similar to: 88.2 Khz files

In this topic on Hydrogen Audio(http://www.hydrogenaud.io/forums/index.php?showtopic=105586) someone asked a question about the sample rates that FLAC supports for ReplayGain. The outcome was that the current documentation of MetaFLAC is outdated since Commit http://git.xiph.org/?p=flac.git;a=commit;h=0554a4aee6966bc5b251364753ef85de72dfab19 because as of 1.3.0 FLAC supports Replaygain with many

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

Hi, I don't think this is the right list but since it is the only speex related list on the XIPH list, her it goes. Flame away if you wish or, if your feeling generous, tell me what list to post on. How do you "denoise" only without compressing the file? The documentation implies it is possible. It says on page 9 of manual.pdf, "The denoiser can be used to reduce the amount

________________________________ 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

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!, 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?

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

Hello, I just installed the openssh 4.0 for Solaris. The users have reported a difference in behavior when using the batch mode of sftp client. Previously they could issue the following command sftp -b batchfile user at hostname and in the absence of publickey authentication they would be issued the password prompt and they could enter password and the process would continue. After upgrading from

Is there a list somewhere of "standard" encoding rates? I know, for example, CDs are encoded at 44100, as is a lot of digital sound, but I've seen programs that specify different levels of quality (like radio, phone, tape, CD) and I'd like to know if there are some encoding rates that are accepted as standardized for recording at different levels of quality. If so, is there

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

hi, is it possible to encode 16 bit, 22 kHz, stereo/mono WAV files to FLAC files or could there be a problem with the low frequency 22 kHz (lower then CD quality)? PS: I'm a FLAC beginner thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/flac/attachments/20070512/63ed58f6/attachment.html

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, another question on the same topic Speex resampler at 44.1kHz seems to be very CPU intensive on Android (even more than the Opus encoder) While Speex at 48kHz is just fine. I wonder any alternate solutions or ideas ? Improve it, look for alternate solution ... I am guessing the NEON optimization are still used for both, etc. On Thu, Apr 2, 2015 at 4:46 PM, Jean-Marc Valin <jmvalin at

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

Hi Peter, Have you considered resampling the raw 44.1kHz stereo source files using a program such as http://audacity.sourceforge.net/, say to 16kHz mono or 32kHz mono, and then using the wideband or ultrawideband speex modes to encode the result? Alternatively, if you want to programmatically do the resampling yourself, you could try the new resampling API in the svn head of speex, or the GPL

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

----- Original Message ----- From: "Alexander Chemeris" <Alexander.Chemeris at sipez.com> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Conrad Parker" <conrad at metadecks.org>; <speex-dev at xiph.org>; "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca> Sent: Thursday, November 13, 2008 11:31 PM Subject: Re:

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

Hi everyone, I've recently been making various changes to the way the modes work and the supported frame size. On new feature that may be of interest to some is that CELT should soon support changing the frame size dynamically within a stream. By that I mean varying the amount of audio (in time) transmitted at once, not the compressed size -- which has always been variable. That would