similar to: CELT 0.4.0 released

Hi everyone, I'm pleased to announce version 0.6.0 of CELT, with many quality improvements. This includes better stereo coupling, better handling of transients, and better handling of highly tonal signals. In addition, packet loss robustness has been improved through the optional use of independent (intra) frames. CELT now supports a larger dynamic range, suitable for encoding 24-bit audio

Hi all I have spent the last three days evaluating CELT on our supported platforms. I found a bug in quant_bands.c, that due to processor/compilation differences did not cause an issue on x86 platforms, but is a problem on the MIPS processor embedded devices. When decoding on the MIPS devices, there was a lot of noise added during the decoding, the noise is mainly in the 15 khz to 21 khz range.

Hi everyone, Just wanted to let you know that 0.8.0 is out. You can get it from the usual place and here's the small notes that go with it: "CELT 0.8.0 brings quality improvements for high bit-rates and for transients in general. It also adds the possibility of changing the frame size dynamically, which requires a minor change in the API. At last, the new PLC algorithm from 0.7.1 is

Hi, FYI, I'm proposing the following update to the Opus spec to improve low-bitrate hybrid quality. If you'd like to comment, you can reply to this email: https://www.ietf.org/mail-archive/web/codec/current/msg03247.html Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [codec] Last proposal for the update draft Date: Fri, 17 Jun 2016 08:26:04 -0400 From: Jean-Marc Valin

hi. i am currently adding celt support to netjack. very nice to see a free low-latency codec :) i currently dont require robustness against packet loss, because the sync code of netjack does not handle packet loss very gracefully. how much bandwidth is wasted for this feature ? is it sensible, to have the data downsampled berfore encoding , in order to reduce bandwidth ? i suspect that 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

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

Aymeric Moizard a ?crit : >> None of that is defined yet, though I'm open to suggestions on how to do >> the mapping. > > CELT/44100 and CELT/48000 > a=fmtp:105 stereo=on > > probably a latency value? It would definitely need a frame_size value > "CELT" doesn't seems to be used for any other existing audio related > stuff? right? Not yet.

Hi, folks, I have an old CELT thing that I'm updating to Opus and I'm trying to get individual pieces upgraded.  My first task is upgrading the CELT encoder to Opus. The old CELT thing treated the L and R channels as independent mono streams, encoded them somehow with CELT to form one bitstream, sent one bitstream across, and then unpacked it manually after the decoder. Can I do that

> As the thing is to encode for human ears (AFAIK), I'd say that 4kHz is already "quite high", > and I wonder who can actually hear pure 20kHz sine. If you read the beginning of RFC 6716, you learn that Opus never encodes any frequencies that are higher than 20 kHz. So at some medium or high bitrates, anything above 20 kHz is filtered out, not because of the bitrate but

Hey everyone. I've been designing my own audio codec with extremely strict decode-performance constraints (including a fixed block size), which led me to attempting a number of unorthodox things to squeeze as much quality as possible. One surprising thing I discovered just earlier today was an extremely cheap method of reducing pre-echo during transients, without using short blocks (and

On Tue, 14 Oct 2008, Jean-Marc Valin wrote: > Diana Cionoiu a ?crit : >> Hi Jean-Marc, >> >> 1. Why CELT versus Speex? > > There's nearly no overlap between Speex and CELT. CELT is much higher > quality, very low latency, and of course, higher bit-rate than Speex. So > it really depends on the application. > >> 2. What's the RTP payload for CELT,

Hi everyone, I've made several posts recently about CELT being "replaced" by the Opus codec ( http://opus-codec.org/ ) and I thought it was time to give an update on what's going on. As many of you know, I've been involved at the IETF on this new Opus codec, which essentially merge (a modified version of) Skype's SILK codec with CELT. This is more than just two codecs

On Thu, 16 Oct 2008, Jean-Marc Valin wrote: > Aymeric Moizard a ?crit : >>> None of that is defined yet, though I'm open to suggestions on how to do >>> the mapping. >> >> CELT/44100 and CELT/48000 >> a=fmtp:105 stereo=on >> >> probably a latency value? > > It would definitely need a frame_size value > >> "CELT"

Hello all, This is slightly off-topic, but should be of interest to some people on this list. I just released version 0.3.2 of the CELT ultra low-delay audio codec (http://www.celt-codec.org/). CELT is designed to encode high quality speech and music with less than 10 ms delay and at rates starting from around 32 kbit/s. This version is "special" in that it is the basis for some

SUMMARY I'm impressed. RC3 withstood anything I could throw at it. "-q 3" is really Vorbis' sweet spot, almost perfect, and "-q 0" is eminently usable, with only marginal defects for normal usage. Already sent money, will do that again. * * * I spent at least 40 hours in the last week testing RC3. I selected a few fragments from pop, classic and jazz CDs in my

Hi, (this is my first post here) A previous thread, starting Date: Tue 19 Nov 2002 - 06:09:56 EST "[vorbis] need speech and music in one" http://www.xiph.org/archives/vorbis/200211/0142.html expressed needs similar to mine, to encode a lengthy speech at low bitrate. I did some tests initially in September then concluded in December, and I was surprised to find Vorbis to be the best

can you file a bug on sourceforge: http://sourceforge.net/tracker/?group_id=13478&atid=113478 and attach the first 100k of each file? it will help to debug since I don't have any wave files that cause this behavior. Josh --- Free Lunch <freelunch@gmail.com> wrote: > Okay.. I love flac but just had a rather serious failure that really > shakes my confidence. It resulted

Okay.. I love flac but just had a rather serious failure that really shakes my confidence. It resulted in the near loss of a master audio recording. Fortunately, I have a backup. Though there may have been other cases where I have lost original material because I have been compressing a lot of originals and deleting them after doing a 'flac -t' on them. Basically, flac failed with a

Hi, I'm currently testing libcelt (celt-0.9.1) on my Openwrt/Kamikaze RDC-3210 platform and although I'm able to Crosscompile on my Ubuntu 10.10 machine without errors, the resulting celtenc doesn't seem to work properly. Ran ./configure via the Openwrt makefile first (activating fixed-point) ; I first produce a S16_LE Raw PCM file (48Khz, stereo) : test.pcm Then I try to encode