similar to: nominal bitrates

Hi all! I installed icecast & ices (from CVS) Enerything runs, but with some problems: When I connect to the server (with XMMS or MPlayer) the buffer runs empty very quickly, causing interruptions. My "server" is an old Pentium 120 with 64MB RAM ,a HD of 8G and an ethernet NIC. (I know it isn't much, but I'm tired of taking my CD collection to work.) I ripped the tracks in

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.

I'm tentatively creating a Windows .dll file using the Vorbis tools demos as a base for it. I've taken oggenc and simply converted it into a .dll file, with some minor changes purely to pass parameters to it (the original uses the command line). I have it working as far as it goes, but there seems to be something going wrong deep inside that I don't understand. The line at fault

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, Is it possible to generate an ogg stream at less than 22KBps ? I tried several configuration with ices and darkice, but I always get an configure encoder error: [2003-03-17 16:07:17] INFO input-oss/oss_open_module Opened audio device /dev/dsp at 2 channel(s), 44100 Hz [2003-03-17 16:07:17] INFO audio/resample_initialise Initialised resampler for 2 channels, from 44100 Hz to 22050 Hz

module. The file may be corrupted. Message-ID: <Pine.LNX.4.43.0406081110070.26162-100000@pilchuck.reedmedia.net> I am unable to pipe to ogg123 with cat, but redirecting with < works fine. rainier:~$ cat ~/audio/acdc-who_made_who.ogg | ogg123 -v - Audio Device: OSS audio driver output Author: Aaron Holtzman <aholtzma@ess.engr.uvic.ca> Comments: Outputs audio to the Open Sound

Greetings: Another test. I've re-encoded a batch of mono 22050 Hz WAVs, average bitrate seems to hang around 30-40 kbps. Icecast.xml is set for three connects, and the URL is here: http://64.108.112.145:8000/shouter.ogg IMO, the sound quality is much better now. Again, the stream should be up for about 45 minutes or so. Please let me know how it works for you... Best regards,

Greetings: Another test. I've re-encoded a batch of mono 22050 Hz WAVs, average bitrate seems to hang around 30-40 kbps. Icecast.xml is set for three connects, and the URL is here: http://64.108.112.145:8000/shouter.ogg IMO, the sound quality is much better now. Again, the stream should be up for about 45 minutes or so. Please let me know how it works for you... Best regards,

Hello I just wrote a little application using speex. I am compressing and packing singel frames to transmit them. Now I read in the manual, that it's possible to achieve higher bit rates by packing more than one frames at once. Unfortunately there is no table included at the manual, that shows the "real" bitrates of ultra wide band mode. So my question is: Can someone provide that

On Sun, 2003-11-23 at 03:51, Kerry Cox wrote: > Hmmm, I guess more is incorrect here than I thought. I changed the > resample in-rate to be 44100 like it should have have been. > But still no joy. Here is the error message as shown in the error.log > file. I'm looking things over and am not seeing my error. > It looks like the audio resample for this particular stream is good.

I'm having a issue with build libThusnelda 20081201,video is blocky, you can clearly see video is blocky in some parts of the video. the Bit rate is high so what is the problem is, on x264 this type of Bit rate will make the video very smooth. can anybody tell me how to get that smooth video and what settings can i use, I'm interested in the advanced usage of Thusnelda with two pass

> Page 3: > > To be compliant with this specification, implementations MUST support > 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate. > The sampling rate MUST be 8, 16 or 32 kHz. > > There is a type above after (narrowband), there is a " extra character. > > I don't understand what is the motivation to specify "SHOULD

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

Hello, thank you for responding. I'm sure I'm confusing something, but I did not explicitly set either kHz or kbps, I just set the mode to narrowband, and traced through libspeex's code to see that sample_rate was 8000 something, I'm guessing Hz. My assumption was: if you set PA to nb-mode, then you should encode 160 bytes to 21. Is that mistaken? ys On 6/13/07, Jean-Marc Valin

hello, i have a question about the documentation: oggenc --help says: -b, --bitrate Choose a nominal bitrate to encode at. Attempt to encode at a bitrate averaging this. Takes an argument in kbps. This uses the bitrate management engine, and is not recommended for most users. See -q, --quality for a

What is the best way to set up multiple streams, at multiple bitrates, from the same line-input audio source? I currently offer a 96 Kbps stream of the line-input of a D-Link DSB-R100 radio tuner. I'd like to offer a 32 Kbps stream as well, for 56K modem users. Should I run multiple copies of liveice on the machine with the soundcard? Or should I have ices reencode the 96 Kbps stream down

This mail depends upon the fact that I don't have a couple of good earphones ;-) I read in the site that q=6 is a very high quality, but does it contain perceivable differencies from the original? (for 95% of people, of course). I also found q=6 to produce files slightly bigger (1/10 bigger) than those produced with lame VBR q=2 (about 192 bps on average). I always thought LAME VBR q=2

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

Greetings: Another test. I've re-encoded a batch of mono 22050 Hz WAVs, average bitrate seems to hang around 30-40 kbps. Icecast.xml is set for three connects, and the URL is here: http://64.108.112.145:8000/shouter.ogg IMO, the sound quality is much better now. Again, the stream should be up for about 45 minutes or so. Please let me know how it works for you... Best regards,