search for: quantised

The following creates a point process version of a sinewave (maybe there's a better way): p<-amp*cos(2*pi*freq*(1:n)/n ) + 0.5 as.numeric(runif(1:n)<p) I get something like this: 1 o oo oo o o ooooo o 0 ooo o o o o o ooo ooo In case it's not obvious, this is a noisy version of 1 ooooooo ooooooo 0 ooooooo

Hello vorbis developer! I'm not very experienced in informatics, cause i'm only an audio engineer and sorry for my broken english but i have an idea... After the filterbank the maximum aplitude in each band decides about the nessesary bits, then the psychoacoustic model decides how much of the signal can be truncated without hearable noise. What do you think about quantising the

...iles changed, 28 insertions(+), 27 deletions(-) diff --git a/libcelt/bands.c b/libcelt/bands.c index d0c3498..315cce8 100644 --- a/libcelt/bands.c +++ b/libcelt/bands.c @@ -247,23 +247,6 @@ void compute_pitch_gain(const CELTMode *m, const celt_norm_t *X, const celt_norm }*/ } -/* Apply the (quantised) gain to each "pitch band" */ -void pitch_quant_bands(const CELTMode *m, celt_norm_t * restrict P, const celt_pgain_t * restrict gains) -{ - int i; - const celt_int16_t *pBands = m->pBands; - const int C = CHANNELS(m); - for (i=0;i<m->nbPBands;i++) - { - int j; -...

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...ve a patent on compressing audio by transforming to the frequency domain, quantising, and then entropy encoding, which is what Ogg Vorbis does, so there's trouble. Now, I've been working on a Haar wavelet image compressor, and since I haven't written the entropy coder yet I just dump my quantised output to a new file. This file is as large as the original file. Then I simply use gzip, which compresses well because of the preprocessing. So, should trouble arise, you could make Ogg Vorbis from a compressor into a transformer. Since you're not doing any entropy encoding, in fact you'r...

--- libcelt/Makefile.am | 6 +- libcelt/bands.c | 26 +++++++++- libcelt/bands.h | 2 +- libcelt/celt.c | 23 +++----- libcelt/pgain_table.h | 133 ------------------------------------------------- libcelt/quant_pitch.c | 117 ------------------------------------------- libcelt/quant_pitch.h | 44 ---------------- 7 files changed, 37 insertions(+), 314

I have been using libvorbis for the past few weeks and have been asked to summarise what I have discovered about the codec. There is an early draft of the document at http://www.geocities.com/gatewaystation/vorbis/vorbis.htm - Please forgive the dodgy formatting (it was formerly a MS word document that got converted with their 'save as html' feature). I still have some additions to

Hello, I am working on Cirrus ARM-EDB 9302 and using speex-1.1.7. I have tried encoding a wav file of 7sec long and it took 6 sec to encode it for default settings of encoder with both files on the RAM.Next I tried encodeing for 1 minute wav file and it took 2.2min for default settings. Why is it that as size of input file increases,encoding time also increases drastically?( I decoded

https://bugzilla.mindrot.org/show_bug.cgi?id=3655 Bug ID: 3655 Summary: Default ObscureKeystrokeTiming makes X forwarding really slow Product: Portable OpenSSH Version: 9.6p1 Hardware: Other OS: Linux Status: NEW Severity: normal Priority: P5 Component: ssh

Hi, I've been reading through the Theora format spec and API reference trying to find an answer to this but haven't come across anything certain. I'm wondering whether it would be theoretically possible to change the encoded bit rate (or target video qiality) during a live stream without a standard theora decoder having any problems. I had assumed that this must be possible, given

...12 bit is useful. > And the DAC is also 12 bit, it means that I just need the high 12 bit > and discard the low 4 bit after decoer? That's what I said. It's not going to work. It's not because your input signal is 12-bit deep that suddenly the LPC and LSP coefs will accept being quantised in 12 bits. Also, as I said in my previous email, I'd recommend shifting your 12-bit data only by 3 (not by 4) when converting to 16-bit. That'll give you a bit of headroom. Cheers, Jean-Marc

Ha ha! I'm don't know what change I made that cleared up the errors, but I can figure that out later. Of course, I immediately tested my new oggenc. Using the same input file and default options, I get these results: [MSVC] File length: 4m 21.0s Elapsed time: 0m 54.0s Rate: 4.8333 Average bitrate: 129.6 kb/s File size: 4,229,428 [BCB]

Hello, I wanted to find out more about square polar mapping, cause that whole stuff is a bit tricky - to me - and not so easy to understand. Searching the web did't get me any results, so I wonder what the background of square polar mapping is. Is it derived from some mathematical publications or just the result of a brilliant intuition of a Vorbis programmer ? ;) Thanks. Stoffke

On Mon, 7 Aug 2023, Chris Rapier wrote: > > The broader issue of hiding all potential keystroke timing is not yet fixed. > > Could some level of obfuscation come from enabling Nagle for interactive > sessions that has an associated TTY? Though that would be of limited > usefulness in low RTT environments. I don't like the idea of having a steady > drip of packets as that

Hi?Jean-Marc? Thanks for your suggestions very much! > > I am porting speex on ARM7TDMI, I have done some optimization, the > > result is that the encoder and decoder need about 60 MCPS for 5.96kbps > > bitrate and complexity 0. Can someone give me informtion about Speex > > optimization on ARM7? > > That's quite good. A few suggestions here: > 1) Don't

...o zero. So point stereo is essentially monaural, since angle=A-B so if angle is 0, A=B? Hence Vorbis "joint" stereo is essentially a mix of lossless stereo and mono, right? It seems like a very strange way of coupling as I would have expected point stereo to at least have a heavily quantised angle.....that is, not many bits go into encoding the angle but we do have at least a rough angle/diffuse sounds. Also, why not implement mid/side stereo instead of this mix of lossless and intensity stereo? I apologise for the very large e-mail but I am very interested in the internals of Vo...

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

I sent a little mail some time ago asking if there was going to be an ogg pic-format, and you replied that PNG, MNG and JNG is good enough (sorry for this late answer btw).. But, consider this: The ogg video-format (tarkin ? where do you get these names from anyway ? :) ) needs a way to compress its frames. Are you going to use MNG for that ? :) .. If you had an ogg pic format, that format could

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello everybody, first of all I want to excuse myself for crossposting. I can't estimate which mailinglist could be the right one for our concern. Struktur AG, member of the Linux Solutions Group e.V., wants to announce an Open Source tender for an implementation of an audio codec compatible with Nellymoser Asao Codec. Struktur wants to pay

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 Hello everybody, first of all I want to excuse myself for crossposting. I can't estimate which mailinglist could be the right one for our concern. Struktur AG, member of the Linux Solutions Group e.V., wants to announce an Open Source tender for an implementation of an audio codec compatible with Nellymoser Asao Codec. Struktur wants to pay