similar to: make lo-fi sound as good as RealAudio?

Shawn sent me a copy of the non-Direct Sound version of this output plugin, and I'm hosting it on my web site's rare software page: http://members.lycos.co.uk/bhafool1/rarities/ I've found it adds far more smoothness and clarity to various low sample-rate audio files, including an 80-minute speech I mentioned previously, due to my soundcard's approach to D-to-A conversion

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

--- vorbiscomment/vorbiscomment.1 | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/vorbiscomment/vorbiscomment.1 b/vorbiscomment/vorbiscomment.1 index 0108e78..2bceb83 100644 --- a/vorbiscomment/vorbiscomment.1 +++ b/vorbiscomment/vorbiscomment.1 @@ -39,13 +39,11 @@ Reads, modifies, and appends Ogg Vorbis audio file metadata tags. .IP "-a, --append" Append

After a bunch of cumbersome testing I was able to transcode a Quicktime MOV file to OGG with theora and Vorbis. The method's a bit fumbly, but it works... It is automatable except for the initial audio decoding. Software used: Xine, oggenc, ffmpeg, ffmpeg2theora, oggzmerge Step 1: Get the video out of Quicktime ffmpeg -i infile.mov -b 5000 -f mpeg2video outfile.mpg Step 2: Get the video

On Sun, 2003-11-23 at 00:35, Kerry Cox wrote: > Now I tried to add a third stream by adding the following <instance> to > the above config: > > <instance> > <hostname>xx.xxx.x.xxx</hostname> > <port>8000</port> > <password>secretpasswd</password> >

darkice is an execellent bit of software, very well written. and the author of the darkice software, akos, is very active on the darkice ML. http://yp.tilos.hu:9000/tilos_high.ogg <= for a darkice stream. b. On Sat, Nov 22, 2003 at 12:05:55PM -0500, Brendan Cully wrote: > On Saturday, 22 November 2003 at 10:50, Kerry Cox wrote: > > So basically, I download and install ices 0.3,

On 2006-06-21T15:21+0100 Karl Heyes wrote: > You capture at 48000 but resample from 44100, that will cause problems. > You resample to 22050 so that will affect quality, you haven't said what > bitrate you want to target for, so resampling/downmixing may be required > anyway. I am unsure about these settings. Any particular suggestions? I don't quality with voice, just

Hi all, I often record radio shows for posterity, and sometimes I have friends who would like to listen to them live over the net. When I am recording for my own purposes, I use a command of the form: "brec | oggenc", (options omitted for clarity) and I send the output to a file, call it radio.ogg. Locally, I use Debian stable. If I want to listen to the show as I record it, I can

> I dont care I just wanted to make sure there is no problem 50 seconds is a bit long. Probably your client buffer is really high and you are streaming at a really low bitrate. But even in that case 50 seconds seems long. If you can post more details on bitrate, client used, buffer size used on the client, etc, that would be helpful for verification. jack. --- >8 ---- List archives:

I have icecast2 running on several machines now and have gotten relaying to work just fine. I've posted my instructions on how to do this to my icecast docs site. I have just a few more questions. Perhaps some one on this list has already addressed these issues but has simply not posted these elsewhere. 1) I'm looking for a startup script for launching the icecast2 icecast ices binaries

I'm trying to encode audio tracks as ogg vorbis at very low bandwidths for peer to peer broadcasting. I.e. need to have two streams uploading from modem users, which means the bitrate needs to be around 18-22 kbps. It's not easy to get good quality at this bitrate. The recipe I'm using at the moment is: oggenc Mon-Sep-20-04-0\:3\:27.wav -b 20 -M 22 --resample 17000 --downmix -o

Folks: I'm trying to set up a streaming server under Fedora Core 2 using Icecast 2.2.0. Everything compiles; everything runs. There are no indications of a problem in any of the log files. When I look at http://192.168.0.128:8000/status.xsl, everything looks ok. But when I click on "Click to listen", it brings up xmms, but there's no indication of audio, and xmms shows a bit

Hi, <p>apparently, I didn't delete my test samples ... I found them. :) Here's a small table of what nominal bitrate fits to which quality at what sample rate. Quality levels inbetween -1 and 0 are possible, but I didn't test them. Nominal bitrates of those quality settings will be somewhere between those of -1 and 0 and therefore can be guessed. q/channels || -1/mono |

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

>it supposed to, check the log file for details but there is no reason >for it not to work with 2 instances. > >Note as well, that you should have 2 instances each of which has the >icecast connection details and an encode section. > I don't quite understand. You always talk about two instances. Do you really mean two instances of ices2 called seperately? Do I have to make

Hello, I had a big trouble with ices (ices-2kh60). It is reading a playlist file containing "-" to tell "read stdin". But I think that when stdin is having problem, then ices become mad. It tries hundreds of time per second to play the playlist file content, generating huge amount of traces, which are rotated, but causing a denial of service on the server which become

Hi, Just one quick question. Can ices re-encode vorbis/ogg format on the fly? I know it can do that with mp3 format, but what about vorbis/ogg format? Thanks. Reuben D. Budiardja --- >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

Hi all, I run a CVS version (i dont remember exactly when did i got it )of Ices2/ Icescast2 on a slackware Linux box . I put this section into the Ices2 conf file : [...] <encode> <samplerate>22000</samplerate> <channels>1</channels> <quality>0</quality> <managed>1</managed> <nominal-bitrate>35000</nominal-bitrate>

I am having problems getting Ices to work with ALSA. If I use arecord (arecord -D plughw:0,0 output.wav) I can record the input from my microphone into a file. The playback works perfectly. However, if I use Ices with the same values I hear nothing on the receiving end of my broadcast (although I can connect to it). In order to do a further test, I setup a <savefile> tag in my