similar to: Best way downsample stream from 128 to 56 on the server?

Hi, Now that I've been able to look at streamtranscoder 3.0, I see there's absolutely no documentation whatsoever. I've not compiled it, but I note that previously when you ran it first time, it would create a sample config file for you to edit. Maybe it still does this. Oddsock's flash demo may shed more light on this, but I can't use it so don't know. Geoff. --

Hi all, First question on this list - I am running into issues with my embedded OGG stream players (so far I've used ffmp3 and just today started trying jlgui), and honestly I'm not sure what's up. I'm hoping someone would be kind enough to help me figure it out. Basically, I have a setup that pulls a number of OGG files from MPD (all the same bitrate/etc AFAIK) In ffmp3, when

On 20 May 2011 05:30, Jack Raats <jack at jarasoft.net> wrote: > Hi, Hi > At this moment I'm running a stream on 128kbps. > I want to add the same stream on 24 kbps. > Can this be done on the server using icecast? > Or do I have to use another programm. The streamTranscoder is a multi-platform utility which can be used to transcode media streams from one format to

We are running an icecast2 server in conjunction with our LPFM radio station and need a little advice. We have been serving a single 128k ogg stream from a remote server and would like to make additional formats available (e.g. mp3, lower bandwidth, etc.) I would think this has been done a few thousand times and there would be a "best practice" but I can't find it. Our basic

What is everyones opinion and expierience with using ODDSOCK streamTranscoder to downsample mp3 streams? I tried it on Windows 2000 and then compared it against shoutCAST DSP plugin. It sucked. It skipped and popped and cut off the first 1/2 second of each track and dropped. It sucked, while shoutCast DSP was flawless. What attracted to me to streamTranscoder was you don't have to run an MP3

hi, so far I'm very happy that I have running an icecast server with streamtranscoder which autimatically transcodes a certein mountpoit (ogg-high-quality) to low, mid and high mp3 stream on-connect. but I'm stuck with two problems: 1. how to determine which stramtranscoderv3 process is reencoding this particular stream? because doing <on-disconnect> just # killall

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

This is all happening on a box with: Win 2k SP3 512K RAM 1.2 GHZ Celeron The PC has noother load...all its doing is downsampling an MP3 stream to lower bitrates. I'm using the LAME_ENC.DLL with streamTranscoder. There is no version number tagged on the lame_enc.dll, but the file is dates July 2002. When I refer to "skipped and popped and dropped" I am referring to your

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,

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

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

thank you, for your fast answer. my problem with ogg is, that the mediaplayer don?t understand it. a lot of users are using ms mediaplayer unfortunately. greets Geoff Shang schrieb: > Carsten Henkel wrote: > >> we would like to use IceCast 2.3.1 as a mp3 streamer source for >> reencoding several sources of ices 0.4. >> >> It is working perfectly already by using

that's right. i can make those fine. i have no problem connecting to the admin interface through a shell, or using the encoder password or anything. that's not the problem. the problem is that when i go to http://server.com:8000/admin?mode=sources in a web browser a dialog box comes up that says +------------------------------------------------------------------+ |

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

Hey Rado, Ack... Yep, I'll give that a shot. Can't just shut down the streams at any given time. This may have to wait for testing. The friend I am helping set this up has decided the quality of Ogg Vorbis FAR outweighs the benefits and performance of MP3. Plus, there are the licensing issues involved with MP3. Any ideas on how to do set up a dual stream, one for low bandwidth users and

Hello together, we would like to use IceCast 2.3.1 as a mp3 streamer source for reencoding several sources of ices 0.4. It is working perfectly already by using an mp3 playlist with local stored mp3s. But our plan is to upload an 160 kbps stream and want ices 0.4 to reencode to 96, 56 and 32 kbps. Uploading 1 getting 4 streams. We are streaming with SAM Broadcaster 2. Icecast runs on SuSE 9.3

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

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 have a single stream going to my icecast server, encoded in hi-quality ogg. goal is to serve hi- and low quality ogg and mp3 streams, re-encoded from the single hi-quality ogg stream. encoding all from e.g. oddcast simultaniously is not an option. it would be easier if source-streams would be static and always named the same .. unfortunately they are not. if so, I could define simple

Hi folks I'm running streamtranscoder on my icecast server. Unfortunately, after 5-15 minutes of transcoding, it sometimes produces jumpy/skippy audio or drops the stream altogether. What I'd like to do is restart streamtranscoder every 5 minutes. However, when I kill streamtranscoder and bring it back, all listeners on the stream get killed. How can I prevent this? In other words,