search for: soundforg

I'm on osx 10.6.4 using the newest version of wine bottler. The wine site says sound forge 4.5 works and everything does except when I try paste audio from a file into itself or anywhere. I get an error that the source sample rate is lower than the destination and it asks if I want to continue. Then it also asks if I want to mix the file down to mono if the audio is mono or mix up to stereo if

...ne of the files that doesn't have the extra 2 bytes you're gonna lose something you didn't want to On 11/1/07, Alex Brims <alex.brims@gmail.com> wrote: > Ok, we actually worked this out - there were 2 extra bytes doing nothing at > the end of the files. Opening the file in SoundForge and saving it (without > changing it) took off the extra bytes and allowed the file to convert to > FLAC. > > Thanks to everyone who emailed me suggestions. > > Is there a decent program for linux that could automatically take these > bytes off, without running the risk of rem...

Hi, How's the testing going? I compressed 194 individual .wav files (totaling 8.54GB) which contained tracks ripped from many varied albums. I unflacced them and compared their md5 signature with the same from the original .wav. They were all perfect. I didn't use the -V option just in case of any chance of mis-reporting. I hope to test it with the complete collection of ~41GB

One thing that has always bothered me about the ogg format is the distortion of high frequency sounds - even at data rates as high 128 and 160 kbps. I find the best way around this is to resample the wav file to 48 kHz (using SoundForge 6.0) before encoding (using CDex) to ogg. It takes a while, and adds a lot of extra wear and tear on my drive, but what a difference! The result is an 80k ogg file that sounds EVERY BIT as good as 128k MP3 at less than two thirds the size! http://www.subgenius.com/ts/hos.html I don't get the...

...ke 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, the bum ones have been tracks I've pulled off of CDs (with CDEX, EAC, and Windows native). Clues? -eric --- >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-reques...

...e issue ) : OGG files w/ variable bit rates generated by wget rip from the icecast2 stream . they stream well , but i can only listen from the beginning , even w/ the .OGG on my local drive _loaded in winamp - the length counter is way off , and i can't seek through the file . NOR can i edit in soundforge . . is this an OGG thing? a variable bitrate thing ? or the wget side-effect ? tx . . <p>> > and let me know if you neEd any scripts .. i even made one to generate the > > proper .m3u files on my web server so that every listen request gets a > > custom .m3u file auto-load...

...I keep getting no results when trying to use speex_echo_capture, speex_echo_playback and speex_echo_cancel in a multi-threaded application, as suggested in the manual. Though, the cancellation works properly when i use a file with human voice for far-end input and the same file with echo added in SoundForge for mic input. When i try to insert and remove silence between words modellng delay variation (about 10 ms) in the mic input file, the echo canceller doesn't make any significant change in the signal. The same happens with real microphone input, looks like it's due to the delay variation w...

...ed after the end of the data chunk without using another WAVE chunk. I've committed the fix (in src/flac/encode.c). it should now ignore anything after the end of the data chunk. Josh --- Mark Powell <M.S.Powell@salford.ac.uk> wrote: > Hi, > I have a large wav file generated by SoundForge. > The last 176bytes of > the file are some sort of comment inserted by > SoundForge. The length field > in the wav "data" chunk correctly specify the data > length as: > > <file-size> - (full wav header size) 44 - 176 > > However, flac seems to ignore...

...s. When FLAC complains about a "partial sample" it means there is a left channel sample without a right channel sample to go with it - a better term might be "partial frame" if you define a sample frame as a group of samples for every channel. The fixed file created by SoundForge either dropped the last sample from the left channel, or added a zero sample to complete the right channel. The long explanation that I gave yesterday, although accurate in itself, did not precisely apply to the bad file in question. Brian Willoughby Sound Consulting On Nov 2, 2007, at 12:...

Hello, I noticed that option --keep-foreign-metadata discards WAV cue markers. Here is how to reproduce the bug: 1) Create a 24-bit 96khz in SoundForge8, add 20 seconds of silence, and add two markers with "m" key shortcut 2) Save it, compress it with "flac --keep-foreign-metadata testmarkers.wav" 3) Decompress it with "flac -d testmarkers.flac" 4) Open the decompressed WAV in SoundForge: the markers have disappeared...

