similar to: Speex development

Essentially, the thing that's been holding up 1.2 was the AEC and preprocessor. The codec itself is pretty stable and I do not really see more work happening on it. As for the AEC and preprocessor, these are really open-ended project and I lack sufficient time to take care of them. Jean-Marc On 10-11-12 03:27 AM, Ken Smith wrote: > Basically, Speex is the brain child of Jean-Marc

Just an update, and a follow-up question: I'm making progress on this issue, and will likely have something working very soon, now that I understand how the jspeex transcoding classes work. Unfortunately, I will need to make a couple small changes to the jspeex source code. My question is - is anybody using jspeex for streaming speex-encoded audio? It works great for static audio, but seems

Hello, We've integrated the speex echo canceller into PulseAudio, and coupled with the AGC/denoiser, it works reasonably well and has been a breeze to plug in, so thank you! One thing we're seeing, though, is that the canceller seems to take some time (a few seconds) to "learn", so initially the echo is clearly audible and it slowly fades out. This sometimes occurs in the middle

Basically, Speex is the brain child of Jean-Marc Valin, and from what I can tell, he has since moved on to other projects, though he continues to monitor this list. So no, it's not really being developed anymore. And what's out there really is good enough for a great many applications. That said, there remains room for improvement, mostly in the pre-processor (the codec itself is very

Hello. I'm from Federal University of Rio Grande do Sul - Brazil and I'm trying to adapt speex_echo.h (speex-1.2beta3-win32) to our video-conference software code. We are working on amplitude values, but I realised that in order to AEC works, one has to convert amplitude to frequency and filter the echo frequencies. So, my question is: Do I have to manually convert amplitude samples to

I'm developing an application that have a video conference component. For that I need echo cancellation, and is looking around for algorithms/implementations of that, and the one in speex is an alternative. In the documentation for speex I find the following sentence however. "Using a different soundcard to do the capture and plaback will *not* work, regardless of what you may

From 1.2rc1 source code: preprocess.c: /* FIXME: This VAD is a kludge */ st->speech_prob = Pframe; if (st->vad_enabled) { if (st->speech_prob > st->speech_prob_start || (st->was_speech && st->speech_prob > st->speech_prob_continue)) { st->was_speech=1; return 1; } else { st->was_speech=0; return 0; } } else { return 1; } AND case

Hi,

Hi, I implemented a small patch that allows the internal convergence state of the echo canceller to be saved in a file for later use, especially after a process restart or machine reboot. This enables immediate echo cancellation the second time the AEC is run. Of course this works only if the acoustic environment of the device doesn't change and if the soundcard latency is constant. To use

I'm completely new to speex and I'm having issues adding it to my application. I can pass raw PCM and that works fine so I know my transmission code is fine however when I try to encode/decode using speex I get noise almost like a whining/buzzing sound. I'm sure it's and issue in my code but I'm not sure where to start looking other then my gut tells me I'm not sizing or

Alright noted, I changed me code so that the state is created in the constructor and destroyed in the destructor of the object. However I'm still getting the same issue although I'm sure that would have bit me sooner or later. The new code is as follows. virtual Enigma::u8* Encode(Enigma::u8* inputBuffer,size_t inputSize, size_t& outputSize) { short *in=(short*)inputBuffer;

Hi, I couldn't find a discussion that specifically addresses this, so here it is. I'm using Speex AEC in my mobile VoIP application to cancel speaker echo. The used version is 1.2rc1 from the website, and I'm compiling with fixed-point. On most occasions, the AEC works very well and cancels most of the echo (combined with the preprocessor). On some devices, where the microphone signal

> As you can tell, the AAGC integration with speex was really a classic > hack. Instead of re-creating the hack, what's probably best here is to > integrate AAGC back into speex, and have a proper API. Agreed here. If you can come up with a clean patch to add that feature, it's something I'd like to see in Speex. > For those of you just tuning in, what I call

> I hate to be a talker and not a do-er, but I won't be able to write this > myself, probably someone on the iaxclient team could do it. Anyway, let me know if/when someone's working on that. >> Hmm, or does that mean the analogue AGC is actually completely >> independent from the "real" AGC. Any thoughts? >> > > It's actually a bit more

Oh I overlooked that one.. Thanks Darren! Btw, I was wondering if anyone has tried using the echo canceller function of speex with other codec like g.711? On Tue, Jan 17, 2012 at 8:32 PM, Darren Longhorn < darren.longhorn at redembedded.com> wrote: > On 14/01/12 12:26, Christopher Adoremos wrote: > > What is the highest quality audio data rate and sample rate Speex can >

Hi, Arun and me are working on a echo-canceler module for pulseaudio build around speex. When using the preprocessor's echo filter, specifically in combination with denoise and AGC, it's unclear whether these have to run before or after the AEC. Right now we've got one instance of speex_preprocessor doing AGC and denoise running before the echo canceler; and a second for echo

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

I have a couple apps running on my machine; one takes a PCM audio stream, converts it to Speex, and sends it over the network using RTP. The other receives the RTP packets, and then converts the Speex data back to PCM. The PCM is then played out the audio system. I'm currently ending up w/ static. Anybody have any pointers as to how to debug the situation? Thanks. -- Jeff Ramin Software

On 06/22/2011 04:57 AM, Arun Raghavan wrote: > On Tue, 2011-06-21 at 11:39 -0700, Arun Raghavan wrote: > [...] >> I'm also running this on x86 (x86_64, technically), and it's all >> floating-point, so I guess this is a regression somewhere. Will try to >> see if I can run it without any optimisations if possible, which I >> assume should serve as an adequate

Hi Jean-Marc, thanks for the response. First, I will clarify again that floating-point solves this - so isn't that a bug in fixed-point? Also, I understand that algorithmically the AEC won't cancel echo properly on a non-linear signal, but why completely distort the output? If the echo just won't get cancelled it would be acceptable, but in the current state it disables the ability to