similar to: Optimizing Vorbis for a DSP

_________________________________________________________________ Trouvez l'âme soeur sur MSN Rencontres ! http://g.msn.fr/FR1000/9551 --- >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-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No

first, congrats to monty and the whole crew for getting this close to 1.0. it's a monumental achievement reflecting a huge amount of hard work -- kudos! econd, i know the bitrate peeling feature has been pushed back until after 1.0 is out. but i'm wondering -- will oggs created with a 1.0 encoder be peelable with an as-yet-unreleased utility, or do "peelable oggs" have to be

As part of the documentation of a Vorbis decoding project, a quick explanation of the encoding procedure is required. I quite up to date with the decoding process but not with the encoding part. Is there any documentation explaining the different steps during encoding? What I'm looking for is a list of encoding steps, very briefly explained and more details on a few fundamental steps, like

hello, My project is to implemente an audio compression algorhitm on a DSP TMS320DM642. I think that SPEEX can be a good choice because fixed-point have been developed. I would want to know how I can choose the fixed point in the program I want to know if it is possible to do a real time acquizition and to compresse in real time the audio input signal. By example, I have on my demonstration card

I wrote to the list earlier about optimizations of the tremor. Specially replacing the IMDCT with a N/4 FFT. The report is available here: http://www.sandvall.nu/thesis.pdf Short summary: Optimization of the Tremor code under 50kB for a "split" decoder version. ~44 MIPS. Huffman highly unoptimized. IMDCT ~ 8 MIPS. Quick overview of the

Hi Frank, Actually, I just merged (in SVN) a patch doing exactly that. AFAIK it allows Speex to run on a C55x. I think it should work for you with the following caveats: 1) The static codebooks still take up 16 bits for every 8-bit value (that's only a code size issue) 2) The bit-packer always returns a multiple of 16 bits instead of 8 bits (but it uses all bits when packing). Jean-Marc

Hello everyone, I am running Ubuntu and have converted my CDs to FLACs (using all the paranoia and fixing all the tags etc.). What is the best way for me to convert all of those FLAC to ogg-Vorbis files, keeping all the tag information and the same directory structure? I am not wanting anything incredibly advanced, but was expecting something like vorbconvert /media/cd_rom/ /home/aaron/vorbis/

Hello everyone, I am running Ubuntu and have converted my CDs to FLACs (using all the paranoia and fixing all the tags etc.). What is the best way for me to convert all of those FLAC to ogg-Vorbis files, keeping all the tag information and the same directory structure? I am not wanting anything incredibly advanced, but was expecting something like vorbconvert /media/cd_rom/ /home/aaron/vorbis/

Is there a archive with a collection of different .ogg files encoded with different settings that can be used to verify compatibility? * Different window size (e.g. != 256,2048) * Much larger codebooks and other extreme cases that are not generated using a standard encoder. * floor0 Regards -- / Johannes Sandvall --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project

We've implemented a vorbis decoder based on Tremor and as part of the documentation we're also writing about psycho acoustic models and encoding. We're quite up to date with the decoding process and psycho acoustics in general but unfortunately not on the psycho acoustic encoding used in Vorbis. We have a few questions that would we would be very thankful to have answer to: Which

"Do you mean doing that on a DSP, or on a PC (before you do it on DSP)?" Yes I want do that with a PC in order to test if I can encode in real time an external audio source and then I want to implemente SPEEX in a dSP. guillaume Jean-Marc Valin a écrit : > > I want to know if it is possible to do a real time acquizition and to > > compresse in real time the audio input

Sorry for starting another topic, this is actually a reply to Segher's post on Sun Oct 22 on the 'mdct question' topic. I wasn't subscribed properly and so I didn't get email confirmation and thus can't add to that thread. So Segher, if the equation is indeed what you say it is, then replacing mdct_backward with this version should work, but it doesn't. Am I applying

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I want to encode wav file into fixed window length vorbis file. I don't know if "window" is good name for that, but i mean MDCT buffer size. This buffer is passed to mdct_backward() as "in" and it's size is determined by "init->n" variable which is also passed to this function. So, when I say "window

Hello, I may be stupid, but I cannot find any data on how "expensive" the Vorbis Decoder is. Looking for an embedded solution I am interested in how many "MIPS" you need to decode a MONO audio stream at reasonable quality (44 kHz, approx. 100 kbps?) Can it be done in a microcontroller, or do I need ultra-expensive DSPs to do it? Thanks for hints Jens --- >8 ---- List

Hi all, I'm attempting to convert all the pointers to arrays the mdct_backward function so it can be partitioned off for a hardware implementation. Although this code is quite short I'm finding it a little tricky. As it stands, mdct_backward is passed values by reference i.e. void mdct_backward(mdct_lookup *init, DATA_TYPE *in, DATA_TYPE *out) o my modified version starts void

Can anybody point me at any resources that would explain how to optimize mdct_backward for a cpu with a fused multiply-accumute unit? >From what I understand from responses to my older postings, Tremor's mdct_backward could be rewritten to take advantage of a muladd. My target machine can do either two-wide 32x32 + Accum(64) -> Accum(64) integer muladd or eight-wide 16x16 + Accum(32)

Hello, Iam interesting in transcoding the algorithm of OGG VORBIs in fixed point because I have to implemente this encoder in a fixed point DSP. Does the source code of ogg vorbis encoder exist in fixed point or is there people that work on this or anything else? Guillaume --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

On Wed, 21 Feb 2024 21:19:23 +0000 evil cRaftKnife via samba <samba at lists.samba.org> wrote: > I do have 'local' users in OpenLDAP. > But how is Samba supposed to know this ? Are you also running a dns server ? I ask this because a Samba AD DC runs kerberos, ldap and a dns server and is a lot easier to set up than your existing setup, I would also recommend you use a

The users in LDAP appear as local users on my file server using the /etc/nsswitch.conf passwd: files ldap group: files ldap [root at fs ~]# getent passwd | grep kev kev:x:10000:10001:Kev:/home/fukr/kev:/bin/bash [root at fs ~]# getent group | grep kev net-users::10001:kev net-admins::10000:kev media::10002:kev Kerberos is working fine, so is NTP and DNS. My problem is that my issue

I have a share setup on a Samba 4.0.8 / CentOS 6.4 box that is successfully replicating with a W2K3 server. I'm following the HOWTO here: https://wiki.samba.org/index.php/Setup_and_configure_file_shares [mytest] path = /home/me/mytestshare <-- with or without trailing slash read only = No On the W2K3 box, I can browse to \\newdc and I see my test share listed there. I can also see