...ily do this? Simple copy? > At least by default > with a flag to specify if you don't want extraneous > data transferred > across? hmm... I'm not sure this is the right thing to do. flac is not really meant to store the metadata of other formats. are we still talking about the SoundForge comments? to answer your question, yes, it's possible, it just feels kludgey. Josh __________________________________________________ Do You Yahoo!? Yahoo! Auctions - Buy the things you want at great prices. http://auctions.yahoo.com/

I'm trying to install asterisk@home, I've just downloaded the latest cd from soundforge. I can get it to install ok (network card didn't auto configure - but I worked out how to use 'netconfig'). I worked out how to add a few grandstream budgetone fine. Worked out how to upload music etc. Worked out how to modify FOP. Voicemail and meetme's work fine. HOWE...

...C complains about a "partial sample" it means > there is a left channel sample without a right channel sample to go > with it - a better term might be "partial frame" if you define a > sample frame as a group of samples for every channel. The fixed file > created by SoundForge either dropped the last sample from the left > channel, or added a zero sample to complete the right channel. > > The long explanation that I gave yesterday, although accurate in > itself, did not precisely apply to the bad file in question. > > Brian Willoughby > Sound Consul...

Hi, I'm having problems encoding certain WAV files to FLACs. I'm working for a download store so we get loads of different WAVs from hundreds of suppliers and generally they encode ok, but sometimes we are getting unexplained errors. I've looked through the archives for the last few months but I can't see any discussions that relate to this specific problem. This file:

"Alex Brims" <alex.brims@gmail.com> wrote: > Ok, we actually worked this out - there were 2 extra bytes doing nothing at > the end of the files. Opening the file in SoundForge and saving it (without > changing it) took off the extra bytes and allowed the file to convert to > FLAC. > > Thanks to everyone who emailed me suggestions. > > Is there a decent program for linux that could automatically take these > bytes off, without running the risk of rem...

...scussion <mailto:asterisk-users@lists.digium.com> Sent: Wednesday, January 26, 2005 5:36 AM Subject: [Asterisk-Users] Am I missing something really basic here????? helpwith Asterisk@home {Scanned} I'm trying to install asterisk@home, I've just downloaded the latest cd from soundforge. I can get it to install ok (network card didn't auto configure - but I worked out how to use 'netconfig'). I worked out how to add a few grandstream budgetone fine. Worked out how to upload music etc. Worked out how to modify FOP. Voicemail and meetme's work fine....

"Didier Dambrin" <didid at skynet.be> wrote: ... > I like FLAC on the paper because of its metadata preservation, in that riff > tag, which is critical for my needs. Try using WavPack, http://www.wavpack.com/ This can losslessly compress 32-bit floating point WAVE-EX files, and faithfully preserves every chunk (which FLAC does not do). It is also free. Regards, Martin --

Ok, we actually worked this out - there were 2 extra bytes doing nothing at the end of the files. Opening the file in SoundForge and saving it (without changing it) took off the extra bytes and allowed the file to convert to FLAC. Thanks to everyone who emailed me suggestions. Is there a decent program for linux that could automatically take these bytes off, without running the risk of removing good data? Or is there a w...

I am using Audacity to record some voice prompts. The .wav files I'm producing are of stellar quality. However, once I turn them into .gsm, they sound buzzy and muffled. I know that some of this comes with the territory, but I wonder if there is anyone out there who does this routinely, and who can advise me as to the MO I could use that results in the highest quality in the resulting

...encoder_example to encode e.g. a 1 sec 16 bit stereo wav file, when I decode using decode_example I get clipping ... any ideas here???? I can hear the original in the clipped mush! The raw file plays just fine (on the SGI) and I have even decreased the gain to 80% and removed any DC offset (using Soundforge on my PC). regards, Rob. P.S. Am am in big need of 4 channel encoding ... anyone done this yet? +============================+====================================+ | Dr R P Fletcher (Rob) | Email R.Fletcher@york.ac.uk | | Graphics Coordinator | Phone +44 (0)1904 43...