similar to: managed mode / max bitrate doesn't have effect

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 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, 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(),

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

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 )

> 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. Strangely > enough, the generated stream's bitrate (according to XMMS and WinAmp) > varies between 49 and 59. Currently min and max are ignored by the library, since bounded bitrates

Michael, > See the examples. If you initialise a managed mode (which this is), you > MUST use vorbis_bitrate_addblock() and vorbis_bitrate_flushpacket(). Thanks for the tip. I added the calls, and it works now. > You SHOULD also give an option to set min/max, since they're now used, > and to use a VBR mode rather than the managed modes here. But when streaming, IMHO it is

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'm not sure that this is not my fault, but here it goes: I'm trying to encode a 22 kHz Mono stream. I started with the encoder example and made it take dynamic channels etc. This is how I init it: vorbis_info_init(&VorbisInfo); vorbis_encode_init(&VorbisInfo, 1, 22050, -1, 128000, -1); then I get a crash in: _vp_quantize_couple(vorbis_look_psy * 0x01e6a6ec,

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

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the a=T38MaxBitRate paramater was sent as "maxBitRate". Without capital "M". Are you closer to

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

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

Hello, We're trying to receive faxes on the Asterisk server, but for the time being T.38 negotiation fails. The SDP that the Asterisk reINVITE sends contains these lines: ---------------------- m=image 4968 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval a=T38FaxTranscodingMMR a=T38FaxTranscodingJBIG a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400

First off, I don't even know if this is the right place to ask these kind of questions, but I haven't been able to find answers anywhere else, so 1. I have found that "pcmTotal * vorbisInfo->channels * 2" gives the uncompressed size of every ogg vorbis file I have used. What is a more robust way of getting the full uncompressed file size? 2. How do I read a certain amount of

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.

Hi there, here are my config files for ices and icecast2: <icecast> <location>Pete's Flat</location> <admin>mr2147@bigfoot.com</admin> <limits> <clients>20</clients> <sources>4</sources> <threadpool>5</threadpool>

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: If you want to use icecast 2 for streaming vorbis audio then don't get it from the CVS repository at icecast.org. That's ancient developer stuff in there. Instead get it from the xiph.org CVS repository (see http://www.xiph.org/cvs.html which I see now lists the icecast stuff (yay!). You'll need the icecast module plus the avl, httpp, log, net, thread and timing modules (check

Hi, I'm not anywhere near an expert, but I had successfully used Darkice, TwoLame, and Icast231 to netcast a mp2 stream. If it will help, here is a snip from the related area of my darkice.cfg: [icecast2-1] format = mp2 bitrateMode = cbr bitrate = 384 quality = 1.0 server = 127.0.0.1 port = 32710 password = (duh!) sampleRate =