search for: 16kbps

...er streaming, so as to cut down the amount of bandwidth that the server needs. This means every listener needs to have enough bandwidth to relay at least one stream out as well as in. What I was thinking of is to broadcast 2 ogg streams ; one at 64kbps stereo for broadband listeners, and another at 16kbps mono for dialup listeners, and maybe get a few people with good connections to have a permanent peercast relay for both these streams. But as far as I can see, ices2 won't encode at below 32 kbps, which won't work with peercast on 56k dialup links. I remember from the vorbis documentation...

...er streaming, so as to cut down the amount of bandwidth that the server needs. This means every listener needs to have enough bandwidth to relay at least one stream out as well as in. What I was thinking of is to broadcast 2 ogg streams ; one at 64kbps stereo for broadband listeners, and another at 16kbps mono for dialup listeners, and maybe get a few people with good connections to have a permanent peercast relay for both these streams. But as far as I can see, ices2 won't encode at below 32 kbps, which won't work with peercast on 56k dialup links. I remember from the vorbis documentation...

When is the target release date for Beta 4? FR --- Michael Smith <msmith@labyrinth.net.au> wrote: > At 05:50 PM 12/21/00 -0600, you wrote: > >Hello, > > > >Does anyone know when the low-bitrate modes (e.g. > 16kbps) might be > supported? I > >was hoping they'd be in Beta 3. The reason I ask > is that I will have to shut > >off mp3 http streaming from my site within the week > and it would be nice if I > >could tell my customers when it would be on again > (in Vorbis format)...

...lassid 1:1 htb rate 9bps ceil 9bps #Default tc class add dev eth1 parent 1: classid 1:100 htb rate 9bps ceil 9bps #ICMP tc class add dev eth1 parent 1: classid 1:5 htb rate 144kbps ceil 256kbps #customer A tc class add dev eth1 parent 1: classid 1:101 htb rate 16kbps ceil 16kbps #customer B tc class add dev eth1 parent 1: classid 1:111 htb rate 32kbps ceil 32kbps #customer C tc class add dev eth1 parent 1: classid 1:121 htb rate 32kbps ceil 32kbps #customer D Can anyone help me on how to limit the the bandwidth to these customers. Regards. D...

...tion Manual has an example iax.conf file that indicates the ulaw codec, so I know that one is good. But we are thinking about using the IAXy over a VPN, to replace our MultiVoip. alaw and ulaw are 64kbps, which is too much for our tunnel. G726 would be good if is has decent sound quality at 16kbps. We currently use MultiVoip's Netcoder at 9.6kbps which works fine. So, is there an official statement somewhere about which codecs the IAXy supports? Thanks! -Michael

On Fri, 6 Apr 2001, John Griffiths wrote: > ok so 24kbps for 56k modems... > > can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? As others have said, 16kbps should do the trick. Keep in mind though that the quality of the sound will also depend on the sampling rate. MP3 will handle some higher sampling rates higher than some of the lame defaults. For example, 16kbps 11khz mono sounds very reasonable. At 24kbps, either 22khz mono or 11khz stereo soun...

...s add dev eth0 parent 1:0 classid 1:1 htb rate 136kbps ceil 136kbps $TC class add dev eth0 parent 1:1 classid 1:2 htb rate 128kbps ceil 128kbps $TC class add dev eth0 parent 1:1 classid 1:3 htb rate 8kbps ceil 8kbps # Classes for local users $TC class add dev eth0 parent 1:2 classid 1:20 htb rate 16kbps ceil 128kbps $TC class add dev eth0 parent 1:2 classid 1:21 htb rate 16kbps ceil 128kbps $TC class add dev eth0 parent 1:2 classid 1:22 htb rate 16kbps ceil 128kbps $TC class add dev eth0 parent 1:2 classid 1:23 htb rate 16kbps ceil 128kbps $TC filter add dev eth0 protocol ip parent 1:0 handle 1 f...

...gt; <out-rate>22050</out-rate> > </resample> Thanks, that does it. > >> Specifically, I have a machine with 2 versions of oggenc on it, one >> from debian woody, and one compiled from source. The woody version will >> encode mono wavs down to 16kbps, but the (later) hand compiled version >> will only go down to 32 kbps. The quality of the 16kbps files isn't that >> great, but it's still ok to listen to - no obvious distortion or spurious >> noises, just a loss of clarity. > > You don't tell us what compress...

Hello, I am new to using opus but so far I really am loving it. At 16kbps spoken content sounds fabulous. I am using opusenc on 64-bit linux - mostly with flac input. The one thing I can't figure out how to do is add album art. I tried passing the vorbis way of adding a metadata_block_picture comment field to opusenc but it complained about the length. I can't...

