search for: audiostream

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 tool ? greets KoS

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 tool ? greets KoS

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 without problem and also let me choose the audiostream. Any suggerence?

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 reason for not...

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 comp...

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: http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true#comment-488728 To quote: "Less compression sounded slightly better than more compression." I'...

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...

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: 0 [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value: 0 [wmv3 @ 0xb7e21b50]Extra data: 8 bits left, value: 0 Input #0, asf, from 'VODChapter_20100323_09030000_12350000_Ch04.wmv': Duration: 0...

...39; IPs in iaxproxy-servers file loaded at startup and will keep track of load on each machine. It does balancing not per IAX connection, but per call - rewriting call numbers and keeping track of connection status. It's going to be optimized for speed - doesn't do any other modification or audiostream translation - only message passing. If someone's interested -- code + short doc is available at http://www.gradwell.com/tmp/iax_proxy.tar.gz Development will continue - any opinions / comments / contributions are appreciated. Stanis?aw Pitucha Gradwell Dot Com

...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} [._[:...

...le is still downloaded to the client: <object id="RPPlayer" height="200" width="220" classid="CLSID:CFCDAA03-8BE4-11CF-B84B-0020AFBBCCFA"> <param name="type" value="audio/wav" /> <param name="src" value="audioStream.aspx" /> </object> code-behind page of audioStream.aspx: ... Response.ContentType = "audio/wav" Response.AddHeader("Content-Disposition", "attachment; filename=""GSM.wav""") Response.TransmitFile("C:\GSM.wav") ... _______...

...ime (ms) between digits for ; feature activation (default is 1000 ms) ;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds. Do you need extra info ?? What setting can I have set in musiconhold.conf or features.conf to affect the audiostream between my clients ??? Before I could call all my clients, I had musiconhold when putting 'on hold' and I was just figuring out how parked calls worked... Thanks for the help ! Jonas Kellens. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.dig...

...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 > pos...

...0 21:15:40] DTMF[2538] channel.c: DTMF end 'A' has duration 59 but want minimum 80, emulating on SIP/sip-out-3-0003a606 [Feb 10 21:15:40] DTMF[2538] channel.c: DTMF end emulation of 'A' queued on SIP/sip-out-3-0003a606 Does this mean asterisk's DSP recognizes this inband in the audiostream? Can this also be RFC2833 sent by the audiocodes? Is there some way I can stop this in asterisk, maybe disable detection of the 'A' through 'F' digits in the sourcecode? The setting 'dtmfmode', is it only used on outgoing DTMF? In other words if I set a SIP peer to inband,...

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, decoded to pcm-bytes there are 1920 byte...

...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...