similar to: Re: Welcome to the "Flac" mailing list

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

dd if=$file ibs=1 count=$(($(stat --printf='%s' $file)-2)) of=$file.new of course if you run this on one 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

"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

That's a handy command, but I'm certain it won't work 100% for the file in question. The chunks in that bad file claim the extra two bytes are part of the file, so a wav format parser could come up short. You have to edit existing data in the file in two places before shortening the file - truncating the file is not enough by itself. The real problem is that the file was

that's why i asked the original poster if the files were odd size, i had that issue before with a 24 bit mono file and wrote this script to fix it: #!/bin/sh # # sfoddfix - Sound File ODD size FIXer # # NOTE: flac v1.1.2 pukes on files that have an odd byte count, this pads them files=${*:-*.wav} for file in $files do size=$(stat --printf='%s' $file) if [ $(($size%2)) -ne 0 ];

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

I'm new to the mailing list but am interested in picking up a thread from earlier in the month but which I thought had become confusing so I am starting again. I should admit from the beginning that I am a colleague of Alex Brims who started the original thread. The thread in question related to a wav file with an extra two bytes at the end causing a partial sample error in the reference

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

> > ( icecast2 only handles .MP3 for fileserving , as far as i know ..) > > Nope. Icecast2 doesn't care about what the format of the files it serves is > for fileserving. (i meant :: .MP3 is handled for fileserving , but .MP3 is not avail for live reencoding, as far as i know .. ) for the record , the OGG streaming w/ icecast2 has been great . tho (seperate issue ) : OGG files

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 jean-marc and everybody, 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

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

I'll rearrange a little and respond: --- Mark Powell <M.S.Powell@salford.ac.uk> wrote: > Also, when flac takes input from stdin it fails to > fill in the wav size > fields correctly, whereas shorten has no problems > with this: i.e. >... > You can see it puts a data chunk size of zero in > there. > OK, this has been fixed in CVS. > Flac refuses

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 --

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

i downloaded the vorbis, ogg & example files and compiled them and now have an executable called 'vorbisfile' that turns 'test.ogg' into 'out.pcm'. is there some simple way to test the out.pcm file in winXP? ~~~~~~~ bob hurt I can't stand cheap people. It makes me real mad when someone says something like "Hey, when are you going to pay me that hundred

Hello, I just signed up to this mailing-list (note: my first mailing list at all), because I'm having some problems related to speex. Let me just introduce you to what I'm doing. I am writing a short (really short) paper about VoIP techniques, especially audio codecs for speech. I pointed out basic technologies behind audio codecs; vector quantization, lpc, long-term prediction and some

Ok, I thought the point of asterisk@home was that it automatically detected the X100P board and configured it correctly. Is this incorrect? You still need to modify /etc/zaptel files? And not just using the AMP configurator. There is no mention of this on the Asterisk@home webpage. Can anyone who has actually used ast6erisk@home confirm this one way or the other? Thanks, Dean

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