search for: 192kbps

Hello, I’ve a question. I have a station now with Shoutcast and shoutcast transcoder for 2 bitrates. 192kbps mp3 and 64 kbps aac+. I want to transfer these to Icecast. I’ve an working icecast server but i need an auto dj that can encode 2 bitrates, 192kbps mp3 and 64kbps aac. I had a working setup, but my nonstop machine is not on the icecast server but externally. Now in shoutcast transcoder i can give...

...ival last weekend and used butt (Broadcast Using This Tool) to send a live music stream to Icecast at the radio station, which was then played by iTunes in the Broadcast Studio, and then to the broadcast FM signal. It all worked perfectly. We were streaming from the festival using mp3 format at 192kbps/stereo and I prepared a fallback-mount file ahead of time in the same format. It was never played because the operation went perfectly, the remote stream was stable for the entire 12 hours of the broadcast. But I was not confident that the remote Internet connection could guarantee 192kbps ban...

Why does opus scale down frequency from 22k to 20k ? 64kbps<https://lh5.googleusercontent.com/-DGsEd1ijTBs/UDE3OFVrE5I/AAAAAAAADBo/bFcWQ6I4-Zk/s588/opus_64_test.png> 128kbps<https://lh3.googleusercontent.com/-3Wfvq6OPGRw/UDE3OClxEdI/AAAAAAAADBk/vsBJ9Gcd_pU/s589/opus_test.png> 192kbps<https://lh4.googleusercontent.com/-tfDAtdGM2Aw/UDE3N3mrFJI/AAAAAAAADBg/38weacOnQ7I/s588/opus_192_test.png> this action will affect the quality of audio. *Can this feature be disable in future ?* And *Can opus add no noise and echo in audio at high kbps *(192kbps +) *?* -------------- next...

...try_TheFall.mp3 Encoding as 44.1 kHz 160 kbps stereo MPEG1 LayerIII ( 8.8x) qval=2 Frame | CPU/estimated | time/estimated | play/CPU | ETA 11275/ 11275(100%)| 0:05:21/ 0:05:21| 0:05:21/ 0:05:21| 0.9177| 0:00:00 real 5m21.174s user 5m19.120s sys 0m1.860s Repeated @ 192kbps: real 4m39.733s user 4m37.680s sys 0m1.800s Repeated @ 192kbps with Variable Bit Rate (default quality setting, 4): real 9m49.699s user 9m37.580s sys 0m2.220s * * * Results: Format Size Encoding Time Original WAV: 51,955,7...

On Sun, Apr 26, 2020 at 11:32:10PM +0200, Marvin | OveralWiFi.nu wrote: > I’ve a question. I have a station now with Shoutcast and shoutcast > transcoder for 2 bitrates. 192kbps mp3 and 64 kbps aac+. > > I want to transfer these to Icecast. I’ve an working icecast server > but i need an auto dj that can encode 2 bitrates, 192kbps mp3 and > 64kbps aac. > > I had a working setup, but my nonstop machine is not on the icecast > server but externally. Now...

...nsure the current codebooks will work. We will really need different, newly-trained codebooks with this change; then the .01 can probably go away, though it won't hurt (I think). Be warned that the bitstreams increase roughly 50% in size with this change, so you should compare 128kbps new with 192kbps old etc. Please try this on tracks you find difficult to encode, especially very tonal stuff, I'll explain later what this really does, it's a bunch of math so I'll make a TeX file of it, Cheers && happy coding, Segher --- >8 ---- List archives: http://www.xiph.org/arch...

Are direct bit-rate converters possible for Ogg Vorbis? Or do they already exist? Even more, is it possible to directly convert MP3/ATRAC3 (i.e., Sony Minidisc) encoding to and from Ogg Vorbis? There has been an earlier discussion on this question. But this discussion centered around the fact that people couldn't see the point, and were rather hostile to the idea. There is a very good

Hi there, I'm using libvorbis in my program and need to encode to target bitrate. I know libvorbis prefer to use quality scale but I can't use it. I've found something at faq http://vorbis.com/faq/#quality *For now, quality 0 is roughly equivalent to 64kbps average, 5 is roughly 160kbps, and 10 gives about 400kbps. Most people seeking very-near-CD-quality audio encode at a quality

...io in the resulting file, taking a couple minutes, which is how long it would take to encode the audio in question). We've tried various things to ensure that the 2 OGG files are similar enough to be able to be concatenated without re-encoding, such as: Encoding a full song to OGG (in stereo, 192Kbps bit rate, 44.1KHz sample rate) using oggenc2 (a command line OGG encoder), creating a WAV file containing nothing but silence, reading the OGG's reported nominal bitrate (using the Vorbis API's vorbis_info structure) and encoding the WAV file to OGG (using the first OGG file's sample ra...

