search for: 44khz

On 24 Jun 2015, at 15:33, Martin Jangowski wrote: > I regularly broadcast with 48kBit streaming rate (mp3, mono, 44kHz). > My fallback-file is a 244kBit, stereo, 44kHz, mp3 file. Works like a > charme.... That depends on the player, if it works you are lucky. It might not work for other formats or with some players.

Hi! I'm using the newest (beta2) speex library on PocketPC (Windows powered) and experiencing some problems with noise suppression preprocessor turned on while encoding 44kHz files. You can hear the example here: http://szalik.net/speex (this is a 44kHz, 16, mono file encoded in UWB mode) I tried using speex cross-compiled with gcc (fixed point + arm4 asm) and VS (just fixed point) and it sounds the same. There doesn't seem to be any problem with 8 and 22kHz input...

...st, we have compressed our wave test files into the FLAC format and tested the files that successfully compressed. Unfortunately, the FLAC encoder did not support our 192kHz wave samples. The test results are as follows: Sample Rate Bit Depth Channels Result 22kHz 16bit Stereo Pass 44kHz 8bit Mono Fail - Noise 44kHz 8bit Stereo Fail - Noise 44kHz 16bit Mono Pass 44kHz 16bit Stereo Pass 44kHz 24bit Mono Fail - Noise 44kHz 24bit Stereo Fail - Crash 48kHz 16bit Stereo Pass 96kHz 16bit Stereo Pass 96kHz 24bit Stereo Fail Unlike our wave tests...

Icecast allows using a fallback mount file which the listener will hear if the main mount stops. It is specified in icecast.xml as this example: <fallback-mount>/example2.ogg</fallback-mount> It is necessary that the file have the same bitrate and same format (single or two-channels) as the main mount. We were at a music festival last weekend and used butt (Broadcast Using This

...fference between 32 kHz and 44.1 kHz so it's probably worth > it. If you don't care about standard sampling rate, you could even to a > 2/3 conversion which would get you 29.4 kHz... I'm curious why not just sample at a lower rate if it's just VoIP anyway? My opinion is that 44kHz sampling for voice is excessive... unless you really do want to do CD quality. Greg --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing only the...

constant prediction see this page http://flac.sourceforge.net/format.html oversampling.. i maean digitally change the wave file rate form 44khz to 440 khz it make next sample easyer predictable if i'll found my old work i'll tell you how much compression i could achieve

...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 sound from the dsp device at 44kHz and then downsample it to 22kHz for one of my two streams. Ideas? DarkIce sure does rule. -samuel --- >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' cont...

Hi, I need some help... I've installed an icecast server but I can't use shout because it's always talking about "bad password". My icecast server use crypted password. I think the problem came from mkpasswd. I always get a different password when I use mkpasswd toto. Help me please. <p><p><p>--- >8 ---- List archives: http://www.xiph.org/archives/

...ing music, but I've noticed that the new VBR code chokes completely when you try to compress horns. I've placed a particularly offensive example up at http://www.utdallas.edu/~matthias/ . Take a look at a-16m*{ogg,spx}. a.ogg is the first minute of an ogg created from the source media (in 44khz stereo). The rest have been mixed down to mono and resampled to 16khz. The fixed-bitrate recording at quality 5 is fairly listenable, but the VBR encoding drops the bitrate down to where the artifacts make it almost entirely unlistenable. -- Matthias Granberry matthias@utdallas.edu (469) 371-05...

...experience and some of my friends' experience is that Linux is quite unreliable. Microsoft is really unreliable but Linux is worse." -Ken Thompson, Interview May 1999. http://www.freebsd.org FreeBSD - The Power to Serve http://www.rfnj.org Radio Free New Jersey - 435 streams - 96kbps @ 44khz Stereo http://namespace.org -- http://name.space Resist the ICANN! Support name.space! --- >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 onl...

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 stereo > outputs at the same time. If you do something for that, please also try to add the possibility to do a local high quality dump,...

...fied in icecast.xml as this example: > > <fallback-mount>/example2.ogg</fallback-mount> > > It is necessary that the file have the same bitrate and same format > (single or two-channels) as the main mount. > I regularly broadcast with 48kBit streaming rate (mp3, mono, 44kHz). My fallback-file is a 244kBit, stereo, 44kHz, mp3 file. Works like a charme.... Martin -------------- next part -------------- A non-text attachment was scrubbed... Name: martin.vcf Type: text/x-vcard Size: 319 bytes Desc: not available Url : http://lists.xiph.org/pipermail/icecast/attachments/...

...am. It seems as if the signal level of the left and right sides are being combined before the encoding process, leaving the sound muddy and overdriven. As such, if I reduce the output level by 6db, the combined streams are at the right level. The settings I'm using are Quality -1, 32kbps/44kHz. SpacialAudio suggested that they are passing the audio info to the library as required. Ideas? -I

Is there support for encoding/compressing 8KHz sampled speech? Am I heading in the right direction with Vorbis? I have heard results with 44KHz audio and am quite impressed. As always, any help appreciated. Regards, Paul -- Paul McHale Work: 937-320-5495 Double E Solutions Mobile: 937-371-2828 1435 Edenwood Dr Fax: 413-215-3232 Beavercreek, Ohio 45434 -- --- >8 ---- List archives: http://w...

Hello all, I am currently doing a partial port of vorbisenc to java. Partial port being that I am just concerned with 44kHz and above stereo coupled modes. I have done the grunt work of converting the 44kHz static code and res book data structures to java classes and want to verify that my data of the Java app matches that of C app. I'm pretty tired so instead of babbling an explanation basiclly what I am looking...

Hi, I've tried vorbis on a stream that has ~500Hz tops and it worked quite good, but the raw data was converted to a fake 16bit 44Khz WAV Are there any chances I can take the code and tune it to work with nonPCM data that has info from about 5 up to 500Hz? It would be nice to know where to start. Thanks you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorb...

...y>0</quality> <managed>1</managed> <nominal-bitrate>35000</nominal-bitrate> <minimum-bitrate>30000</minimum-bitrate> <maximum-bitrate>40000</maximum-bitrate> </encode> [...] Original files are about 128-200 kbps and 44KHz The problem is that the stream is 90% of the time 42Kbps ( when i listen to it over my LAN with xmms ) Is that normal ? Among that , i would like to know (if you can help me) the lowest bitrate for that . I stream Music in mono ( 1 channel ) mode , and want to sample at 22Khz . i juste want to...

...io. Some audio editors will stop after the 2GB limit is reached, or start recording a new file, but Sound Forge will keep on trucking when recording in .w64 format. I can see this format becoming more popular in the future. I can then edit the raw .w64 file in Sound Forge and convert it to 16bit-44Khz for tracking/burning to a CD. The problem comes when I try to archive the raw audio .w64 files to DVD - sometimes I need to compress them to make it fit. Right now I have to break a .w64 file up into 2 .wav files with Sound Forge, then compress each .wav file individually with flac. Or burn extr...

...st (or the original author) be interested in running with it? Monty > Hello - > I hacked up an ogg winamp output plugin based on the vorbis > example_encoder and the out_raw winamp sdk sample. It requires the same > deal as the winamp input plugin to compile. It will only encode 44kHz > streams correctly though. > It is not really very noteworthy. But all of my windows friends that I > have turned on to ogg can't get the command line encoder to work :) , so > I had to pass this out. Since it is "in distribution," I thought it > would be good to let...

What have people found to be the ideal setup for recording asterisk prompts? I'm looking for both the ideal application to record them in, the ideal format, as well as hardware (do I need a fancy studio mic or will a headset mic work?). Thanks, Justin