similar to: Variable frame size and API changes

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

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, 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

Hi all I know that this has been tested for one digital radio standard DRM+. I'm wondering if anyone has tried to test opus on the DAB digital radio system. Peter

I've been playing with speex for use in a VoIP application between PC's. One thing I've found (correlating to the documentation) it that speex runs much faster and produced much better output when it's fed a 32kHz signal instead of a 44.1kHz sample rate. This is whether I tell it a 44.1kHz sample rate and feed it 44.1kHz or tell it 32kHz and feed it 44.1kHz. What part of the

Hi, I was just trying to use speex for sampling frequency >48KHz. In the original Speex-1.0.5 its restricted only upto 48KHz. I tired to modify it by changing the boundary conditions( the error conditions, i.e. if sampling freq >48KHz, it gives error) in /src/speexenc.c and then it atleast doesnt give the error, there is flow in decoding or encoding(i think). I suspect there are other

Hi all, Just a quick note to let you know that we are currently integrating CELT with our DAB transmission and reception software tools. We plan to have a working demo at the broadcast conference IBC 2009 in September where we have a small booth. We hope to be able to demonstrate a real-time DAB CELT encoder as part our our Live CD based on Ubuntu and GNU/Radio (

Hi, is there any way to tell the decoder the output sampling Fz we want ? opus_decoder_create = Sampling rate of input signal (Hz) Considering this example (VoIP-out from WebRTC/RTP) MICROPHONE(44.1/48kHz) >> [encoder created at 48kHz but with internalSampleRate set to 8kHz]>> INTERNET >> [decoder(created with 48kHz)] >> 48kHz(?) >> G.711(8kHz) This leaves us with

Just to make sure, what's the goal here? Is the goal 1) to have a fast Opus implementation or are you 2) looking for an interesting FPGA implementation project? If 1), then an FPGA is most likely not necessary since Opus is not computationally expensive. If 2), then it depends on the desired size of the project and the desired quality. The simplest encoder possible is indeed simpler than the

----- Original Message ----- From: "Conrad Parker" <conrad at metadecks.org> To: "Vincent Burel" <vincent.burel at vb-audio.com> Cc: "Jean-Marc Valin" <jean-marc.valin at usherbrooke.ca>; <speex-dev at xiph.org> Sent: Thursday, November 13, 2008 1:18 AM Subject: Re: [Speex-dev] SPEEX on iPhone ? > 2008/11/13 Vincent Burel

Hello, To start with, thanks a lot for making such a great voice codec available! Having recently upgrading to speex rc1, It occurred to us that there seems to have been a regression in the quality of encoding since version beta1. We are compressing some 22khz wave files in wb mode with maximum quality / complexity in VBR, and the result was really great with speex beta1. With rc1 (or beta3),

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 Jean-Marc, On Jun 15, 2013, at 2:23 AMEDT, Jean-Marc Valin wrote: >> I'm looking at how to run Opus at 44.1K. I have flexibility in the >> frame sizes of the unencoded audio, and packet sizes on the RF link. > > You should probably consider resampling. It's not that expensive and it > would make things easy. But otherwise, see below. Yes, considering your and

int frameSize; speex_encoder_ctl(enc_state, SPEEX_GET_FRAME_SIZE, &frameSize); Is frameSize in bytes? If not, what unit is it in? I need to know so I know how big to make each element in my circular buffer. Also, do I need to call speex_encoder_ctl( enc_state, SPEEX_SET_SAMPLING_RATE, &sampleRate ); Depending on sample rate I record my audio in? If not, is there any benefit to

----- 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:

Would speex be a valid method for compressing/decompressing audio for live communications? Trying to find something that I can use that would decrease the bandwidth requirements tremendously, and still be able to be performed quick enough for live discussion over the air between two communications devices. I have connected two communications devices via the Ethernet, streaming audio via UDP, and

On 6/7/14, 1:55 AM, Jean-Marc Valin wrote: > Actually... no! 24-bit can indeed be useful as extra margin and Opus > can actually represent even more dynamic range than 24-bit PCM. That's > not the case for 192 kHz. There's no "margin" that 192 kHz buys you > over 48 kHz. You can do as much linear filtering as you like, the > stuff above 20 kHz isn't going to

Hi, We would like to use the OPUS codec @ 16 kHz sampling rate and max 32 kbps. What about implementing an OPUS coder and decoder in an FPGA? Has this been done? Would either coder or decoder more suitable for FPGA implementation? Best regards Fredrik Bonde -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hi everyone, I've just changed the API for CELT, but at least there's a good reason for that. It's now possible to use the same mode data for both mono and stereo. So here's the change: - The celt_mode_create() function has a "channels" parameter - The celt_encode_create() and celt_decoder_create() functions now have an additional "channels" parameter. - I