search for: audiostreaming

Hi i have seen that the dbPowerAmp ripping and encoding software supports a new so-called "FLAC uncompressed" format, e.g. http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true i know only the normal flac compression levels from 0 to 8. have i missed an option on the flac comamnd line tool or how could i achieve that on the linux command line flac

Hi i have seen that the dbPowerAmp ripping and encoding software supports a new so-called "FLAC uncompressed" format, e.g. http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true i know only the normal flac compression levels from 0 to 8. have i missed an option on the flac comamnd line tool or how could i achieve that on the linux command line

Hi there, first mail to this list, I hope to not disturb with my question. I'm trying to encode a VOB file (from a DVD) with ffmpeg2theora, but it have a lot of subtitles (about 26) and I can't find the correct --audiostream id to encode the file in my language. I have tried with ids from 20 to 40 and so on and all times the audio kbps it's 0. mplayer and VLC can play the .vob file

Hi, While building an ogg-vorbis stream encoder, I encountered some problems with silence in the audiostream. The bitrate drops to almost zero, and pages going out less then ones a minute what makes the stream to stop / buffer. In earlier postings I read that I shout use ogg_stream_flush as an alternative to ogg_stream_pageout, in case of silence. My question is this; Is there any

Jonathan Billings wrote: > > Amazingly it appears that Red Hat has released another Firefox: Red Hat just follow the Mozilla ESR stream release cycle, which is currently ESR 78 with point releases come out every 4 weeks or so Each ESR stream is supported by Mozilla for about a year, with each major release based on the standard 'rapid release' version at the time - i.e. the

Hi All! I've seen, that icecast can stream videos! Great! Audiostreaming with icecast is well known, but how about videostreaming? Can you tell me some basics about it? - which software is required? (Icecast, ok ;) what else? - by the way: my OS is a linux ;) ) - I want to stream a live-video with 1024x768 resolution - is it possible? - is it wise? - what a compres...

On Oct 23, 2020, at 14:45, Leon Fauster via CentOS <centos at centos.org> wrote: > Mozilla released version 68.12.0, on August 25, 2020 -> > https://www.mozilla.org/en-US/firefox/68.12.0/releasenotes/ > > RH has an ELS phase - if it gets fixed then only for paying customers. Amazingly it appears that Red Hat has released another Firefox:

Martin Kos wrote: > Hi > > i have seen that the dbPowerAmp ripping and encoding software supports a > new so-called "FLAC uncompressed" format, e.g. > > http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true Wow, check this comment:

Hi Icecast-Mailing-Group, I am going to hold a lecture about the audiostreaming technology (Icecast) we are using at our University Radio Station. The lecture is broadcasted to the internet and it is taking place tomorrow, June 29th, starting at 17:30 (5:30 p.m.) CEST. Unfortunately the lecture is going to be in German but at least it might be interesting to all the German Ice...

Hi, I am trying to reencode a video from the European Parliament (WMV) which has multiple audio tracks for the different languages: ================================================================================================ Debian-50-lenny-32-minimal:/var/www/tmp# ffmpeg2theora --audiostream 0 VODChapter_20100323_09030000_12350000_Ch04.wmv [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value:

Hi list I've written a tool that works as a lightweight (standalone - no asterisk) balancer for IAX servers. It's in early development now, but seems to be stable enough and handles couple hundred simultaneous calls with not much latency (SIPp + asterisks tested). It's configurable by listing servers' IPs in iaxproxy-servers file loaded at startup and will keep track of load on

...eo channel connected directly to a mixer is missing in search.*$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usbaudio: constructing mixer for Terminal [0-9]+ type 0x[0-9]+$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usbaudio: device [0-9] audiocontrol interface [0-9] has [0-9] input and [0-9] output AudioStreaming interfaces$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usbaudio: device [0-9]+ interface [0-9]+ altsetting [0-9]+: format 0x[0-9]+ sratelo [0-9]+ sratehi [0-9]+ attributes 0x[0-9]+$ ^\w{3} [ :0-9]{11} [._[:alnum:]-]+ kernel: usb_audio_parsecontrol: usb_audio_state at .*$ ^\w{3} [ :0-9]{11} [._[:aln...

Hi, I am trying to implement a web application that allows the user to playback WAV files inside a web browser. __________ Background __________ - The file server stores audio files as GSM 6.10 or TrueSpeech encoded WAV files. - The web server (most probably ASP.NET on IIS) offers the client the ability to search audio files on the file server (a different machine). - The client will be

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

...our stream fast enough at that resolution. That said, I've never done this so you'll probably have to test the waters yourself to get a good idea of the settings you'll need. Good Luck, Chris Anatol wrote: > Hi All! > > I've seen, that icecast can stream videos! Great! Audiostreaming with > icecast is well known, but how about videostreaming? Can you tell me > some basics about it? > > - which software is required? (Icecast, ok ;) what else? - by the way: > my OS is a linux ;) ) > - I want to stream a live-video with 1024x768 resolution - is it > possib...

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Hi all, i try to set up something like a mixer and soundbridge. There are several sender clients which sends opus encoded audiostreams. One basestation is decoding the streams and mixing them all together. After that, the resulting stream is encoded with opus again and send to one or two receiving clients. Opus is set up to encode 10ms packets in stereo with 16bit per sample and 44kHz. So,

...ne per sending client and one per receiving client. In each thread there is either an opus_encode() or an opus_decode() but there is in total only one OpusEncoder and one OpusDecoder defined which where used in every thread. Now every thread has it own OpusEncoder or OpusDecoder. After 20minutes of audiostreaming there is not even one occurance of the previous described problem. Kind regards, Jan

Hi, I've managed to get Skype running under Debian with the latest Debian release of wine. The sound is perfect, and there are no connection problems, but I noticed the following: 1) Sound from my microphone disappears ONLY when I start some other application or switch to another application. I can only vouch for this in KDE, and it happens whenever anything accesses the soundcard, so

Hi, Even excessive load shouldn't kick out clients. Although the playing of the audiostream may become a bit shaky, it wouldn't stop completely. Or, if it would indeed stop, it wouldn't restart without the computer being rebooted, and for sure not restart after 10 minutes. That doesn't make any sense. What's more: missing interrupts and shaky playback would only eventually get