similar to: Low-bitrate audio encode/stream application

Hello Thomas, You are correct - the application is basically for secure remote access to ham CW (morse code) station. I believe the solution will be based on low bit rate MP3 streaming for receiver side (MP3 allows use with existing CODEC in client side, which could include Win CE or Linux palmtop type form factor, and the audio source would just be "voice grade" material on a mono,

On Wed, Dec 22, 2004 at 02:28:18PM -0700, david feldman wrote: > I am totally new to icecast, and would appreciate a pointer in the right > direction. My application is to encode audio (at the linux server) at a low > bit rate, say ~16 kbps, for transmission to a decoder at another location. > The audio source will be bandlimited (basically voice grade circuit such as >

Question from new subscriber - I'm working on a project to connect to remotely connect to a short-wave receiver via a dial-up PPP/IP circuit. Turns out the dial-up circuit is only stable (useful) to 14.4 kbps (faster modem training produces so many link errors that the net circuit quality is unusable - one end is in a remote, rural location), so looking for codec that can fit within this

I am writing a long-run sound recording application using OggVorbis. The user can adjust sound quality parameter to make balance between storage 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

dear vorbis developers, it would be very useful, if oggenc would support constant low-bandwith bitrates for testing. something from 20kbps to 44kbps, so anyone could test streaming over modem and isdn. audio quality doesn't matter. mörk --- >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

Hello- I recently downloaded Verbs and started playing with the encoder. The quality was great when I encoded a few samples at medium and high bitrates, but when trying to encode the same clips at 16 kbps, artifacts are audible during playback- short bursts of screeching noise, not just distortion. I'm using oggenc 1.0 beta 4 to encode, sonique to play back on Windows NT. I've also

Hello, the voice-communication TeamSpeak (www.teamspeak.org) is currently testing a version that supports speex codecs. The quality for high bitrates is quite good. BUT, the low-bandwidth speex codecs that are currently used arent very good. What I did to find this out: I comprared a speex AVB with 6.3 KBit/sec (total, overhead for packets and stuff included) and the 6.3 Kbit/sec Celp Codec

In testing, I have found that both MP3 and RealAudio 3(!) are much better sounding at bitrates from 8 to 40-something kbps than Ogg Vorbis. This definatly needs drastic improvents and I know that 1.1 will be focusing on improvements in those areas but that seems pretty far off(a few months minimum) which leaves low-bitrate streams sounding pretty bad. --- >8 ---- List archives:

Hi, all. I'm new to this list, and I know that my question touches on a FAQ. I'm just hoping to get more information than "it's a priority item, but it's not done yet." (Basically what the FAQ says.) I've been encoding about 20 minutes of mono audio each week to as small a file as I can, so it can be served via html. As long as the result is understandable,

Am Dienstag, 13. Mai 2003 03:22 schrieb Allen Drennan: > Hello, > > HawkVoice doesn't have a 6.3kbps codec for CELP, it has a 4.5kbps CELP > codec and I do not believe it is being used by TeamSpeak. The 6.4kbps CELP > being used in TeamSpeak, to which you are referring I believe comes from > Lernout & Hauspie's LHACM.ACM file which it appears you are redistributing

Hello, I'm not sure if this list is the proper place for this post; please correct me if it is not. I am attempting to use libtheora and I have been looking through the examples in the distribution Excerpt from encoder_example.c -- case 'V': video_r=rint(atof(optarg)*1000); if(video_r<45000 || video_r>2000000) { fprintf(stderr,"Illegal

Hello everyone. I would like to know which are the available bitrate using the ultra-wideband compression. Thank you! Paolo Gruppo Telecom Italia - Direzione e coordinamento di Telecom Italia S.p.A. ================================================ CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you

the Vorbis FAQ says: "mid to high quality (8kHz-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel." What is the difference between Hz and bitrate? Doesn't MP3 support higher bitrates? Pointers for more reading are welcome.

Hi. Sorry if this is the wrong list - a pointer to the right would be appreciated. I have a number of Ogg files, that have a VBR bitrate of approx. 256 kbps. I have bought a portabel Ogg player (iRiver iFP 795) that unfortunately only support bitrates up to approx. 225 kbps. So my question is, if there is a tool to change the bitrate down to e.g. 220 kbps. I know I can do this with sox, or

Hi, How can I broadcast the same source in 128 kbps, 64 kbps and 24 kbps ? Regards -- EISELE Pascal <lemmingsml@nerim.fr> --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a message to 'icecast-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Is there any way to bring the bitrate in ices2 down below 32 kbps? I'm setting up a demo for someone of how to use linux to do net radio broadcasting. The setup I'm thinking of is to use ardour plus jack to mix two (maybe more) input sources (live audio and recorded tracks/programmes), then send the mixed audio to ices2 for streaming to icecast2, using the jackified version of ices2. This

Hi, I've been testing out the speex narrowband codec at low data rates (using linphone and Counterpath's Eyebeam). I'm finding that at data rates of ~25 kbps, the quality of the voice call is very poor. I know speex is supposed to work at much lower data rates (~2 kbps). Has anyone verified that speex will produce reasonable quality at low data rates? Are there any existing

On Sun, 29 Aug 2004 17:53:29 -0700, Ralph Giles wrote: > On Mon, Aug 30, 2004 at 03:03:28AM +0100, Andy Baxter wrote: > >> Is there any way to bring the bitrate in ices2 down below 32 kbps? > > Generally the trick for this is to downsample the audio before encoding. > You can ask ices to do this with a resample stanza in the config file: > > <resample>

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best