similar to: mixing N sounds together...

> -------------------------from aoTuV_b3_En.txt----------------- > Manuke's patch is used for improvement in the speed of sort > processing > (thank you!). > -------------------------------------------------------------- http://www.geocities.com/Area51/Dungeon/6809/vorbis-optsort-en.html This patch accelerate encoding time about 10%. Why not include in 1.1.2?

> -------------------------from aoTuV_b3_En.txt----------------- > Manuke's patch is used for improvement in the speed of sort > processing > (thank you!). > -------------------------------------------------------------- http://www.geocities.com/Area51/Dungeon/6809/vorbis-optsort-en.html This patch accelerate encoding time about 10%. Why not include in 1.1.2?

Hello, Since the list was upgraded I have noticed i am getting 2 copies of each message. I only today had a chance to look into this and wonder if someone can help me figure out what's happening & clean up my list subscription. For every post I am seeing identical posts from 2 different hosts: 1) westfish.xiph.org 2) master.debian.org I am guessing that there is a listserv running on

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It seems like more efficient file transfer technology may be about as important to bringing video production and distribution to the masses as more efficient video compression. I'm wondering how much of a difference BitTorrent made in the downloads of the theora sample files. Is there a record of the number of downloads that were made, and the number of Gbytes that were sent from the server

Hi I am a student and working on Speech Coding and Speex. I just want to know where to get the document describing subvector quantization of innovation. Is there any document which can help me know how C code works or any paper exactly explainin the idea? regards Faisal __________________________________ Do you Yahoo!? Friends. Fun. Try the all-new Yahoo! Messenger.

Hello, I am working with speex codec, especially its VAD part. I need to know what algorithm it is using for silence detection and where I can get some documentation for it. I will be very greatful for your help. -Abhishek

Hello everybody, I can't download Honey from theora.ogg via torrent. The error message I got is : [16:58:19] Problem connecting to tracker - <urlopen error (111, 'Connection refused')> The command i tried was : btdownloadcurses.py --max_upload_rate 350 --url http://www.theora.org/torrents/Honey-small.torrent Please help me. Thanks in Advance regards Manilal --

Hi there. Could someone please upload the minutes from the last monthly meeting to the xiph.org website (when it comes up again)? They're always a very interesting read... Thanks, --Adam

hi i'm looking for information about the multichannel spec. of vorbis. 1/ the documentation specifies that 256 channels are available ( -c 256) ... it sounds very well but how may i specify the original files ??? (i haven't 256 physical entries on my soundcard ... arffff) so is there a way to declare something like oggenc file1.wav file2.wav file3.wav -c3 final.ogg ??? 2/ panoramics are

The first problem is that if you make a builder or .rxml file it always displays as xml in a browser. declare! it and the browser gives and error and if you change rxml to rhtml then it thinks you should be making erb. SO quetion #1 is how do I make builder files render in the browser as html. The next thing that I want to be able to do is be able to render parcials or call other files of other

I'm trying to get a recursive file listing with smbclient: smbclient //host/share -Uusername%password -c "ls -alFR" But I get an error: NT_STATUS_NO_SUCH_FILE listing \-alFR Evidently, smbclient thinks I'm asking for a file mask. Is there a way to get a recursive file listing on a samba share so I can pipe it to wc -l for a file count? Ideally, I'd like to: (This won't

Hi, I am currently working on a thin-client "low-cpu" usage telephone application. The current setup is that you can make phonecalls using an alsa supported usb headset and a smartlink based usb modem (a sweex usb modem, which -in large quantities- goes for 17.95 euro's) for analog lines. Unfortunately, it is really made as a modem and not for audio purposes, so, there is a pretty

OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line

> One thing I've noticed so far in the filter_mem2 code is the calls to > SATURATE(x, 805306368). 805306368 is 0x30000000. I was expecting that > to be on a bit boundary, say 0x3fffffff? In which case the arithmetic > saturation logic could be used. I don't think it would make that big of a difference, since the saturation is outside of the inner loop. If it's that

Hi all, The patches linked below introduce a new family of intrinsics, for integer saturation: @llvm.usat, and @llvm.ssat (unsigned/signed). Quoting the added documentation: %r = call i32 @llvm.ssat.i32(i32 %x, i32 %n) is equivalent to the expression min(max(x, -2^(n-1)), 2^(n-1)-1), itself implementable as the following IR: %min_sint_n = i32 ... ; the min. signed integer of

I want to get a lighter shade of a color...I have a lot of colored objects and want each one printed as a foreground against a slightly lighter background. I thought I could try something like changing the alpha channel by first converting it to rgb. But prior to trying that, I'm stuck with how to get the color after converting using col2rgb() to be interpreted again as a color, rather than

Hi all, I'm proposing integer saturation intrinsics. def int_ssat : Intrinsic<[llvm_anyint_ty], [LLVMMatchType<0>, llvm_i32_ty]>; def int_usat : Intrinsic<[llvm_anyint_ty], [LLVMMatchType<0>, llvm_i32_ty]>; The first operand is the integer value being saturated, and second is the saturation bit position. For scalar integer types, the semantics are: int_ssat: x <

I'm trying to get the X100P card working in AU. So far I have managed to get it to handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db

Hi Everyone, I cannot seem to get the voicemail gain option g(#) work in Asterisk 1.4.11. I am using it like so... Voicemail(4444 at mycontext,bg(10)) ; for busy announce and 10dB record gain This has absolutely NO affect on the resulting voicemail wav file. I have also tried using "format=wav" instead of wav49 in voicemail.conf to increase the volume as well. This also has no affect