search for: goldwav

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 the color d...

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 paramet...

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_i...

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...

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 wh...

...es\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 in the background. To compare, I have created an ogg file with the same configuration(same quality) using GoldWav. I have observed that even the headerpages differ from the file i have generated. I am using TInt ret=vorbis_encode_init_vbr(&iInfo,2,44100,.4); for the setup Why are the headers of my file and the file generated by the GoldWav differ when both files are to be recorded with same quality...

...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). Thank you in advance for your help. Andras --- >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-request@xiph.org' containing only the word 'unsubscribe' in the body....

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

...mmands from the dialplan does not actually capture the tones as received from the wire. This capturing is turned on and off by an AGI command, so that my AGI program can turn it on before waiting for the DTMF string and off again afterwards. Examining this captured audio in an audio editor such as Goldwave does not provide any clue why the digits might have been missed. On most occasions the digits are clear, long enough and well spaced. Yet Asterisk still misses them. The system does not seem to have been heavily loaded at the time either. Can anyone offer any clues as to why this might be the ca...

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.

> 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: http://www...

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 w...

...or watching before these warnings were scrolled away by the displays from work on other files in the batch, I wouldn't have known: the front-end reported nothing about the errors. The wavs were ripped by EAC, and I couldn't imagine what might be troublesome about them. I resaved them with GoldWave and re-flacced them, but FLAC gave the same warnings. The resulting .flac files played properly in WinAmp, and the wavs I decoded from them matched exactly (according to EAC's Compare Wavs function) to the originals from which they had been encoded. Was there a problem or wasn't there?

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 doing the resample on these files and then it sounds like something from the wizard of oz... Any help that you can provide would be appreciated. Thanks, -Nate --------------...

...20af0be560}: 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 expirienced ... There is a tool, called voice_tweak.exe too. it i try stating that, i get : couldn't find mic volume. ( but goldwave ( www.goldwave.com ) works ... )... please somone help me out !!!!!

There was a post in the Opus subthread on HA where there was a link to the file called Opus-tools-0.2-34-g98f3ddc_(using_libopus_1.5.2-21-gff6dea5)_Win_GCC142.7z. That's the one I'm currently using. First, I used GoldWave to convert your input file from ?-law to PCM and saved it as a ".wav" file. Next, I encoded it with Opusenc, I decoded its result with Opusdec with the target sample rate being set to 48 kHz, and I got no excessive amplification. Not sure what's taking place on your side. Petr

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

...rs, for evaluating memory and MIPs requirements for these processors. The zip includes a readme file describing the changed and added files. I have done no assembly optimizations, just made working builds. I originally had a test audio file in here, but that made the zip much bigger. I used GoldWave to convert the sample files from the Speex website to headerless audio files. Note that this build includes a (selectable) manual memory allocation, which allows the user to allocate memory out of a fixed array, instead of using calloc. This improves visibility into the allocation process, ma...

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