similar to: Downsampling mp3 on-demand streams

This does sound like a grounding problem. My (limited) understanding is that when you have 2 audio components plugged in to different outlets, there can be a voltage differential between the ground on the 2 devices, which will create an electrical flow between the 2 components, which is the source of the hum. I have had this problem before and solved with a ground loop isolator, (from Radio Shack)

Sorry, here is a link to more information. http://www.epanorama.net/documents/groundloop/index.html -----Original Message----- From: Benninghoff, John Sent: Wednesday, April 24, 2002 2:28 PM To: 'icecast@xiph.org' Subject: RE: [icecast] stream sounds like poo for no apparent reason <p>This does sound like a grounding problem. My (limited) understanding is that when you have 2

On Wed, 24 Apr 2002, Alex Walker wrote: > On Tue, Apr 23, 2002 at 01:16:34AM -0400, jim wrote: > > here at wkdu.org our minds are blown trying to figure out what our > > problem is, we are using a dell optiplex gx1 with a soundblaster live > > card. we are getting a nasty hum on the line in. the thing is we plug > > a cd player directly into the soundcard the hum goes

It is commonly said here that if I want to make AM radio-quality stuff at very low bitrates, a good way is to downsample. I downsampled a song to 11025Hz mono and encoded with -q 0, the result is about 18kbps and is at least radio quality. The downsampler I used is from Edinburgh speech tools, named ch_wave. `sox' performs terribly, so I didn't use it. However, I heard some unpleasant

I guess that answers my question :P -----Original Message----- From: Hathaway, Shaun Sent: Wednesday, March 13, 2002 8:24 AM To: icecast@xiph.org Subject: RE: [icecast] ideas for on-demand streaming <p>How do I submit an email to icecast? -----Original Message----- From: Gary Major [mailto:gr_major@hotmail.com] Sent: Wednesday, February 20, 2002 1:34 PM To: icecast@xiph.org Subject: Re:

Dear Vorbis Gurus to whom I owe a debt of gratitude for creating such a kick-ass audio encoding scheme: I'm probably using the wrong term of "downsampling" here, but here goes. I remember reading about Vorbis being designed with streaming in mind. I was wondering if one aspect was to allow easy ad-hoc downsampling (e.g. going from 192kb/s to 128kb/s) without re-encoding. Does such

Jack Moffitt wrote: > This isn't good enough. Just rip out lame's downsampling code (or > sox's) and use that (as long as your also under a compatible license). DarkIce is GPL, should be OK. > Those are both pretty good. Averaging every two samples just doesn't > cut it :) I thought so :( But I wouldn't need to rip out the code, if vorbis supported

Hi, I am looking for ways to donwsample one-dimensional vectors. For example, x=sample(1:5, 115, replace=TRUE) How do I downsample this vector to 100 entries? Are there any R functions or packages that provide such functionality. I did find the zoo package and the aggregate() function, but these appear to be rather specific for time-series. Thanks in advance, Jan

Hi, For transforming stereo to mono, averaging is fine and that's what everybody does. For sampling rate conversion, it's another matter (too long for this email) and you should read a bit about it a perhaps grab a library that does that. As for echo cancellation, it will be less complex (and as good) on a (cleanly) down-sampled signal (and certainly not on stereo). Jean-Marc Le mardi

Ok, this is slightly offtopic, but relates to the quality of input for speex :) I'm working on echo cancellation by means of sampling the wave mix of the sound card as well as the microphone. I originally had two sound cards, which had some synchronization problems (now solved, more or less), but I have also discovered a much better solution using ASIO 2.0, which enables me to sample

On Sat, Nov 17, 2001 at 06:57:46PM +0100, Maroy Akos wrote: > On Fri, 16 Nov 2001, Samuel Hathaway wrote: > > > I agree! Also, something I've been looking for is a way to pull sound from > > the dsp device at 44kHz and then downsample it to 22kHz for one of my two > > streams. Ideas? > > DarkIce does this already. What it doesn't do, is to have mono and

Thomas was gracious enough to provide a second stream of his beats.dk vorbis broadcast. http://vorbis.arkena.com:8004/test.ogg is 128kbps http://vorbis.arkena.com:8004/test_64.ogg is 64kbps Now even us lowspeed dsl people can enjoy :) Everyone should give a listen to the 64kbps and see what they think of the quality (it's being reencoded from 128kbsp Ogg). jack. --- >8 ---- List

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,

Hi I have data that is sampled (in time) with a certain frequency and I would like to express this time series as a time series of a higher (or lower) frequency with the newly added time points being filled in with NA, 0, or perhaps interpolated. My data might be regularly or irregularly spaced. For example, I might have quarterly data that I would like to handle as a monthly time series with

Samuel, I had the same problem. I think it's because the port in darkice does not match the port icecast or shoutcast is listening on. I thought it was 8000 but it was actually 8001. Check that. Matt -----Original Message----- From: owner-icecast@xiph.org [mailto:owner-icecast@xiph.org]On Behalf Of Samuel Hathaway Sent: Friday, October 05, 2001 2:43 PM To: icecast@xiph.org Cc:

On Fri, 2001-11-16 at 19:29, Samuel Hathaway wrote: > On Fri, 16 Nov 2001, Jerome Alet wrote: > > > one thing that would be nice in DarkIce would be to allow the user to pass > > specific reencoding options for each server, e.g. DarkIce could acquire > > the audio in stereo and send it to a server in mono and in stereo to > > another server, which is AFAICT

On Fri, 16 Nov 2001, Jerome Alet wrote: > one thing that would be nice in DarkIce would be to allow the user to pass > specific reencoding options for each server, e.g. DarkIce could acquire > the audio in stereo and send it to a server in mono and in stereo to > another server, which is AFAICT impossible today. I agree! Also, something I've been looking for is a way to pull

I have been following flac development for some time now, but with the release of 1.0.4 I thought I would finally start the task of encoding my entire cd collection into flac format. After looking through the list archives and the webpage I still have a few unanswered questions. 1. What is the best method tagging method to use? Josh mentioned that id3v1 was preferred over id3v2 since it

Hi To continue the interesting streamTranscode thread but in a slightly different direction, I have a user station that would like to send us a 64kbps AAC stream and have us transcode that to 64kbps MP3 and 32kbps AAC - is this possible? Points to consider: a) we would be happy to go with another permutation such as: she sends us 64kbps MP3 and we convert that to 64kbps AAC and 32kbps ACC. b)

> behavior you are seeing....(that also is a difference between iceast1 and > icecast2 - no server-side buffering in icecast2 and there was in > icecast1)... What do you mean stream buffering? We never buffered in either version that I remember. If you fall behind, you get kicked. It's always been that way. jack. --- >8 ---- List archives: http://www.xiph.org/archives/