similar to: 24k, 56k, and 96k, is it possible?

I have tried changing the sampling rates. I have tried the two you mentioned below. Still no luck. Plus, if I disable one of the streams (either the 56 or the 96) the 24k stream works fine. Is LAME having issues doing 3 streams? Thank You, Mike <p><p><p><p>On Fri, 14 Jun 2002, Geoff Shang wrote: > Hi: > > The reason 24kbps sounds bad is in part due to the fact

Hi: The reason 24kbps sounds bad is in part due to the fact that LAME has annoying defaults for that rate. Using mono, it defaults to 16khz. Stereo it defaults to 8khz. It can cope with 22.05khz and 11.025 khz respectively just fine at those rates. I've not used liveice, but if it's possible to specify the --resample command line option with either 22050 or 11025 as its argument

How does one unsubscribe too? I don't stream audio ever since that fuckin' CARP ruling... -Mike <p><p>On Fri, 26 Jul 2002, Dave Yerrington wrote: > Does anyone moderate this list? what is up with developers too? It seems > like they dont care either about this project :( > > > ----- Original Message ----- > From: "COL. KENNETH ABELANGE"

FROM: COL. KENNETH ABELANGE. DEMOCRATIC REPUBLIC OF CONGO. Tel No: Your country Intl. access code +873762692483 Fax No: your country Intl. Access code +873762692485 kennethabelange@africamail.com Dear Sir/Madam <p>SEEKING YOUR IMMEDIATE ASSISTANCE. Please permit me to make your acquaintance in so informal a manner. This is necessitated by my urgent need to reach a dependable and

Hi, Maybe a bit off topic for this list, bt anyway. I have received several feature requests for DarkIce to support downsampling of the audio input before passing it to lame or ogg vorbis. For example the audio read from the soundcard would be 44.1kHz, and lame would get it at 22.05kHz. I figure two ways of doing this: 1. For lame, one can specify the input and the desired mp3 sampling rate,

yep, they simply forward the bitstream to icecast. And in general, bitrate changes are handled by most listening clients, although very few I have found (if any) can handle samplerate changes appropriately. oddsock At 10:03 AM 9/14/2004, you wrote: >You mentioned these programs and their "no-reencoding" mode. Can they >handle collections of MP3s of different bitrate? >

Hello, I've got two streams, one for broadband, one for dialup. Well, having had occation to use a dialup connection recently i checked the dialup stream. Although it was streaming what the broadband stream was, the audio quality was audibly worse. It didn't buffer, but it didn't sound as clear as the broadband stream. I used lame to encode the tracks to mp3 and used it's

Ok, is this possible: I want to have a 128k and a 24k stream of a particualr audio program, plugged into the line in of my Ensoniq AudioPCI 128 (es1370 chipset) Is there any way to do this with just one soundcard, or do I need two? Thanks Scott W --- >8 ---- List archives: http://www.xiph.org/archives/ icecast project homepage: http://www.icecast.org/ To unsubscribe from this list, send a

Hello All, I'm a newbie in icecast and I'm trying to put it runnig nice, (please, be patient with my english) I got it running with shout, with a big directory of mp3 in my playlist, but after few seconds of transmission it begins to fail. My mp3 files are all 128 at 44. May be the bandwidth to low?? Other question is how can I set a low stream to dial conections of 56K with

Hi Justin, I have a suggestion, but it may not be very convenient. You could try converting the 24/96 BWF to AIFF, and then use flac to compress the 24/96 AIFF. There is no difference in audio quality between the FLAC file generated from BWF (WAV) vs. AIFF, so perhaps this extra step will solve your problem in the short term. Sorry I haven't used BWF or WAV very frequently, but

Hello, I have a matrix with 6 columns and 12 rows. I want to extract out those IDs (rownames) from my matrix which have a negative values. For each ID(row) if the negative value is even under 1 column it needs to be extracted out. I will be grateful for any correct suggestion. Thanks Manisha Here is the matrix that I am working on: ID A B C D E F 1 -4.18972 -3.8946

Hi, I am attempting to use flac to encode 96k/24-bit broadcast wav (BWF) files. BWFs are wav files with some extra meta-data chunks, and is the favoured archival format for many institutions around the world. These files are encoded successfully by flac, however the resulting flac file is not playable on all flac players - it plays successfully in foobar2000 but is silent in winamp, and when

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

In this topic on Hydrogen Audio(http://www.hydrogenaud.io/forums/index.php?showtopic=105586) someone asked a question about the sample rates that FLAC supports for ReplayGain. The outcome was that the current documentation of MetaFLAC is outdated since Commit http://git.xiph.org/?p=flac.git;a=commit;h=0554a4aee6966bc5b251364753ef85de72dfab19 because as of 1.3.0 FLAC supports Replaygain with many

Hello everybody, I've got a 56k usb modem, lsusb says: Bus 002 Device 002: ID 0572:130 Conexant Systems (Rockwell), Inc. I'd like to let it work with Asterisk. I think that I should use chan_modem and/or chan_modem_bestdata, but I found little or no documentation. Can anybody please post some instructions? Thanks in advance, -- Dr. Andrea Spadaccini Multimedia Technologies Institute

Hi all, I'm trying to use the FLAC C libraries to encode audio. I'm doing something like: FLAC__seekable_stream_encoder_set_channels(pflac->fse, 1); FLAC__seekable_stream_encoder_set_sample_rate(pflac->fse, 11025); FLAC__seekable_stream_encoder_set_bits_per_sample(pflac->fse, 8); if ((bps = FLAC__seekable_stream_encoder_init(pflac->fse)) !=

Hello, Please excuse my complete xen ignorance but I am hoping someone will be able ot help me out here. I recently built my first Xen box using a 2, 1TB drives in a raid configuration, 2TB total. I installed Xen, I am able to access it via XenCenter. However, I am trying to use some of that 2TB as a local datastore but for the life of me I cannot get it mounted, I keep on getting this error:

Hello, First of all, thank you for developing a fantastic IMAP server! I've just upgraded to 2.0.4 (from 2.0.1) and when I try to run /usr/lib/dovecot/deliver (either directly from the commandline, or from within the MTA) I get the following error: doveconf: Fatal: Error in configuration file /etc/dovecot/dovecot.conf: ssl enabled, but ssl_key not set I converted my old config file

Anybody have suggestions on having a 56K dialpool and VOIP connections with an Asterisk box over the same set of PRIs? We've done the PM3 with PRIs for just dialup, but are looking for a way to integrate our Asterisk box and move our voice calls onto the same PRIs. Ian -- Ian White Victoria Free-Net Association email: iwhite@victoria.tc.ca http://victoria.tc.ca/