similar to: icecast sound compressor

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

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

At 11:08 AM -0500 7/12/04, Steven Critchfield wrote: >[snip] > >Would it be something people would like to be able to add filters to a >line? Consider normalization as a filter. Monitor could then be moved to >a filter as well. Echo cancel could be a filter. Set it up so multiple >filters could be added and chained together. This could help those with >echo chain a couple of

Hi: I want to stream audio from my soundcard, but I'd like to apply some dynamic compression to it first (ala winamp's audiostocker plus shoutmuxer thingy). I heard rumours of compression in liveice, but Iv'e been told that this only works when it plays from prerecorded MP3, not the live input. I'd like to do this with darkice, but can't see how it could be done. I guess

On Thu, 4 Apr 2002, Akos Maroy wrote: > can you tell me more about these LADSPA plugins? LADSPA stands for Linux Audio Developer's Simple Plugin API (see http://www.ladspa.org/). Basically, it was pointed out on the linux audio dev (LAD) mailing list that numerous programs were using plugin architectures and all were different. So they fleshed out a plugin API and the rest, as they say,

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

On Thu, 4 Apr 2002, Jack Moffitt wrote: > While dynamic compression makes the transmission 'smaller' on FM radio, > it makes music harder to compress by psychoacoustic codecs, and therefor > you might have somewhat of a quality hit, although I haven't tested > this. Quite possible, though I'm only using 56kbps 22khz stereo anyway so that's not such a big deal. I

At 11:40 AM 11/14/2001 -0500, sublime@mac.com wrote: >the ACG function that the compressor provides is limited at best. its >adjustment is way too audible to make full use of it in balancing levels. >yes, it is last in the chain. i'm just wondering if there's anything out >there in the way of software. if a minidisc recorder can do it, why can't a >$1000 pc? tons of

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

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

hello, I've just got ices-kh60 running with jack. It's working OK apart from a problem where the latency on the stream creeps up from about 15-20 secs to around 7 minutes. The stream plays OK, but the audio coming out is 7 minutes behind what goes in. I'm not sure how this can be happening, seeing as I don't think there are any buffers that long in the system, but it is.

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

hi all i want to provide audio recording facility with my Ror application , which can be stored in DB at server , for later use can anybody tell, how to implement this with RoR, as i dont hav much idea abt that thanx --------------------------------- Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. -------------- next part -------------- An HTML attachment

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

> On Dec 17, 2009, at 8:50 PM, dathead2 at gmail.com wrote: > > On Dec 17, 2009, at 8:42 PM, Ron Decline <rutlecorps at gmail.com> wrote: > >> Is it possible to switch the Left / Right channel when encoding in FLAC? >> (I have some flac files with incorrect left/right channel assignment that I would like to re-encode with flac) >> _____________________________

>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

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 all: I am managing a radio station that has 7 streams. I am using icecast, ices and ezstream on the streams and overall it is working pretty well. One of the streams is an old time radio stream and I use ezstream to run the prerecorded shows. The streams are 128K mp3 streams. Also this station is running on a VPS using Ubuntu 14.04. My question: Is there a package or some sort of

How are people doing gain control, out of curiosity? I know boxes exist (anyone have names and mfrs?) that ride the gain for big radio stations, ensuring that there is no distortion and raising the volume of songs recorded at a lower volume. I'd probably buy one if I knew what to buy. ------ Dave Hayes - Consultant - Altadena CA, USA - dave@jetcafe.org Now Playing at:

I am looking for a command-line driven audio splicer for .wav files. Any ideas? todd@toolz.com --- >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 needed. Unsubscribe messages sent