similar to: New beta bug reports/bug report

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

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

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).

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

yes I know I can do that, but my question is if there is software to edit FLAC files without having to uncompress to WAV/recompress to FLAC. 2007/7/23, Avuton Olrich <avuton@gmail.com>: > > On 7/23/07, Harry Sack <tranzedude@gmail.com> wrote: > > hi > > > > does somebody know software for editing FLAC files? So I just want to > cut > > some pieces

I've had this problem encoding oggs where the output file is small, like 24kbytes for a 4 minute song (tested at 128 and 160kbit). I'm running Windows 2000 and this has happened in oggenc, oggdrop, and CDEX, though I've also been able to get good encodings with each of these. I think the only clean encodings have been .wav's that I've made myself with SoundForge or CoolEdit,

Hello, I've tried encoding a FLAC file from a WAVE file saved with 24-bits-per-sample, using the flac executable. I get this warning: WARNING: legacy WAVE file has format type 1 but bits-per-sample=24 The code is currently here on line 373: src/flac/encode.c where the comment says let these slide with a warning since they're unambiguous I do not really understand the warning

I've put together a first cut for a Win32 DLL using the BladeEnc interface. Right now, it's just a drop-in replacement for BladeEnc.dll that ignores all encoding parameters passed to it and uses the info_A vorbis_info struct (same as the encoder_example). It's not particularly useful as of yet, but you can rename it to BladeEnc.dll and use it with any number of rippers out there

Hi all, I've built Speex 1.03 on Mac OS X. I encoded a raw PCM file successfully using: /usr/local/bin/speexenc raw_samples.raw --rate 44100 --be --16bit raw_enc.spx Warning: Speex is only optimized for 8, 16 and 32 kHz. It will still work at 44100 Hz but your mileage may vary Encoding 44100 Hz audio using ultra-wideband (sub-band CELP) mode (mono) If I decode to .wav, it works fine.

On 2007-09-13, Brian wrote: > On Sep 13, 2007, at 12:08, Daniel Aleksandersen wrote: >> I would really like to see support for .cda as input files. The cda >> format >> is the one used on regular audio CDs. >> >> Support for this format would make it even easier to encode to FLAC >> from CDs > > You are mistaken, Daniel. Regular audio CDs do not use

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

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:

<speex-dev at xiph.org> Hi all, I'm a begginer with DSP and i need your help and suggestions. I'm trying to use the Speex and a DSP DM642 to implement a solution with voice! I'm using the 1.2 beta 3 distribution and the TI's Code Composer Studio v3.1 simulator. I coosed speex_C64_test.pjt and modified the speex_C64_test.cmd to only use the DM642 external RAM memory. I'm

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 )

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 aplication that I am developing is like a sound recorder (goes directly to the soundboard and start capturing the sound and writing the file, using the directsound api). I'm already capable of writing to normal wav files but I wanted to encode the audio with speex to have smaller size (and still good quality). How can I write the wav files with speex content using the ACM codec? (like

Hi , Is there any speex plugin for windows media player? Thanks, Sailaja Yahoo! India Matrimony: Find your partner online.Post your profile. --- >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 'speex-dev-request@xiph.org' containing only the word 'unsubscribe' in the body.

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

I've learned that I can make a tag with vorbisgain that tells the player at what volume a track is to be played to achieve a certain level of loudness. Is it possible that there will be a tag that tells the player for instance to add more bass when it plays a track? Ore more treble? With such tags it would be possible to customize the sound of a track according to my wishes - without