similar to: Wav -> Ogg = Chipmunks

Ok. Now it will run. But when i try to play back my stream(either ogg/vorbis or mp3) it sounds like the chipmunks. Could it be that I'm broadcasting at a sample rate of 22050 in mono? This shouldn't make a difference as the source is also 22050 samplerate and mono(its the line-in plugin for winamp). I'm using Winamp 2.81 with the is the Oddcast DSP for Winamp 2.xx plugin to

I'm using ices2's metadata facility to update the name of a track in a vorbis stream: <input> <module>alsa</module> <param name="rate">44100</param> <param name="channels">2</param> <param name="device">hw:1,0</param> <param name="metadata">1</param>

HI! So I’ve been at this for a few days with no luck. I am trying to pre-pend my stream with an intro file (mp3). I’ve saved the mp3 file (for the intro) the same as the stream > in my test right now it is set up stereo, 112kbps, 44100hz. I can play the file independently in a browser from the webfoot location. I can play the stream with no problems. I have modified my icecast.xml to include

I have been thinking about what is needed to make language teaching/learning tools. (Like talking flash cards.) The main thing needed is a low bit-rate encoding of human voice. At first I thought I could take one of the gevernment standard vocoders and embed it as an Ogg stream. But: (1) there is not a standard vocoder, there is are half a dozen, at least. (2) they are fixed bit rate, we

Hi, I'm new here. I would like to write 2 simple utilities to maniputale OGG files: 1) oggjoin --output-file=result.ogg file1.ogg file2.ogg ... Merge file1, file2, ... into result.ogg file 2) oggpause --output-file=result.ogg --duration=0.5 Generate result.ogg with silence for 0.5 seconds oggpause --output-file=result.ogg --input-file=source.ogg Generate result.ogg with silence equal

Hi Folks, First, I would like to thank the eople in development who are doing so much to develope these great video and audio tools. I use Ogg Vorbis to store most of my music projects in studio and the results are stunning. Quite frankly, in the audio file reduction game, nothing beats Ogg Vorbis and I can personally say this from my own experience. So a huge thank you to the team. I've

hello, people! could you advice me some kind of computer audio recording programme to write music 'on-the-fly' from line-in sound card exit to ogg vorbis file directly. for example, there is an 'absolute mp3 recorder', but i would like to record audio in ogg vorbis. -- kind regards, maxim abalenkov http://maniac.times.lv copy writer mail to: vault13@inbox.lv

On 2/28/07, Ivo Emanuel Gon?alves <justivo@gmail.com> wrote: > On 2/28/07, Ralph Giles <giles@xiph.org> wrote: > > Well, there are todo pages at wiki.xiph.org, but I meant more in the > > community folklore sense. My point is a roadmap doesn't help much unless > > there are people committed to making things happen. That's been the > > problem with a

On 2/28/07, Ivo Emanuel Gon?alves <justivo@gmail.com> wrote: > On 2/28/07, Ralph Giles <giles@xiph.org> wrote: > > Well, there are todo pages at wiki.xiph.org, but I meant more in the > > community folklore sense. My point is a roadmap doesn't help much unless > > there are people committed to making things happen. That's been the > > problem with a

If this is the wrong place to post this, please forgive me, and PLEASE point me to the correct place. I am looking to hire someone to write a function to scan a ogg-vorbis stream running on Debian etch and checking for two things. The first is static, the second is "dead-air". The function require no GUI, just read a configuration file for the url to scan and how many seconds of

I?ve successfully implemented IceCast on a Raspberry Pi and can access the stream ONLY if I use an ffmpeg statement to start the stream. Unfortunately, the stream is choppy and the sound quality is poor. I?ve tried using ICES, but I don?t get any sound. Terminal command that works: ffmpeg -ac 2 -f alsa -i hw:0,0 -acodec mp3 -ab 48k -ac 2 -content_type audio/mpeg -f mp3 icecast://source:password

Hey All: I've got a working samba/ldap domain with a PDC in a datacenter and a BDC in my local office. I'm not able to reliably join a windows XP Pro machine to the domain by specifying the PDC as a wins server. I get the following error 90% of the time or more, with no discernible patterns or errors in any logs: --------------------------------- The following error occurred attempting

Hello Ben, The working kernel code I based my investigation was a 1.3.17 + gentoo patch (so yes, pretty old) The new one below, not working is the vanilla 3.19 (from the gentoo repo, should be identical to the latest 3.19 stable) I've narrowed down the issue to the size of the eld. The new patch (that indeed uses drm_eld_size) (with this patch, I have a working 3.19 kernel): I will need to

On 2006-06-21T15:21+0100 Karl Heyes wrote: > You capture at 48000 but resample from 44100, that will cause problems. > You resample to 22050 so that will affect quality, you haven't said what > bitrate you want to target for, so resampling/downmixing may be required > anyway. I am unsure about these settings. Any particular suggestions? I don't quality with voice, just

Hi, I'm trying to record something for about 24 hours but oggenc tells me that I have 203m remaining. Quote: durand at Deuterium ~> arecord -f cd -t wav | oggenc -b 192 -o file.ogg - Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Opening with wav module: WAV file reader Encoding standard input to "file.ogg" at approximate bitrate

I am trying to convert some a few WAV files to ogg format (using vorbis-tools-1.0beta4 oggenc). I tried: $ oggenc -o nash.ogg nash.wav ERROR: Wav file is unsupported subformat (must be 16 bit PCM or floating point PCM ERROR: Input file "nash.wav" is not a supported format And: splay -d - nash.wav | oggenc -r -o nash.ogg - The resulting file when played with ogg123 played so

Dear experts, I am trying to use an example I found here: http://svn.xiph.org/trunk/vorbis/examples/encoder_example.c Could you please tell me what float** buffer is needed for (I don't see it used anywhere). Also, could you please tell me what this code is doing: for(i=0;i<bytes/4;i++){ buffer[0][i]=((readbuffer[i*4+1]<<8)|

On Sat, 7 Jun 2003, Karl Heyes wrote: > your sound card/driver does not deal with 44100hz samplerates, try 48000 > instead. Remember to change the resample tags according. Are there many cards in this category? I'dve thought that all cards would support 44100, it being the standard that it is. Your comments however suggest experience of this problem. Geoff. --- >8 ---- List

John, Thanks for the reply! You mentioned output sample rates should be 44100 or 48000, should I worry about input (Mic) Sample rates as well? (Currently I was requesting the sample rate on both ends to be 16000 samplesPerSecond, for ease of passing into the codec) Also, do you recommend any particular resampler that I should use, or are any of the ones out there probably okay, or should

Have you tried to send mp3 files with other formats than 44khz stereo? I'm not talking about the encoding quality for send the data to icecast, else about the quality of mp3 files. If you do that, the listeners will hear the music faster and higher, like the little chipmunks in donald duck's, and it will be broken by no-sound intervals. It happens if the files are mono too. This is