similar to: Questions about bitrates

I did some testing of vorbis codec (Winamp plugin) on a Pentium 166 MMX, and found that the performance is pretty good. I got 3 % cpu utisization if I turned dithering off, and 5 % otherwise. This was with a 138 kbit song. It's pretty long to mp3 performance that was 0 to 1 % on that machine for a 128 kbit song. Also vorbis seems to have a larger working set, because if I play two songs at

On Aug 07 08:30:31, hans at stare.cz wrote: > On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > > with Opusenc and then decoded the resulting file with Opusdec. > 1) Opusenc --bitrate 12 --downmix-mono Sweep50.wav Sweep50.opus Why are you using a stereo file containing the same sweep in both

Dear Vorbis Devteam, My name is Michael Feilen and I've been working on Digital Radio Mondiale (DRM) transmitters and receivers for quite a while now. DRM uses HE-AACv2 by Dolby to encode the audio content (see http://www.drm.org/uploads/media/es_201980v030101p.pdf - pages 23 ff). As I think Vorbis is an excellent alternative, I'd like to implement and define an interface for Vorbis

Dear all, I am an active FLAC user and like the program a lot! Thanks for your great work. Since I am not a developer I cannot modify the program on my own. Maybe you might be interested in features I would find useful. Maybe others find them interesting as well and you find the time to implement them. 1. Add encoder options: a) Skip silence (leading and ending) would give the option to skip

Thanks to my provider, my old email alias (hongli@telekabel.nl) had been deleted. Now I can only use h.lai@chello.nl However, I'm subscribed to Vorbis's mailing lists using hongli@telekabel.nl Is there any way to change my subscription's address? <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

Thanks to my provider, my old email alias (hongli@telekabel.nl) had been deleted. Now I can only use h.lai@chello.nl However, I'm subscribed to Vorbis's mailing lists using hongli@telekabel.nl Is there any way to change my subscription's address? <p><p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe

Hello, A question: Can we use uncoupled encoding in the 1.0? (current CVS) How? Because I see some uncoupled definitions (books\uncoupled\res_books_uncoupled.h), but I don't know how could I switch to it manually. Perhaps with an option in the vorbis_encode_ctl() function... Because I think so it's better if I don't use any kind of channel coupling (lossless neither) on high bitrates

On Aug 07 00:41:52, petrparizek2000 at yahoo.com wrote: > ????#1. To test encoding at low bitrates, I encoded a sine sweep at 12 kbps > with Opusenc and then decoded the resulting file with Opusdec. What sine sweep exactly? How did you obtain it, and how exactly did you encode and decode it? Jan > The strange > thing was that even though the output wave file was at 48 kHz, it

I just read some of the discussion on the list about 'bitrate peeling' and remembered an interview of Monty that I have read recently. In it he says that Vorbis uses MCDTs <sp> and that these are theoretically reversable. And now, I learn that theoretically we can use bitrate peeling to make smaller files from larger ones, and that leads to my question. Could I theoretically

Hello, the voice-communication TeamSpeak (www.teamspeak.org) is currently testing a version that supports speex codecs. The quality for high bitrates is quite good. BUT, the low-bandwidth speex codecs that are currently used arent very good. What I did to find this out: I comprared a speex AVB with 6.3 KBit/sec (total, overhead for packets and stuff included) and the 6.3 Kbit/sec Celp Codec

I had expected that the default bitrate for the encoder would be the same as setting it to OPUS_AUTO, but I'm getting difference results: >opusenc --comp 4 sample.wav sample.opus Encoding using libopus 1.3-rc2 (audio) ----------------------------------------------------- Input: 8 kHz, 1 channel Output: 1 channel (1 uncoupled) 20ms packets, 25 kbit/s VBR Preskip: 312

Ogg Vorbis is still not known by the mainstream and not widely used these days. I've been thinking about this... MP3 is so popular because people can download music for free (and thus pirating music). I recommend Vorbis to everybody I know, but most of them refuse to even try it because it's not popular (or because they are hardcore MP3 zealots), which results in a circle (little people

Hello, I'm not sure if this list is the proper place for this post; please correct me if it is not. I am attempting to use libtheora and I have been looking through the examples in the distribution Excerpt from encoder_example.c -- case 'V': video_r=rint(atof(optarg)*1000); if(video_r<45000 || video_r>2000000) { fprintf(stderr,"Illegal

grep -i bitrate /usr/include/FLAC/* doesn't work...

Hello all, I'm glad to announce to you that Vorbis now has full ReplayGain support. If you're not familiar with ReplayGain, take a look at www.replaygain.org. The main features are: a) all songs play back with equal loudness b) removes the need for normalization c) allows for clipping prevention Using it is very simple. Get a compatible decoder (ogg123, XMMS and WinAmp all support it

Looks good to me. Jean-Marc Alfred E. Heggestad a ?crit : > Hi > > Please find an updated version of the Speex I-D attached. The only > change is addition of the copyright conditions in Appendix A, > as requested by Ivo. > > Many thanks for your input. > > I will give you a few more days before submitting to AVT working group > > > /alfred > > Ivo

After doing an informal (128k) listening test, I have concluded that I prefer Beta4 over RC2. The 16kHz low-pass on the RC2 encoder makes it sound like FM radio. Both encoders SEEM to have a couple of dB bump at 10kHz. JT --- >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

Hello FLAC-dev, Do you know if there is some good educational documentation showing FLAC's internal design in a few steps, like this doc for monkeysaudio : http://www.monkeysaudio.com/theory.html If not, what is the principle of the algorithm in a few big steps (example : 1. Conversion to mid/side instead of L/R channels 2. Predictor 3. Entropy coding) ? Best regards. --------------

>Dear Speex developers >I am going to port Speex on LPC2368 >I tested Speex encoding and the mesurments shows ~40ms cpu time for one >frame >Do you know who ported speex to NXP or other 32bit platform? >Best Regards >Zohar fox I have an own optimisation for Speex for different versions of ARM architecture. Here is Measurements of performance for STM32 72 MHz. 4000

This is more of an EAC question, but I have not yet received an answer from my post on that page. Since they are closely related I thought I would give it a try here. I just bought a BIG stack of CDs (70+) and decided to recheck all of my EAC (beta 4) + FLAC (1.1.2) settings prior to ripping and encoding them. I also calibrated a new external Plextor CD-ROM too. The setting string I use in