similar to: High Latency of vorbisencoder on a hardware with no floatpoint unit

Hi All, It would be really helpful for me if you comment on the following. My VorbisEncoder uses libvorbis to encode into the vorbis data. I have followed the call sequence given in libvorbis\examples\encoder_example.c. I am taking a 16bit pcm Stereo wav file and generating an oggvorbis file using libvorbis and libogg. There is lots of noise in the file except music is heard very feable

Hi, I have tried using the VorbisEncoder class available with the vorbis-java port of the ogg-vorbis and shout and the resulting ogg file from the encoder is jumbed up and distorted. I have tried compiling the sources and changing the quality in the source with no success. Has anyone managed to use the encoder successfully. if so your response will be of great. -- Solomon Kariri, Software

hello, I have recently compiled speexenc for WinCE/StrongARM SA-1110-processor. It seems to run with no errors, however the encoding time is very slow - 10 seconds of speech takes aprox 3 minutes to encode. Even though the target architecture is significantly less powerfull than a standard PC, I am trying to figure out why it is this slow. Does speex use floatpoint calculations? The input raw

Beta release of a limited* native Java Vorbis encoder is now available on the Xiph website. This limited version of the encoder has been ported to Java based on libogg-1.1.3, libvorbis-1.1.2 and a very infant port of libshout-2.2.2 for streaming. *The encoder itself is currently limited to 44100 hz, 2 channel input, and can only encode at Variable Bit Rate.* The release package with binary and

Hey list, I've recently done the first test run of bumping our Burst compiler from LLVM 10 -> 11 now that the branch has been cut, and have noticed an apparent loop vectorization codegen regression for X86 with AVX or AVX2 enabled. The following IR example is vectorized to 4 wide with LLVM 11 and trunk whereas in LLVM 10 it (correctly as per what we want) vectorized it 8 wide matching the

Tried a bunch of them there (x86-64, haswell, znver2) and they all defaulted to 4-wide - haswell additionally caused some extra loop unrolling but still with 8-wide pows. Cheers, -Neil. On Thu, Jul 16, 2020 at 2:39 PM Roman Lebedev <lebedev.ri at gmail.com> wrote: > Did you specify the target CPU the code should be optimized for? > For clang that is -march=native/znver2/... /

So for us we use SLEEF to actually implement the libcalls (LLVM intrinsics) that LLVM by default would generate - and since SLEEF has highly optimal 8-wide pow, optimized for AVX and AVX2, we really want to use that. So we would not see 4/8 libcalls and instead see 1 call to something that lights up the ymm registers. I guess the problem then is that the default expectation is that pow would be

Im talking about this http://downloads.xiph.org/releases/vorbis-java/vorbis-java-1.0.0-beta.tar.gz Do you know it? On Thu, Dec 10, 2009 at 8:56 PM, Monty Montgomery <monty at xiph.org> wrote: > On Fri, Dec 4, 2009 at 5:28 AM, solomon kariri <solomonkariri at gmail.com> > wrote: > > Hi, > > I have tried using the VorbisEncoder class available with the vorbis-java

Hi, I am using asterisk 1.2.1 for building an ippbx for my setup. I am having problem in accessing advanced options option that comes on pressing 5. It says to leave a msg and asks for extension. Now when i dial the extension the IVR silently goes into the top IVR menu. And the asterisk console show the message: "No entry for <EXTEN> in voicemail config file" As i

Hi all, I m in a trouble using festival voices in asterisk. I am not able to change the default male voice of festival. Although i downloaded the us1 female voice and it iw working good in festival's CLI but it is not coming when i am usinf Festival in asterisk. I changed the default-voice-priority list directive and set us1_mbrole as first entry and also changed voice in festival CLI. But

I am using asterisk-1.2.1-15 and want to use it to replace my normal PBX with it. For creating IVR menus i tried festival, the text which was passed into it was said, but the problem was at stating of every line a "tick" sound comes. As festival app in asterisk connects with festval server at each line(as it required festival_server for its functioning and logs can be seen on

I am testing asterisk-1.2.1-15 on RedHat 9(i386) for SIP-to-SIP call and i found a strange problem. When an extension gets a ring and it picks up the call a "tick" sound comes at start. This happens on both sides. I tried Xten's softphones and also hardphones. A thing which was common in both(soft/hard phones) was the selected codec. When i used g711ulaw on both soft/hard phones

HAllo! This is Lokesh. I want to use speex codec in my project for encryption in GSM phones.Is it possible to use the availabale API in SPEEX in eclipse,i.e java IDE tool.Or else do u hav already available compatible package for java.Eagerly waiting for your reply.Please suggest me how to use speex in my project.Desperately waiting for your reply.Thanks for your time. regards,

Hi As per the Linphone, Readme.arm I tried to compile the speex. -------------------------------------readme.arm-------------------------------------------------- ........... Cross compiling speex for ARM: ******************************** First you need to remove ogg headers from your build system to avoid a dirty conflict between your build machine binaries and the arm binaries. They

Hi there again.. I'm still writing my particular-case-vorbis-encoder-and-player, and it worked fine until now. I'm having a problem trying to encoding the audio. I need to send the data very quickly, as soon as i capture it from the sound card. Currently i have the following (audio captured 16 bits, mono): void VorbisEncoder::encode(void *data, int r) { char *buf = (char *)

Hi, I was really interested in the java version of the same since I wanted to use it in my java application in a platform independent way. Anybody who have managed to use the java port to encode wav to ogg, this is only audio, can assist me to solve my problem. I have been looking at the code and the svn but no updates seem to be available. Some guidance on what could be the problem on the source

Hi Till now we used the rails pagination.But now onwards shifting to will_paginate The code I use currently is as below .This is with the help of find_where pluggin..A fantastic pluggin which we still want to use...I would like to know how can i modify the following code to work with will_paginate step 1: set the variables you''ll need page = (params[:page] ||= 1).to_i

http://radiostudio.org/ostream/ostream-ogg.zip imple (and maybe first?) windows software for live streaming (from soundcard). it's far from perferct, but it works. _______________________________________________________________________________ Alles unter einem Dach: Informationen, Fun, E-Mails. Bei WEB.DE: http://web.de Die gro?e Welt der Kommunikation: E-Mail, Fax, SMS, WAP:

Hi I want to contribute some changes to icecast2 server; Some improvements to logging engine of icecast ( daily and monthly aggregation ); <p>During every logging process I check if the actual logfile is still the actual one; if not, I generate a new logfile and write the logentry in the new logfile; in case, nothing changed, I write to the old logfile; <p>it is just 3 little

Soon, Firefox 3.1 and other browsers will start to support vorbis and users will increase vorbis use on the net. Most users (as most Linux distributions) are however using the reference xiph.org vorbis library which generate worse files than the aoTuV library from http://www.geocities.jp/aoyoume/aotuv/ . According to latest discussion: https://trac.xiph.org/ticket/1251 the aoTuV code should have