similar to: Goldwave 5

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 )

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

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).

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.

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

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

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. Joined the list since this seems to be the only point of communication. Just have two comments/questions. 1) I looked all around the developer's site and vorbis.com and don't see any mention of where to send bug reports for 1.0 beta. I'd think it would be a FAQ. 2) My bug for the new beta. (I'm hoping someone of the PTB is reading ... I definitely don't belong on the

Hi. Joined the list since this seems to be the only point of communication. Just have two comments/questions. 1) I looked all around the developer's site and vorbis.com and don't see any mention of where to send bug reports for 1.0 beta. I'd think it would be a FAQ. 2) My bug for the new beta. (I'm hoping someone of the PTB is reading ... I definitely don't belong on the

I've only begun investigating R as a substitute for SPSS. I have a need to identify for each CASE the closest (or most similar) 5 other CASES (not including itself as it is automatically the closest). I have a fairly large matrix (50000 cases by 50 vars). In SPSS, I can use Correlate > Distances to generate a matrix of similarity, but only on a small sample. The entire matrix can not

________________________________ Hi David, Thank you very much for your reply. Since I need to resample the audio in the program itself, I decided to try out the resampling API in speex. But now, I have another problem. The resampled sound is very much distorted and clicks appear quite often. (I have attached the source code I used for testing it below). The test data I had was a file sampled

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.

hi, since HL 1.1.0.8 supportes VoiceMessages. i get allways the error for: waveIN ... an at the same time in the log this message appears .... : fixme:dsound:IDirectSoundBufferImpl_QueryInterface {279afa86-4981-11ce-a521-0020af0be560}: I know about this GUID, but don't support it yet does anyone has a fix for that ? it's like 1914 i think inthe dsound_main.c but i'm too less

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