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1 RX packets:2094689 errors:0 dropped:0 overruns:0 frame:0 TX packets:1645734 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:1000 RX

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

The problem was because it is missing the "loop" module. the hotplug loads it automatically, while udev does not. The error xenoprof: Initialization failed. Intel processor model 15 for P6 class family is not supported" does not interference the winxp load.. so its ok. Does anyone know how to load osx86? It hangs at "booting from cd-rom". do I need kqemu/qemu xen? I

Here's a problem that I have uncovered on two entriely separate machines using the latest icecast2 and ices2 software. Even when the onboard soundcard is disabled, the PCI soundcard (which has been both a Creative Live! and an Ensoniq soundcard) is unable to accept input or stdin audio. If I enable the onboard soundcard everything works well and I can stream audio using icecast and ices. But

Ok, is this possible: I want to have a 128k and a 24k stream of a particualr audio program, plugged into the line in of my Ensoniq AudioPCI 128 (es1370 chipset) Is there any way to do this with just one soundcard, or do I need two? Thanks Scott W --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a

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

Good day! I create diagram of encoder process. Using it i create implementation of encoder in FPGA (Xilinx/Altera). Please critique it. Is there missing stages? Here is blog http://developer-fpga.blogspot.com/ Here is picture of encoding stage 1 https://lh4.googleusercontent.com/-NV8o9DG3jvE/TYjYXr-dYGI/AAAAAAAAAos/U06O-YvhSI0/s1600/stage1.jpg Here is picture of encoding stage 2

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

NSQ_LPC_BUF_LENGTH is independent of DECISION_DELAY. --- silk/define.h | 4 ---- 1 file changed, 4 deletions(-) diff --git a/silk/define.h b/silk/define.h index 781cfdc..1286048 100644 --- a/silk/define.h +++ b/silk/define.h @@ -173,11 +173,7 @@ extern "C" #define MAX_MATRIX_SIZE MAX_LPC_ORDER /* Max of LPC Order and LTP order */ -#if( MAX_LPC_ORDER >

I noticed that my previous message is not very complete so I send here an "enhanced version". Please disregard the old one an reply to this one only. ( you can delete the ATTENTION word from subject ) Wilson (defiler@null.net) wrote : > There are two ways to decode multi-channel audio. In hardware, or in > software. > Hardware: A receiver or processor takes a Dolby Digital

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

> On Feb 5, 2017, at 7:13 PM, Rodney M. Bates via llvm-dev <llvm-dev at lists.llvm.org> wrote: > > A couple of Modula-3 developers have worked on splicing LLVM on as an alternative > back end to the Modula-3 compiler, out-of-tree (the LLVM tree), of course. A major > portion of the necessary glue code is there, and at one time, I was able to get the > M3 compiler and the

Hi. I have been talking with Psytel MPEG-4 AAC developer (www.psytel.com) about various compression formats, and to make it short he said that one of the main drawbacks of Vorbis is its vector quantization - it introduces more error then other quantization methods (like non-uniform scalar quantization, if I got it right), probably used in AAC and other similar codecs. I know vector quantization

Hi all. I'm starting to study how speex works because I would like to understand in deep how a CELP coder/decoder works. I have some difficult understanding the vector quantization performed in the split_cb_search_shape_sign function. It's a bit hard to understand. Could you please explain me the algorithm step by step? I'm indagating the 5950 bps narrowband. Thank you. Gio.

Hi, Attached is a patch for cosmetics purpose. Please review. Thanks, Linfeng Zhang -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.xiph.org/pipermail/opus/attachments/20170419/34354707/attachment.html> -------------- next part -------------- A non-text attachment was scrubbed... Name: 0001-cosmetics-silk-correct-input-output-arg-comments.patch

Comments? ---------------------------------------- "Mark Taylor" <mt@sulaco.org> wrote: [...] > This is related to one minor objection I have to vector quantization > based codecs like Vorbis and the MPEG4 VQ codec: they do not compute > the quantization noise during the encoding process. The choice of > codebooks (use a big codebook: low quantization noise, use a

Hi folks, My wife and I are taking a long weekend to celibrate our anniversary. I won't be around again until Wednesday to answer email or do anything else on Vorbis. Yeah, I know, I usually answer my email once a month anyway (so likely no one would notice me being gone), but just so folks know if anything really juicy comes up :-) I'll be merging my latest branch with mainline

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