similar to: Adding RC3 support to GoldWave

Hi all, I am relatively new to Linux and wine and I am trying to run an audio editing software called Goldwave 5.06 (version 4.26 works relatively well). 5.06 installs ok but when I go to run it I get the following error: "Driver does not support 16 bit colour depth or visual buffer could not be created. Visual disabled." And the meter displays do not work. I have tried adjusting

GoldWave v 5.08 does not work. It starts, but when try to load any .wav file after pressing "Play" button it plays file endlessly. After pressing "Stop" button, playing loops - it plays endless some final part until program will closed. I had done log with +relay,+seh,+text parameters. It is relative big, so I do not attach it. Instead, I pack it with BZip2 and place to

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

I just tried this with Goldwave.. it decoded the file, i edited out some stuff, and saved the file (flac format) and the tags appear to be intact. Greg M. --- Tomas Valusek <tvalusek@seznam.cz> wrote: > Hello, > > and what if I want to preserve FLAC tags while I'm > editing its contents? > Is there a way e.g. to cut silent block from FLAC > and preserve tags? By

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

I'm a loyal user of GoldWave (http://www.goldwave.com/), and recently the Ogg Vorbis support module has been updated to RC3. Is it still safer for me to save files as WAV and then use the "official" tool (oggenc) to encode my files, or are all RC3 implementations the same, and it makes no difference with what program I encode the files, as long as the module uses the latest

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'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

Hello, Do you know of any Windows software that is able to record in mono from the sound card's line input and at the same time convert to *.ogg? Ultimately, one without ogg conversion would be OK too provided it can record mono wave files. Important: the software should be able to record WITHOUT first creating an empty wave file with a set length (Adaptec's Spin Doctor versus Goldwave).

Hi, While testing ReplayGain (so it could be related to a bug), I noticed the following gain comments for a file: RG_PEAK=1.71580 RG_RADIO=-7.91 dB RG_AUDIOPHILE=-6.72 dB I've never seen such a large peak. Not that I've looked much, nor have I analyzed the file further. I just thought I should mention it. :) The file is the track Board Burner by Mixmaster Mike, available at:

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(),

I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing

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 All, It would be really helpful for me if you comment on the following. My VorbisEncoder uses libvorbis to encode into the vorbis data. I have followed the call sequence given in libvorbis\examples\encoder_example.c. I am taking a 16bit pcm Stereo wav file and generating an oggvorbis file using libvorbis and libogg. There is lots of noise in the file except music is heard very feable

Hello All, I have several soundfiles that are recorded ub 44100Hz, 16-bit Mono. What is the best way and right tools to use to downsample these to 8000Hz so that they can be used with Asterisk. I've tried using sox with the -r switch and Audacity on the mac and Goldwave on Windows and they all generate files that sound like a bad acid trip. I tried increasing the speed 551.25 percent after

> While on the subject of plug-ins, has anyone written a Speex plug-in > for the Goldwave audio editor? There is an ACM codec for I/O of WAVE files encoded in Speex. ( www.openacm.org ) But there is no plugin to load or save Vorbis Speex files natively, as far as I know. Christian <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage:

Hello - I am new to the list and new to FLAC and I have an encoding question. I am using Goldwave to create a file which is 32 bits, signed, little endian. I am doing this to avoid the added step of converting from 16-bit samples to 32-bit samples. All works well except that encoding a sine wave does not output an exact replica of the sine wave -- far from it. When I encode a 16-bit sine

Two wavs I tried to encode tonight got errors near their ends, saying something like "skipped unknown sub-chunk 'LIST'." These occurred at 97% into one file and 98% into the other. If I hadn't brought the DOS command window to the foreground and hadn't been at the monitor watching before these warnings were scrolled away by the displays from work on other files in the

The attached file contains build files for TI's Code Composer Studio (CCS) for the C54x, C55x, and C6x DSPs. I had intended to post this a couple of months ago, but it took a long time to get around to doing the little bit of cleanup required. This is a file I/O loopback application suitable for running with the CCS simulators, for evaluating memory and MIPs requirements for these

Hi, Regarding to my previous question, I think I have found out from oggenc source code that the correct way to initialize encoding is: vorbis_encode_init_vbr (or vorbis_encode_setup_managed for cbr etc.) vorbis_encode_ctl (to turn off or use avg bitrate management) vorbis_encode_setup_init By this way, there is no problem with encoded stream. However, there is no documentation of the three