similar to: optmizing encoding

(well, YMMV, but it got me 12% on Mac OS X Public Beta on my test case). In vorbis/lib/scales.h, add the following after fromdB() #define sqrt_fromdB(x) (exp((x)*.11512925*0.5)) In vorbis/lib/psy.c at the bottom of bark_noise(), do: < noise[i]=sqrt(fromdB(v)); > noise[i]=sqrt_fromdB(v); This avoids the sqrt() call entirely by just doing it as part of the exp(). (I'd

Before re-inventing the wheel I would like to ask: does anyone know about an optimizer in R which can reliably identify which value of X (Xopt) leads to Y (Yopt) closest to Ytarget in Y <- MonotoneStepFun(X) optionally with the restriction that Yopt <= Ytarget (at least if any Y <= Ytarget, otherwise any Yopt > Ytarget would be the preferred answer) If none is known, I will write

Is anyone working on making oggenc use some nice library that reads lots of different audio file formats so that people can encode from files other than WAV? For example, the 'AudioFile' library (just found it via a search on freshmeat) reads wav, aiff, aiff-c, .au, and .snd. Maybe there are other better libraries (OpenAL, maybe), but this library does what I want (read AIFF). -tim ---

Lancer 20060302 (based on aoTuV 4.51) http://homepage3.nifty.com/blacksword/ Changes: * update to oggenc v2.81 * optimized SSE2 code implementation * bark_noise_hybridmp and seed_curve are inline assembled * modified mdct_forward's SSE optimization * double step bresebham algorithm for render_line, render_line0 implementation

Dear all I'm currently using a simple encoding implementation based on the encoding example delivered with theora. Now i tried to change the quality of the output file. Its my understanding, that either a bitrate can be set for Constant Bitrate (CBR) encoding, or a quality (0-63) for Variable Bitrate (VBR). As you might know, the basic Encoder Example uses the so-called 2-Pass method for

Apologies; a filter mistriggered on the original send of this message. Monty Date: Fri, 10 Jan 2003 01:11:16 -0500 (EST) Message-Id: <200301100611.BAA12395@biohazard-cafe.mit.edu> To: vorbis@xiph.org From: ben-extra@MIT.EDU Subject: Hi, I am hoping for some guidance. I have a *bunch* of audio in the form of realaudio streams. As is, they are of a quality similar to fair AM quality.

Hi! There are some little inconvenience in libao-0.8.5. - The biggest is may that: the documentation and the header file declare the ao_file_extension function, which give a hint for the file extension where the device is realy a sound file. This function is missing. -An other: the alsa 0.5 and the alsa 0.9+ drivers short name. It will be better if the alsa 0.5's name will be alsa05 and the

I have encoded successfully a YUV stream by using the encoder_example. Can We encode a row stream (get from a web cam) by using encoder_example. If we can not, is there any other way to encode a row stream by using theora codec. -- Nuwan Millawitiya -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have encoded successfully a YUV stream by using the encoder_example. Can We encode a row stream (get from a web cam) by using encoder_example. If we can not, is there any other way to encode a row stream by using theora codec. -- Nuwan Millawitiya -------------- next part -------------- An HTML attachment was scrubbed... URL:

I tried to encode two minutes from an AVI DV file 720x480: ffmpeg latest release produces a FLV 320x240 25fps 360k video 40k audio in about 45 sec. ffmpeg latest release produces an OGG 320x240 25fps 360k video 40k audio in about 1:20 sec. ffmpeg2theora the same in about 2 minutes or more. The FLV quality it's quite good, some dropped frames maybe. I was wondering to know if ffmpeg team

thank god. thank you monty. thank you so much. good grief. again, thanks. thanks for ending this. wow. let me just say this. my experiences here on the list has been nothing but hell. wow. there might be a light shining at the end of the tunle? wow. thanks for putting an end to this. jeez. again, thanks. Timothy Your friend in the music industry follow me on twitter

Fair warning after helping Timothy Clark out, is ill advised and warn you stay away from this guy. While he talks a good talk, and "Christian", come forth his ways are divisive and he is all over the place. I helped him and maybe its my fault for being so helpful, and then getting involved in it. I can assure you this guys motives are not right. I let my faith in common man of course

Pretty basic. -fno-common is required to produce dynamic shared libraries on Darwin based systems. -tim Index: configure.in =================================================================== RCS file: /usr/local/cvsroot/ogg/configure.in,v retrieving revision 1.10 diff -u -r1.10 configure.in --- configure.in 2001/01/22 22:49:45 1.10 +++ configure.in 2001/01/28 03:53:05 @@

Everybody of vorvis-dev hello. I am the program developer of Japan and develop the derivation library of libVorbis and be distributing in conformity with LGPL. http://hp.vector.co.jp/authors/VA012897/download/vorbisacm_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

Hi, I'm planning to re-encode all my MP3s with Ogg Vorbis but I wanted to wait until 1.0 was released because of channel-coupling. I read that would increase quality about 40% at the same bitrate and I didn't want to encode my files wasting all that bitrate ... Now, since 1.0rc1 is released I am wondering if I am actually able to use channel-coupling right now. I installed all the newest

Chris Pearce wrote: > Hi Guys & Gals, > > I need you guys to decide whether we want to include extra granulepos > fields to Skeleton 4. Given the underwhelming discussion regarding this, > I'm guessing the need and/or desire for these fields isn't really there. I haven't commented mostly because I don't know what Monty's plans for a "grand unified

Could someone please try this version of bark_noise_median() on a x86 box? On my PPC box, this runs about 1.85x as fast as the current version, making the encoding speed about 6.5% faster (at least on my test case). The basic idea is to (a) avoid needless float/int conversions and (b) use multiplies instead of divides. I'm assuming (after having done some tests), that

Hi, everyone! I'm a newbie of speex develog. I tried to encode and decode with the sample program in the speex manual. I compile those program with gcc sampleenc.c -o enc -lspeex gcc sampledec.c -o dec -lspeex I run these two program in this way and then speak loudly: ./enc /dev/dsp ? ./dec /dev/dsp I thought I would hear my voice. However, what i heard was a lot of noise. I

I was optimizing a 650MB using ferret (0.11.3) and I received the following error. I''ve seen some people have similar issues but I haven''t seen any resolutions. The contents of the index directory follow the error. Has anyone seen anything like this and found a resolution? Many thanks. /mnt/apps/search/releases/20070622175637/script/../config/../vendor/

I have some speech encoded at 140 kbps that I would like to have encoded at 40-60 kbps instead. I could just convert the file to wav and re-encode the file, but I seem to recall that ogg-files very easily can be converted to lower bitrates (something with cutting off the last bits). I would assume that this also will preserve more of the original data. Is there a utility for this? /Ole --