search for: 32kbps

Hi, good people! I wanted to limit my 4 customers to 144, 16, 32, and 32kbps. I used the following tc commands BUT IT FAILED TO LIMIT each and everyone of them to its bandwidth. What am I doing wrong: My tc scripts are: tc qdisc add dev eth1 root handle 1: htb default 1 #Classes# tc class add dev eth1 parent 1: classid 1:1 htb rate 9bps c...

...rates. There were some small differences, but nothing horrible. So, I decided to run with 24KHz sample rate with 16 bit samples. This seemed like a reasonable tradeoff against the quality degradation that my folks could perceive. Then I tried some bitstream rate changes 128Kbps, 64Kbps, and 32Kbps. 128K and 64K are fine. 32K is *terrible*. Really, really horrendous. Is this expected? Is there a program I can run to estimate the error or something? I expected degradation, but this just falls off a cliff with really objectionable artifacts. My application is going into the embedded...

Hi To continue the interesting streamTranscode thread but in a slightly different direction, I have a user station that would like to send us a 64kbps AAC stream and have us transcode that to 64kbps MP3 and 32kbps AAC - is this possible? Points to consider: a) we would be happy to go with another permutation such as: she sends us 64kbps MP3 and we convert that to 64kbps AAC and 32kbps ACC. b) this AAC patch is undoubtedly going to be useful: http://www.brain4free.org/wiki/doku.php/blog:streamtranscoderv3...

The background noise is unacceptable. I'll have to stick with 32kbps mp3s eventhough the files are 70% larger.

Hi, I want to encode audio at very low bitrates. however, vorbis_encode_init() does not allow me to set bitrates below 32kbps. How can i set 24kbps? Thanks, Flo --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No su...

...ev eth0 parent 1: classid 1:1 htb rate 512kbps ceil 512kbps tc class add dev eth0 parent 1:1 classid 1:10 htb rate 384kbps ceil 512kbps-----Admin-Network tc class add dev eth0 parent 1:1 classid 1:11 htb rate 64kbps ceil 256kbps------cyber-Cafe tc class add dev eth0 parent 1:1 classid 1:12 htb rate 32kbps ceil 32kbps-------Constructions tc class add dev eth0 parent 1:1 classid 1:13 htb rate 32kbps ceil 32kbps--------Consultants tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match ip src 202.79.x.x match ip 0xffff flowid 1:10 tc filter add dev eth0 protocol ip parent 1:0 prio 1 u32 match i...

Hello everyone! I am having a issue with what I am doing wrong here to get listed in the xiph.org public directory. I have the newest icecast-2.3.99.3_beta3, loaded on a Win7 box acting as server, and using BUTT to encode the source at 32kbps, mono, .mp3, from VLC playlists. I have the server config set to public, and have this code in the icecast.xml file: <directory> <yp-url-timeout>15</yp-url-timeout> <yp-url>http://dir.xiph.org/cgi-bin/yp-cgi</yp-url> </directory> The Ser...

I have a CentOS release 4.7 box running asterisk-1.4.26.1 with dahdi-linux-2.2.0.2 and dahdi-tools-2.2.0 I regularly get these messages, is this something i should be worried about? [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_g726.so => (ITU G.726-32kbps G726 Transcoder) [Aug 21 01:05:07] ERROR[4343] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory [Aug 21 01:05:07] VERBOSE[4343] logger.c: codec_dahdi.so => (Generic DAHDI Transcoder Codec Translator) How to fix this ? Google was not my friend this time :)

...arger frame sizes for those of use who don't care about latency. Inbetween the two, the hybrid mode appears to do better than other codecs with similar latency- but Christian Hoene's results showed it losing pretty convincingly to AMR-WB+ (which was able to use 4x larger frame sizes) at 32kbps. (How much of this was due to the test being stereo, I wonder? Some mono tests seem to have given 32kbps Opus rather high marks.) For audiobook use, I don't know that the SILK modes or anything else with that low of a bitrate will be good enough, and when you're storing hundreds of hou...

ok, interesting ! Do you know how much CPU it takes for a real time encoding/decoding ? it's more around 20% or 80% instead ? Do you know if it works for 44100 Hz / 32kbps / Quality =8 ? thanks again. Vincent ----- Original Message ----- From: "Ashhar Farhan" <farhan at phonestack.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: <speex-dev at xiph.org> Sent: Wednesday, November 12, 2008 3:05 AM Subject: Re: [Speex-d...

...ered my license, copied the codec_g729a.so into the /usr/lib/asterisk/modules folder and restarted my asterisk But on the CLI when I type asterisk*CLI> show modules like 72 Module Description Use Count codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g729.so Raw G729 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g723.so G.723.1 Simple Timestamp File Format 0 The codec_g729a.so doesn't appear.............

