similar to: icecast sound compressor

hm, to have a look at Pure Data and external called oggcast~ http://www.akustische-kunst.org/puredata/ is recomended, you can set quality/bitrate/samplerate ... while streaming then, and do whateweryou want to your sound before sending it to icecast, including building your compressor-limiter. cheers Ales Zemene -- http://ales.mur.at irc.kunstlabor.at #kunstlabor citation of

Hi, I searched the Internet to find an answers, but I didn't find anything useful, so I'm turning to you, maybe someone has the answer. I want to make a realtime broadcast from a Linux box. The source is the soundcard's line-in, and it sends the stream to an Icecast server. I would like to have realtime compressor/limiter functionalities on this Linux box, so the outgoing signal

On Wed, 2002-07-17 at 15:34, Josh Coalson wrote: > > Have you seen the API changes in 1.0.3? Now all metadata is > parsed and at each decoder layer you can specify which blocks > get passed up to the metadata callback. See the > *_decoder_set_metadata_respond/ignore functions. > > Also, on the encoder size, you can now pass a list of arbitrary > metadata blocks to

Hi Guys, I was wondering what the idea behind the "Key Frame" settings are in the VP3 codec settings. As far as I could ascertain (from browsing through the source code) the VP3 codec ignores the lpbiPrev and lpPrev members of the ICCOMPRESS structure. My conclusion is thus that each frame passed to the codec is compressed individually and that the redundant information based upon

Its been a while since I was discussing a SoundFont compressor based on FLAC. I've recently implemented the compressor using an application metadata block with the ID 'SFFL' that I registered, which contains my own header and a block of zlib (gzip) compressed SoundFont info. The audio chunk (a block of consecutive 16 bit signed samples separated by 46 zero samples) is then encoded with

Good day! I'm creating HDL IP CORE (for using in FPGA) for theora encoder (now only I-frames). I don't undestand one moment. Now i develop such stages: 1. From RBG(byer) to YCbCr converter 2. DCT processing (8x8 pixels blocks) 3. Quantizator of DCT coeff. 4. Zig-Zag of quantized DCT coeff. and now i have uresolved last stage of compression - how i must send 8x8 blocks to huffman

Hi Christian, Aaron Wolfe has already sent me a script he used for his online radio. I only need to customize it -- and maybe create a Debian package from it. It knows much more than I need (requests, cancels, more stations, etc.), I'll have to cut out some parts. To answer your questions I use Debian Linux, I'll use ices2 as source, not a media player. Thanks, Jacint Christian

>Fromwhat you describe, your comp/limiter can't possibly be working correctly. It should be the last unit in line before the sound card, and needs to be adjusted properly. You also need to balance the levels on your mixing board (so that the correc t level comes at predictable place on the slider). It might be worthwhile to find someone with some sound-mixing or radio engineering experience

Hello all, I have subscribed to this list hoping I might find someone who can suggest a fix for a problem I'm experiencing with Icecast (or possibly something else in my setup). Here is my configuration: Foobar2000 is running on a Windows XP PC. It has the Oddcast plugin and it is streaming to a Linux machine that is running Icecast Server 2.3.2. Foobar2000 is using the Crossfade,

Stefan Neufeind wrote: > Could you be a BIT more precise? If you use SQRSoft crossfading, > doesn't it work as desired with the normal Oddcast DSP? And if it > does: Where's the problem with using the Null output plugin? You can only select one output plugin in Winamp and that needs to the the SQRSoft one, not the Null output plugin. The audio needs to pass through the

> we have a web-based station running liveice and aumix and the levels are all > over the place. is there a way to do automatic gain control on the soundcard > input? > > -peter Run the signal through a Compressor/Limiter before sending it to your soundcard. I use Behringer Ultra-Dyne Pro DSP9024. Very nice. If you want to buy it look here:

--- Josh Green <jgreen@users.sourceforge.net> wrote: > I'll let the list know when SF-FLAC is finished, should anyone be > interested. I'm considering starting a FREE SoundFont compressor > campaign, so users will stop using SoundFont encoders that aren't > available on platforms such as Linux. This would also pressure some > of > the existing formats to release

Hi all, I'd like to launch a radio station that would show the current, previous and upcoming tracks on a dynamic website. Is there any software around that would do just this task, or do I have to create my own scripts? The source (ices) is on the same computer where the icecast server is. Thanks in advance! Yours, Jacint

Sounds like you need to fix the problem at the source first: balancing the levels going *into* your mixing board so that they're not all over the place coming out. You'll never fix that with any hardware or software device. I use a program called ecasound to do software dynamics processing, but you'd have to hack liveice pretty extensively to use it in that situation. Software

--- Josh Green <jgreen@users.sourceforge.net> wrote: > Its been a while since I was discussing a SoundFont compressor based > on > FLAC. I've recently implemented the compressor using an application > metadata block with the ID 'SFFL' that I registered, which contains > my > own header and a block of zlib (gzip) compressed SoundFont info. The > audio chunk (a

Hi, It's going to depend a bit on what format you're going to use. If you plan to stream in ogg vorbis, you can use Ices 2.x and accept PCM via standard input. This means you could run a sox effect or ecasound to get and compress the sound from the soundcard before sending it on to ices. Ecasound can make use of LADSPA plugins which opens you to a range of compressors, such as the

Using the latest dovecot-stable (20050213) with pine, I get an empty "from" header in my e-mails. This doesn't happen with 1.0.59-test, nor did it happen with the first dovecot-stable (20050131). A partial connection log is below: 1.0-test59: IMAP DEBUG 15:23:32 2/13: 00000007 FETCH 1:41 (UID ENVELOPE BODY.PEEK[HEADER.FIELDS (Newsgroups Content-MD5 Content-Disposition

Hi, We're planning to launch a video stream. We do not have anything yet (no camera, no computer, no software, only a Linux server that will do the broadcasting with icecast installed). Has anyone ever done such a thing? What hardware do you suggest? What software can be used (both for Linux and Windows on source side). Can Icecast be used for this video streaming purpose (on server side)?

Hi, I'm planning to broadcast live stream from a place, where there's only ADSL connection, which may be used during broadcast, so temporary bandwidth-sortage can happen. Is there any way to create a huge buffer on server-side? This way there would be no problem, if the source couldn't send the stream, the server could serve from the buffer until the source gets bandwidth again.

_+icecast@sucs.org wrote: > Ices, Darkice, most of the normal stream creation bits don't need X11 > Do you mean normalisation? I mean dynamic compression. > WMA is Microsoft only so you will probably need to use Microsoft > software to do this. > > But if you already have MP3 I realy don't see the reason for WMA, as > anything that plays WMA is likely to also