similar to: encoding 22 kHz

Hi, I've been using FLAC for several years, but mostly with 16-bit files (I never actually checked until now, but that would be my guess). I recently got errors in Nero trying to load WAV audio files decoded from FLAC, and the only (obvious) difference between these and all the others that work w/o probs is that these input files are 24-bit. I didn't think there were special settings

Hi, I'd like to know the reasons why CELT supports only signals with sampling frequency in the range of 32-96 kHz. In effect, it can clearly outperform speex at high bitrates, and has potential to be used in high quality voice communications even for 11, 16 and 22 kHz speech signals. It could also compete with SILK codec (to be soon released by Skype). See this page for more specifications

Hi, Does anyone know of a technical reason why FLAC cannot support 88.2 Khz files? I have a reason to uses this rate since it is easy to perform quality conversions from 24 bit 88.1 Khz master files (stored as flac files) to 16 bit 44.1 khz files for CD mastering purposes. I suppose I could Kludge the wav files so that they were half speed wav files at 44.1 khz and then hand the over to Flac, but

You asked in the Vorbis list, but your text only mentions OGG. The codec commonly used in OGG containers that is limited to 48 khz is Opus. Maybe you are trying to use the wrong codec (i.e. Opus instead of Vorbis)? Using a 44.1 khz wav file, I was able to encode a 192 khz ogg-vorbis file with the following command: $ oggenc --resample 192000 input.wav Of course, if your original material is

I found some posts regarding this issue dating of September 2003, but no real answer. The ADPCM format supported by Asterisk (the .vox files) is 4-bit, 8 kHz. I need 4-bit, 6 kHz, which is also a widespread Dialogic format, to help migration. Is there an existing format/codec for this? If not, can I make myself a shared object in /usr/lib/asterisk/modules? Is this easy??? :-( Thanks, Yves

Hello, congratulations to the Ogg Vorbis team - RC2 sounds good. But... RC2 in 128 kbps mode seems to clip off all frequencys beyond 16 KHz. On the tracks I tested Beta 4 gave response even beyond 18 KHz. Some testings on a randomly chosen track: (other tracks gave similar results) Artist: Judas Priest Album: Jugulator Title: Bullet Train Beta4: 127 kbps, ~ 18 KHz (!) RC2: 132 kbps (!), ~ 16

Hi there, I've been playing with a SNOM 360 and 190 trying to get them talk to each other using g722 with 16 kHz. However all I see in the SIP log codec negotiation is "g722/8000" which makes me believe that this is only a 8 kHz link (and that's what it sounds like). Anyone every managed to establish a 16 kHz wideband call between SNOM phones? Cheers, Philipp

On May 13, 2007, at 05:45, Harry Sack wrote: > If I encode 192 kHz sound @ 24 bit for some days (WAV file) and I > encode it to FLAC, I think you can have a very big file and 1.5 TB > is reached very quickly. > And in the future audio will even get bigger, when used for HD-DVD > en Blu-ray media and 5.1 channels is considered the 'minimum' > setting for surround

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

________________________________ 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 list, I have an 44100 Hz file with human voice, stereo with 16Bit. When convertig this to 8 kHz, mono I loose a lot of quality and have some ground noise. I tried several sox options but without success. Can somebody help.... best regards Thomas

Hi everyone, I recently began using libspeex 1.2 Beta 1 on Windows using MS Visual C++. I have gotten a decoder and an encoder to work fine from the excellent sample code posted at the website. But I face a problem. I am working on using Speex in a program to play and create audio books encoded using Speex (currently testing it only; for these tests, I do not use Ogg to save the encoded

hi I have some questions about re-encoding existing FLAC files to FLAC 1.2.0.: - can older 1.1.x FLAC files be re-encoded to FLAC 1.2.0 by using the FLAC 1.2.0 encoder? - can FLAC files encoded with the FLAC Flake SVN encoder (or any other 'unofficial' FLAC encoder) be re-encoded by using the FLAC 1.2.0 encoder? thx in advance! -------------- next part -------------- An HTML attachment

On 5/13/07, Brian Willoughby <brianw@sounds.wa.com> wrote: > If we ever reach this 64 GigaSample limit, the fact that FLAC is a > stream should allow multiple FLAC headers to be concatenated in a > single file - although that might be tricky. I believe you can do this with Ogg FLAC. The Ogg container manages the multiple FLAC streams. Theoretically, we'll never see limit

Hi!, I wondering if It's possible to change 48khz sample rate limit?, I'm Planing to encode with OGG codec a audio signal but I need that OGG Encoder works with 192khz of sample rate. It's Possible? Any Suggestions?

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/7/25, Harry Sack <tranzedude@gmail.com>: > > > > Hi > > > > I have downloaded a FLAC file somewhere and when trying to decode > it to > > WAV it gives the error message: ERROR, MD5 signature mismatch > > So my question is now: are FLAC files that give the error message > above > >

Chuck, I'm doing a very similar project. I'm not an audio expert, but here's my take on the subject. 22KHz will take more space, but it is higher quality, and would probably be better if you ever decide to put them on CD (44.1KHz, stereo is the required format for CD Audio). There was a noticeable difference in the two when I tried recording at 11KHz vs 22KHz; I ended up just

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

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that

Sorry, but I think generic extension names are far from perfect. Here are some additional problems to consider: 1) Language. When people talk about file types, they almost never say "dot" at the beginning. They say "MP3 files". For example, "Does that player support MP3 files?" If you have an extension of ".music" this ends up being "Does that