...rs can be supported by typical server boxes, assuming unlimited bandwidth is available? My specific case: My server (http://www.AirProgressive.org) is on a Linux VPS server located in a datacenter and on a optical backbone so that bandwidth is not the problem. Each connection I am running is at 32kbps and 22050 Hz sampling rate (mono) since the program is "talk radio". The bandwidth available per month is 6000 GB and at an average of 4.5 hours per week of listening, which is what similar stations had in this area of the same genre, I can service about 21,000 listeners per month. T...

Hi, I use liveice/icecast on a linux box redhat 6 (2.2.18) and my problem is that the streamed audio gets chopped -on the LAN!!! The audio is encoded in 32kbps and I have a 512kbps/512kbps connection, so it should be enough to stream several connections at once. It sounds just like when a cd gets dirty or scracthed on the back. Anyway, it seems to go away if you can endure the terrible sound for a while, or you restart winamp. F. ex. suddenly it would wor...

...Dennis Heerema -----Original Message----- From: "Greg J. Ogonowski" <greg@orban.com> To: qiang Bao <jakobao@yahoo.com>, icecast@xiph.org Date: Wed, 29 Dec 2004 11:08:04 -0800 Subject: Re: [Icecast] icecast2.2 and aac? Icecast 2.2 works fine with AAC/aacPlus at any bitrate. 32kbps here: http://stream.orban.com:80/eval -greg. At 00:21 2004-12-29, qiang Bao wrote: it seems that icecast 2.2 can only stream aac at 128kbps! mp3 and ogg are ok at all bitrates. is it a bug or my config wrong? Do you Yahoo!? Jazz up your holiday email with celebrity designs. Learn more....

...drop packets. Here is my implementation of it: IFUP=48kbps tc qdisc add ppp0 root handle 1:0 htb default 20 r2q 4 $addclass 1:0 classid $cMAIN htb rate $IFUP mtu 1492 $addclass parent $cMAIN classid $cEMULE htb rate 8kbps ceil $IFUP prio 4 $addclass parent $cMAIN classid $cAPACHE htb rate 32kbps ceil $IFUP prio 2 $addclass parent $cMAIN classid $cDEFAULT htb rate $IFUP burst 6k prio 1 $addqdisc parent $cEMULE handle 10: sfq perturb 10 $addqdisc parent $cAPACHE handle 20: sfq perturb 10 $addqdisc parent $cDEFAULT handle 30: sfq perturb 10 As you can see, eMule can use all bandwidth a...

...; whats a good bitrate to encode for streaming audio? > >> > >> I think this might be the problem 128kbps isn't going to work over a modem. > >> > >> I think 24kbps is probably the pratical limit for 56k modems. 56k modems might > >> be able to get a 32kbps stream but only under optimal conditions. > > > >I believe he said the outgoing stream was two 128k ISDN lines, which > >from my ISDN memory I remember as actually having a 112k throughput (the > >other channel being used for signalling, etc). Since you can only push > &...

...URL for the streams is at http://www.virginradio.co.uk/thestation/listen/ogg.html. Ices2 takes slightly different parameters to the darkice encoder, so I'd be grateful for any feedback on whether the streams sound better or worse. The main change I've made is that our lower bandwidth '32kbps' stream is now broadcasting in mono, which should sound nicer than 32kbps stereo. Also, does anyone know if I should be able to get our streams to appear in the YP directories? http://www.oddsock.org/icecast2yp/test/FAQ suggests that I can't, but then some of the messages in the list archi...

...ow bitrates), the software only needs to be included in the decoder, and technically could still work with older compressed Speex files. I just wanted to throw in this idea, since it works quite nice on WMA; and for now I'm encoding speech files with 48kbps (speech with background music) or 32kbps (speech as being preaching, bookreading or lectures). I wish to see that number going down though. If Speex performs better, has smaller files, and can compete with 32kbps WMA9 encoded files, I might become interested in testing out the product, and perhaps switching over! _______________________...

Hello, We're streaming radio programs at both 128kbsp and 32kbps, but only archiving the 128kbps stream to save storage space. I'd like to give users a similar choice of bitrates when they request an archived stream (served through icecast's /file/ functionality). Is there a way to change the bitrate on the fly, or do I really need to save archive both b...

> behavior you are seeing....(that also is a difference between iceast1 and > icecast2 - no server-side buffering in icecast2 and there was in > icecast1)... What do you mean stream buffering? We never buffered in either version that I remember. If you fall behind, you get kicked. It's always been that way. jack. --- >8 ---- List archives: http://www.xiph.org/archives/