similar to: MCDT,FFT and more

I'm trying to get these cleaned up and landed, but I'm running into some trouble with this patch. Using commit a08b29d88e3c (July 21) of Ne10, I'm seeing test failures for 60-point FFTs: nfft=60 inverse=0,snr = -3.312408 ** poor snr: -3.312408 ** nfft=60 inverse=1,snr = -16.079597 ** poor snr: -16.079597 ** All other sizes tested appear to work fine (84 to 140 dB of SNR). This

>I maintain such a page here: > >http://thebasement.org/sound/vorbis_info > >It's probably woefully incomplete. Well, there are only a few more, AFAIK: - The Quinntissential Player [Windows] http://www.quinnware.com/ - Zinf [Windows & Linux] http://www.zinf.org/ - MacAmp [Mac] http://www.subband.com/macamp/ - Audion 3 [Mac] http://www.panic.com/audion/ --- >8 ----

CELP based coders perform poorly when you mix streams. Perhaps, we need a transform code or a subband code at somewhat higher rate 12-16 kbps that will retain the quality after transcoding and mixing. Is the speex ultra-wideband based on subband coding? -Devendra Jalihal <p>On Fri, 21 Nov 2003, Ulrich B. Staudinger wrote: > Hi, > been silent in this discussion for quite long ...

Hi I have gone through the document available in the net regarding the Vorbis encoder /Decoder. Based on that i have prepared a understanding document on the encoder/decoder block. I would like to know whether my understanding of the coder is OK. If there are any other additional block /information pl. provide me with the same. Thanks and regards S.Padmashri <HR NOSHADE> <UL>

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>>

Hi, Our DSP has a built-in bitrev instruction so we're exploring the possibility of calculating the bitrev every time instead of filling the table during initialisation, hence saving some memory. Our frame size is fixed to 320 samples. The two FFTs sizes for normal block and short block are 160 and 40 respectively. It's not really clear how the function

After reading http://mp3.radified.com/mp3.htm I sent Rad an explanation of some things as I understand them. He liked it and posted it on his site (still unlinked, use the URL below). Can somebody with better understanding of psycho-acoustic terms and the vorbis model check it and comment on it? In particular I didn't know how vorbis handles quantization noise. If you reply with

Hi Timothy, Sorry for late reply. I have upstreamed the patch to fix the regression here: https://github.com/projectNe10/Ne10/commit/ee5d856cd9cb8c4a15ace567df4239f4e788d043 I have tested it with Vish's branch: http://git.linaro.org/people/viswanath.puttagunta/opus.git/shortlog/refs/heads/rfcv3_fft_fixed) Both unit test dft and unit test mdct passed on ARM v7/v8, floating point/fixed

(This is almost out of topic but anyway...) It is surprising how little research effort have been put into psy-acou models for CELP. The basic problem lies in that it is not easy to alter the LP model without distroying the minimum-phase property (ie. the stability of the predictor). That leaves us with psy-acou modelling of the noise-part only. However, my own research is in constrained

http://www.subband.com server down? I want to regist MacAmp... and now, Does anybody know, free-vorbis decoder for macintosh. Kaz Shioda shioda@bgm.ne.jp --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-request@xiph.org' containing only the word 'unsubscribe' in

This is not the same as the recent thread on a waterfall graph. I'm thinking about the rolling FFT display used in acoustics and other spectrum analysis tasks. Here's an example of a very fancy 3-D waterfall display: http://www.ultimaserial.com/UltimaWaterfall.html I was just wondering if there are any simple hidden-line tools in R that I could use to draw simple waterfall displays.

> I had the idea of implementing a lot of the operations in FFTs. ( for > example, it is possible to do auto-correlation and FIR filtering using > FFTs.) There are two advantages to this. > 1. It's almost always faster > 2. By swapping fft implementations, it could be easy to recompile for > fixed or floating point versions. No. FFT's require higher precision than

I have an irregular time series in a Zoo object, and I've been unable to find any way to do an FFT on it. More precisely, I'd like to do an NFFT (non-equispaced / non-uniform time FFT) on the data. The data is timestamped samples from a cheap self-logging accelerometer. The data is weakly regular, with the following characteristics: - short gaps every ~20ms - large gaps every ~200ms

Well, i don't know if we speak of the same ... The quality is ok, but the encoding of a PCM frame on the tested machine took 2ms, which (if the PCM frame is 7ms in length) means the machine can only encode three streams in realtime ... <p><p>Am Sa, 2003-11-22 um 03.30 schrieb Devendra Jalihal: > CELP based coders perform poorly when you mix streams. Perhaps, we need > a

Hello Timothy, Great to hear from you! Fired up my hardware today and this issue looks like a regression in Ne10 library. The commit in Ne10 [1] that I tested to be working successfully back in May 5b63074db45000f9688460990ee3f5e147d93782 which is the Patch Phil at ARM added to fix the overflow issue in nfft=60 case. After git-bisect, looks like the culprit patch in Ne10 [1] is

<with Prof. Farnsworth voice> "Good News, everyone". I've just kicked off project "Sphinx". Which is supposed to sound like "Speex" merged with "INT". ;) Meaning I am working on an integer encoder and decoder. It looks like I will be pulling in a new "integer plumbing" into the foundation of the codec, comparing the results with the old

Hi, Does anyone have an idea about the possibility to apply psychoacoustic models as the ones in mp3 or AAC to a CELP coder? Thanks! /Pontus <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the

Analog Devices sponsored a fixed point version for AEC which was posted early this year, but it seems from the source code that the fft routines are still in floating point. Is the port still not complete or am I missing something? Has anyone out there ported speex AEC on RISC architecture? Please let me know. Thanks in advance, -Deepa -------------- next part -------------- An HTML

On 29 April 2015 at 17:22, Timothy B. Terriberry <tterribe at xiph.org> wrote: > > Viswanath Puttagunta wrote: >> >> This patch series is follow up on work I posted on [1]. >> In addition to what was posted on [1], this patch series mainly >> integrates Fixed point FFT implementations in NE10 library into opus. >> You can view my opus wip code at [2]. >

Hi again. Since mostly stupid people gave me stupid answers last time, I will try again. MP3 and Ogg Vorbis use different models in order to remove most of the bits. What is the names of these models and how do they differ? <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to