...ty than > .mp3 compression at 128kbps.* > > so 0 == 64kbps > 3 == 110kbps > 5 == 160kbps > 10 == 400kbps > > Could anyone give me a complete mapping from quality to bitrate ？ It doesn't really work that way. Those are about right nominally (and I know 6 == 192kbps), but in practice they tend to produce slightly larger bitrates. The bitrates vary quite a bit, since quality implies VBR or ABR. For example, with q 6 you might get an average bitrate anywhere from 180 to 220 kbps with significantly more variation within the encoded file. (And, of course, quality...

...seems to require a kodi plugin, is there something standalone I can use? I'm also looking for a currently playing script, searching has only revealed again a kodi plugin, I am looking for something standalone. I'm wanting to do multiple streams at various bitrates, the original files are 192KBPS .mp3 files, i'd like for those .opus files to be the high bandwidth files, then encode the .mp3 files to 128KBPS (for medium bandwidth), 64KBPS (low bandwidth), and 56KBPS (dialup). How would I get ices2 set up like this and how would I get the files in to the various bitrates? Thanks. Dave.

..._20001212.zip (yet the documents and source comment it is Japanese) I am worrying to that a particular file is not encoded normally with all the version, after, it that included beta3 release. http://hp.vector.co.jp/authors/VA012897/download/testwav.zip When this file is encoded with the rate of 192kbps over the problem occurs. The problem occurs even oggdrop.exe and, oggenc.exe in addition to my program. Therefore old libVorbis immediately before beta3 release is being used in my program. It should not be encoded correctly in all the version subsequent to it. vorbis_nightly_cvs.tgz (2000/11/17 0...

Hi! What are the assumed listening conditions in vorbis ? For example there are big differences in perceived sound when listening with headphones compared to listening with speakers. ( "super stereo effect" , different frequency response due to HRTF etc ... ). Is this stuff worth considering in a vorbis class encoder ? David Balazic --- >8 ---- List archives:

at 1 Jul 2005, I request help about HTB setting because my HTB setting did not work. After days, I found the source code bug lead to problem. My linux kernel is v2.4.17 port for mips by boardcom. boardcom set HZ to 200 in linux/include/asm-mips/param.h. In linux/include/net/pkt_sched.h, SCHED_JSCALE is set to 0 according to HZ=200. When I set SCHED_JSCALE to 13, the problem is solved. Shun

...download from one machine, the speed is limited at 256Kbit/s (this is OK) When I start a download from both machines, the speed for each machine is limited to 256Kbit/s. They are getting 512Kbit/s together. I want limit this speed at 384Kbit/s, so traffic from each machine will never be more than 192Kbps when both are in use. Why it is not working for me ??? What is the mistake ? Thank You. Ricardo

I don''t know if this has been suggested before but.. What about adding support for a bookmark or hyperlinked index? The idea is that a large compilation such as a long speech or an album would be maintained as a single file, but on playback, the player would show numerous ''tracks'' which are really just bookmarks relating to starting position of the specific part or

...udio in the resulting file, taking a couple minutes, which is how long it would take to encode the audio in question). We've tried various things to ensure that the 2 OGG files are similar enough to be able to be concatenated without re-encoding, such as: Encoding a full song to OGG (in stereo, 192Kbps bit rate, 44.1KHz sample rate) using oggenc2 (a command line OGG encoder), creating a WAV file containing nothing but silence, reading the OGG's reported nominal bitrate (using the Vorbis API's vorbis_info structure) and encoding the WAV file to OGG (using the first OGG file's sample ra...

christophe.guerin@etud.univ-pau.fr wrote: > > * Enough bandwidth to run the server. If you want to broadcast to 100 > listeners at 24kbps, you'll need about 24kbps*100 = 2,400kbps = 2.4Mbps > of bandwidth. That's about 2 T1 lines worth of bandwidth. Trying to > push 100 128kbps listeners down your 768kbps cable modem isn't going > to work :) It

Hi: A friend asked me a question that I was unable to answer, so I'm posting it here. Can files created by the beta4 encoder be peeled down when that becomes available, or is the ability to encode peelable files also not implemented? My friend wants to know if they should switch over now or hold off until this becomes possible. Geoff. --- >8 ---- List archives:

i'm recently started to encode my cd's into ogg vorbis format (192kb/s), and i am very much pleased with the quality. however, i will certainly feel awful after encoding 1gb of files and beta5 comes out, claiming to increase the sound quality. in other words, should i be using ogg vorbis for long-term audio storage without fear that those files are crappy quality and should be