similar to: CELT solution for beginner

Hi Guys Sorry, this is off-topic here but relates to current use of CELT and/or Opus in real-world systems. If there's a better forum to ask this question please let me know. I'm wondering if anyone can point me in the direction of software I can use right now that will give me high-quality point-to-point live audio streaming accross the internet. Kind of like Skype, but designed

Any way of writing definition lists? Andreas -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 3147 bytes Desc: not available Url : <http://six.pairlist.net/pipermail/markdown-discuss/attachments/20081114/cb1ee902/attachment.bin>

Use apps like mumble of Teamtalk, they are build to do what you want to do. Kind regards, Dennis Op 19-9-2015 om 00:32 schreef Marvin Scholz: > On 18 Sep 2015, at 23:31, Orion Jensen wrote: > >> I'm investigating the build out of a Push to Talk server with multiple >> subscribers as part of a mobile app. > That most likely will not work with Icecast, you might want to

I am working on a project that transports audio in realtime from a linux box to a windows box throuh libcelt. The windows client program is written in C# and C++ and I therefore need the celt library in its most recent version (0.7) to be able to compile in Visual Studio 2008. In order to do this I, as I understand it, need a .vcproj file for the project. Are there any plans on making such a

Hi all, I've been looking at opus lately as a replacement to vorbis and it sure seems to me that, at least on paper, it is the 'next generation' streaming codec. Has anyone been using opus as a streaming codec to Icecast? If so, how do the following general characteristics compare to vorbis? + Quality + Latency + Bandwidth + Performance (on stream client and Icecast server) +

Hello Dennis! Please send me the links to your streams. I'm interested in the audioquality. regards Gagarin M. Stockholm www.mrs.se Dennis Heerema <Dennis at Heerema.net> wrote: >Hello Jordan, > >I run a test stream in OPUS format. >I find the quality of OPUS way better at low bitrates then VORBIS. >Thus better quality at lower bitrates, but also on higher bitrates the

Hello The st->zeta pointer isn't freed in the speex_preprocess_state_destroy() function of the preprocess.c file (alloced in line 167). It's in Speex 1.1.4 by the way. I'm trying to make the denoiser work with my application and has got reasonable noise reduction after applying the denoiser. I, however, haven't been able to find any information of what the purpose of

Hello there I'm having trouble decoding speex frames when using 32KHz audio and quality setting 4. If I encode three frames and then decode the three frames then the speex_bits_remaining() still reports that there are bits remaining. All other band modes and qualities reports that the buffer is emptied. Here's an example that shows 4 bits remaing in the buffer after the frames have

Hi all, I'm setting up a backup script for my Mac, and I'm wondering about the rsync ability of dealing with HFS+ filesystem (resource-fork and such...). I've searched the mailing list archive for this topic and I've found some thread regarding it, but it's not clear to me what is the actual situation. Is the HFS+ officially supported? Will it be in the near future?

Hi, I am a newbe here, so I hope I am not asking something crazy. I was looking on the Celt downloads and I noticed that the latest command line builds were for version 0.6. I know it is still early in development, and that the devs probably don't have time for developing CL versions for everything. since I know nothing about sourcecode and whatnot, I was wondering if there was a fairly simple

I'm investigating the build out of a Push to Talk server with multiple subscribers as part of a mobile app. Has anyone seen this usecase with Icecast? Any suggestions or places to look. Thanks, *Orion Jensen* CEO | ClearLaunch 1408 East 13th Street | Austin, TX 78702 Skype: orion.jensen | Mobile: 1.512.270.3976 orionjensen at ClearLaunch.com <peyton at clearlaunch.com> |

hi, I saw that Mike Dransfield tried to port 'state' plugin from beryl. What does it do? It should be able to place windows, based on name, class etc., to specific viewports. I recently converted to compiz window manager from WindowMaker (used it for 8 years), and I miss automatic 'pinning' of specific windows to particular workspace (or viewport in this case). Mike's

Hi! My mail server (ubuntu breezey badger) is connected by ADSL and has a dynamic address. I have noticed that, every day after the IP address changes, dovecot is no longer reachable from the net. Iguess this is because dovecot doesn't register this change and continues to listen on a non-existing interface/address. I'm trying to solve this problem by putting a script in /etc/ppp/ip-up/

First, I would like to compliment you all for a great work you put in developing compiz. I have it running for more that a month and it did not crash. Stability is very good. Great work! Today I updated compiz from git tree and found that rotate generates this error: compiz (core) - Error: Couldn't activate plugin 'rotate' Rotate does not function. Any ideas ? I run gconf

Hello, I am trying to build llvm/clang using cmake and visual studio 2008 on windows vista. I want to convert the llvm bitcode file hello.bc into C code. I saw in the documentation that "llc -march=c hello.bc" would convert the bitcode and generate a c code file. but it says llc:error: invalid target "c" could you tell me why is it not working. and what is the way to convert

I have successfully installed oH323, but when I make a call the recipient can hear me but I can't hear anything at my end, so voice is only travelling in one direction. Client Softphone (SIP) => Asterisk => H323 Gateway. All Codecs have been loaded. Any help would be appreciated. Ade. --------------------------------- To help you stay safe and secure online, we've

Hello, I have the following situation: A firewall is blocking all SIP and RTP traffic in the side of some of my clients. My clients cannot change settings of the firewall. I need to solve this problem and I need some help from you. I have this idea: implement a SIP user agent which does not use well known SIP ports (uses http port 80 for example) and use other ports that are not blocked

I am developing an R package which has a function I decided not to export. I believe the roxygen information states that the function is not going to be exported, yet I still see the function in the manual PDF (as generated in command line via `CMD Rd2pdf package_dir`). What is wrong with my preamble that the function is still being documented in a manual? #' Generates plots for demo of

People, where to read about how to make the most of the current Dovecot search capabilities, including squat and possibly also the related future plans? There are many aspects like IMAP protocol itself, extensions, current support into Dovecot, support among MUAs... Could you post some good links, please? -------------- next part -------------- A non-text attachment was scrubbed... Name:

Hello all, this may be slightly offtopic :-) I have some Cisco 7940 phones with SIP firmware, connected to an Asterisk 1.2.18-BRIstuffed-0.3.0-PRE-1y-g (HorstBox Pro with custom extensions.conf). On some of the phones, two lines are configured, one for business, one for private calls. When forwarding a line to another destination (e.g. to voicemail), we can't use the phone's own