similar to: oggenc --help text buglet

Just got around to compiling RC3 under beos and came across an anomaly when using managed bitrates. I haven't changed the code for the beos encoder but I now get double the bitrates so vorbis_encode_init(&vi,mediaFormat.u.raw_audio.channel_count,(long)mediaFormat.u.raw_audio.frame_rate , -1, 128000, -1); now gives me vorbis files that average around 325 - 350. Is this now the correct

Here are some minor things I noticed when updating the vorbis module for GoldWave: Bitwise.c, line 175 and 207 Warning: Negative unsigned value ret=-1UL; Info.c, line 385 Warning: Unreachable code break; vorbisfile.c, line 1407 Warning: Call to function with no prototype int host_endian = host_is_big_endian(); fix: add 'void' to line 1339: static int host_is_big_endian( void )

Hi, It's a bit offtopic for this list, but you might be able to help. For libvorbis rc3, what is the correct bitrate parametrization for the call vorbis_encode_init()? For rc2, it worked calling it the following way: vorbis_encode_init( &vorbisInfo, 2, // two channels 44100, // 44.1kHz

Hi, I'm experimenting with managed mode encoding with specifying maximum bitrate. I call: vorbis_encode_init( &vorbisInfo, 2, 44100, -1, 96000, 96000); to initialize the encoding. To my surprise, it seems the maxbitrate value of 96000 doesn't have an effect, the bitrate of the

Hi, I'm experimenting with IceCast2, using DarkIce to generate the stream. I have found some peculiarities with the vorbis bitrates. In DarkIce, I call vorbis_encode_init() with about the following values: vorbis_encode_init( &vorbisInfo, 2, 44100, 96, 96, 96); which by all reasons should generate a 96 kb/s stream, as all max_bitrate, nominal_bitrate and min_bitrate are set to 96.

Hi devs, I am developing winLAME, a frontend for LAME, and it also supports Ogg Vorbis decoding and encoding. I've got a question about the new API function vorbis_encode_init_vbr(). Do I have to call vorbis_encode_init() before to set the bitrate(s)? And what is a good default value for the "base_quality" parameter? I read on the mailing list that ABR use in Ogg Vorbis is not so

Hi, I want to encode audio at very low bitrates. however, vorbis_encode_init() does not allow me to set bitrates below 32kbps. How can i set 24kbps? Thanks, Flo --- >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 'vorbis-dev-request@xiph.org' containing only the word

I've noticed that oggenc reliably crashes my K6-200 box after a few minutes of encoding. Has anyone else experienced crashes like this? Surely it must be a hardward fault, or perhaps a K6 problem (this processor has some errata). Still, I'm surprised that nothing else I do with it seems to hang it. Has anyone else seen anything like this? --Mike --- >8 ---- List archives:

Hi, I discovered a little error in vorbis.m4. The source for the test compile uses vorbis_encode_init() but does not include vorbisenc.h. This causes the test to fail when AC_LANG(C++) is used in my configure.ac script. For now my work-around is to surround the Ogg Vorbis tests with AC_LANG_PUSH(C) ... AC_LANG_POP(C) calls, but it should probably be fixed. Thanks, Davy --- >8

Hi, I'm trying to load the vorbis libs dynamicly on Mac OS X. I've got success for libogg, libvorbis and libvorbisfile. But libvorbisenc drives me crazy. The function "vorbis_encode_init" seems not to be in there if I am using "NSLookupSymbolInImage". But the tools "otool" and "nm" tell me that the function is declared. Does anyone know, what I

Conrad, Small buglet with the 0.9.1 liboggz release (go dude!) http://thaumas.net/~giles/xiph/elphel/clips/elphel_00017.ogg causes a float exception in oggzinfo when it tries to calculate the bitrate. It fails to measure the duration and tries to divide by zero. :) There may well be something wrong with the file, although oggz-validate doesn't complain. Also, the configure script

Hi, Maybe this is documented somewhere, if so, please send me a link to the documentation. My question is: how to set up different (VBR, CBR, ABR) modes when calling the Ogg Vorbis API? Currently I do: CBR: ret = vorbis_encode_setup_managed( &vorbisInfo, getInChannel(), getOutSampleRate(),

Hi, Why do I get an error message when i'm running the "encoder_example.c" which is included in the SDK? The error occurs then the 44 bits are read and the vorbis_info struct is initiated (i.e. when the vorbis_encode_init(&vi,2,44100, -1, 128000, -1) - funcion is called. I also tried to read the data from a 16 bits 44,1 KHz stereo PCM - wave file but recived the same error.

Hello folks I am encoding music to ogg vorbis format for the first time. From what i have researched, ogg vorbis is natively a vbr format. Also encoding using the -q switch is encouraged instead of specifying a bitrate through -b. Now, while encoding using -b i get this: Encoding "file.wav" to "file.ogg" at approximate bitrate 224 kbps (VBR encoding enabled) Notice

At 09:49 AM 2/20/02 +0100, you wrote: >Hi, > >It's a bit offtopic for this list, but you might be able to help. For >libvorbis rc3, what is the correct bitrate parametrization for the call >vorbis_encode_init()? > >For rc2, it worked calling it the following way: > >vorbis_encode_init( &vorbisInfo, > 2, // two channels

First let me say that I was absolutely astounded at the sound quality when using oggenc at q = -1 (around 50 kbps). So much so, that I'm anxious to try some of the lower bit rates that were mentioned in the announcement for 1.0: ... audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. But I am having trouble finding how to use either oggdrop or oggenc to get these

Dear Developer Team (Monty/Michael Smith/and others), It seems you get angry for me, but I don't understand why. I don't want to dispute with you, but maybe I forgot to say some things... Here they are: 1. My program (clone) is not a finalized, and NOT A RELEASED (public) program. This is just a test, I wrote it to myself, to the (sound quality) comparsion of MP3 and OGG. 2.

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is getting its

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is getting its

...even worse, it *appears* to work, and produces crap files. When specifying a -b <foo> option for a batch encode, -b <foo> is only obeyed for the first file. All subsequent files encode with the equivalent of -b 0 (despite the chatty text output claiming otherwise) resulting in overly low bitrates that sound like shit. I don't know if this affects -q as well, but it a safe