...orage space and sound quality. The input sound is 22050Hz, mono. To maximize the recording capability on a given storage volume, we want the result bitrate as low as possible. I use -q0 parameter and it produces a 32.0kbps stream. It is too high (our competitor using Microsoft WMA can produce 16kbps streams at the lowest quality setting). I tried -q-1 and the bitrate is still 24.2kbps. -q-2 and -b16 both produce error messages. How can I get lower kbps using oggvorbis?

hi, i have posted earlier a few days back, i apologise for the re-post. my problem is that the moment i start using mp3 files with lesser bit rate (16kbps) it does not stream properly. it keeps getting stuck in the middle. i can stream continuously with 128 kbps files. I'm using icecast(1.3.7) with ices(beta 5) in "LAN" on Linux Redhat 6.2. The same problem occurs also on internet, when i run the same setup on the server. this help...

...t bad though, except the machine only has a 2GB harddrive). Currently I am archiving my radio show using the Fraunhofer MP3 codec with some software called AudioGrabber. I just have Audiograbber to record from the line-in on my soundcard. I set it to use Fraunhofer to record directly as MP3 mono, 16kbps, 11kHz. I believe speex could produce better quality/smaller file size. The problem I have is not being able to choose the speex codec in Audiograbber instead of Fraunhofer MP3. Just wondering if there's a way to make speex a selectable encoding codec. I'd like to use speex in real-time to...

...te familiar with ACELP as implemented in G.729. As far as I know Speex is also ACELP ... yes/no? Question ... what do you see as the advantages of ACELP compared to MP3? I had thought that ACELP yielded much smaller files, but now I'm not sure because I've listened to what I think were 16kbps MP3 files (????) with Apple's ITunes that sounded pretty good ... comparable to G.729's 8kbps. I'm speaking about G.729 because I'm not up on Speex attributes ... e.g. what Speex bps compares to G.729's 8kbps quality. -- ----- B. Mitchell Loebel...

...g AMBE (4Kbps) for our Traffic/Weather Channels. If you have ever had a chance to hear the service, you will know that AMBE does not do us well. I understand that 2Kbps is low quality, but any poorer than AMBE? If can get a decent quality for other low bandwidth talk channels, such as about 10-16Kbps and have it sound rather clean, then I would be gaining a bit of growth in the areas I'm seeking. None of this would be Music, understand. Music needs it's bandwidth... just voice only channels like: - Emergency Announcements - Barkers (This channel is not yet scheduled for Air - Traff...

...ld think having speex as a selectable codec in windows would help out alot of people (or at least get alot more people interested). Anyone up to the task of coding it to do this?;) Please, please. :) I was just thinking Speex could beat Fraunhofer MP3 for what I do. My current MP3 settings are 16kbps, 11Khz, mono. A five hour recording is approximately 35MB. Speex sounds like it would be a perfect application for this and achieve smaller/better quality recordings. What do you think? Ben <p><p><p>>In AudioGrabber, you have the option to set up an external encoder, you &g...

Depending on what you are broadcasting, for 28.8 I would go 16kbps and be really safe... Lithium ----- Original Message ----- From: "John Griffiths" <john@capmon.com> To: <icecast@xiph.org> Cc: <icecast@xiph.org> Sent: Thursday, April 05, 2001 3:12 PM Subject: [icecast] bitrate for slow modems > ok so 24kbps for 56k modems... >...

Hello, I've been using Opus on an STM32 M4 platform for speech coding in mono mode. I thought I'd try stereo for grins to see if I can handle the CPU load, and I'm getting a return code of -1 from opus_decode_float (using CBR and 40ms frames). I decided to try the opusenc and opusdec tools to just see how the command line apps would behave. I am getting decoded audio, but I am

...to a certain extent 1) The source code seems to be codec-centric with encode and decode in many of the same files. Does anyone have any hints on what can be removed easily that is used only for encoding? 2) Any estimates on MIPS usage for decode only on the blackfin? Wideband (16kHz operation) 16kbps data rate. Code Space? 3) The folks I am doing this work for use VDSP++ and at this point I need to as well. Any hints at porting from gcc? Any assistance will be appreciated. Thanks Bob Gorry AeVee Laboratories bgorry-x1 at aevee.com -------------- next part -------------- An HTML attac...

...on Linux Redhat 6.2. It works perfectly fine with a bit rate of 128 kbps in LAN but somehow when i change the bit rate of songs to 16 kbps it gets stuck in the middle, all the time 'like someone is scratching the disk'. Earlier i thought it happens because the LAN speed is too fast for a 16kbps streaming, so i tried to put it on the server, wherein i face the same problem. Is it a problem with Ices beta 5 or is it some problem with the encoding technique. Also pls. if u can suggest me which is the best way to encode mp3's for 16kbps or less than that kind of bit rate. Thanks very mu...

I'm streaming a police scanner online. What would be the best encoding to use the ogg or mp3 or mp3 pro Right now i'm using MP3 16KBPS 11KHZ and it uses about 3.1KB/sec of bandwidth. Is there something I can use that has the same quality with lower bandwith. I've tried MP3pro using the lowest profile which I think was 20KBPS at 16KHZ and it uses 3.7KB/sec --- >8 ---- List archives: http://www.xiph.org/archives/ icecast p...