search for: reverb

Dear all I have a problem with splitting up a data frame called ReVerb: ?? str(ReVerb) `data.frame': 92713 obs. of 16 variables: $ CHILD : Factor w/ 7 levels "ABE","ADA","EVE",..: 1 1 1 1 1 1 1 1 1 1 ... $ AGE : Factor w/ 484 levels "1;06.00","1;06.16",..: 43 43 43 99 99 99 99 99 99 99 ... $ AGE_Q...

...where it can flatten or expand the dynamic range of an entire channel (or the entire recording) without any artifacts. Equalization: 30 band -60/+30dB 1/3-octave beat-less EQ per input channel and full output group. Deverberator: Live recordings have a tendency to end up with too much reverb, especially when one is forced to use ambient miking. The deverberator dries out overly 'wet' live signals. Reverberator: ...for adding reverb to signals that are too 'dry', especially to even out apparent depth when mixing close miked signals (like vocals) to an ambie...

...where it can flatten or expand the dynamic range of an entire channel (or the entire recording) without any artifacts. Equalization: 30 band -60/+30dB 1/3-octave beat-less EQ per input channel and full output group. Deverberator: Live recordings have a tendency to end up with too much reverb, especially when one is forced to use ambient miking. The deverberator dries out overly 'wet' live signals. Reverberator: ...for adding reverb to signals that are too 'dry', especially to even out apparent depth when mixing close miked signals (like vocals) to an ambie...

I'm experience echo on outgoing calls: Snom 200 ----> Asterisk ----> T100P ----> PRI ----> called party I am getting echo on the Snom 200 phone. The called party does not hear the echo. Since the PRI is digital, I don't really understand where the echo is coming from. I turned on echotraining, echocancellation=yes (128), and echowhenbridged in zapata.conf, to no avail.

Oooh Plate Echo was what people had before digital reverbs! (When they didn't have spring reverb!) - Long ago... They are quite rare now because only big studios had them as they took up a lot of room! (and then they threw them all away) *jealous* Do you really have a plate reverb or just a digital simulation? love Freya --- Robert Voigt <robe...

...show the picture at all. The files with pictures I tested are MaxMoritz.flac and Rabe.flac. You can find them in that directory: http://akfoerster.de/download/audio/AKFoerster/ It's in German language. By the way Rabe is a translation of "The raven" by E.A.Poe. In that file I used a reverb effect, which seems to get lowered by encoding. Is that intentional? Is there a way to keep the effect intact? -- AKFoerster <http://AKFoerster.de/>

Hello, I'm using Speex1.1.11.1 source code. I enabled dereverb by speex_preprocess_ctl(). I try different dereverb_level and dereverb_decay values and it seems to not work. I generate some test files in CoolEdit using reverb or echo and then preprocess it by speex_preprocess(preprocess, input, NULL); and output is the same as input:(. Denoising and VAD works v...

Hello Jean-Marc and all, I?ve got a problem with speex_echo_cancel usage. I only managed to get an attenuation of 5-8 db using testecho.c and it depends heavily on the playback/capture delay. I used clear speech as far-end input and same speech with a bit of reverb, attenuation and delay as mic nput. I also tried using played and recorded speech for mic input. I tried different constant delays and got the following resuls. When the delay is 50 ms, the attenuation decreases twice, and with a 100 ms delay there?s almost no echo cancellation. Unfortunately, the...

...esting various echo cancellers. To simulate the "far end" speech, I have a clean close-mic recording of an utterance which I pass as the "echo" argument to testecho.c. To simulate the mixture coming from the mic, I delay that same signal by 10 ms, add a mild amount of artificial reverb, reduce the volume by several db, and then mix in a different clean utterance. That is passed as the "ref" argument. This test is slightly difficult because there is more double-talk than single-talk in it. Also, the 10ms delay is fairly long. However, other vendor's echo cancellers...

Hello Jean-Marc, I solved the variable delay problem, but I still have trouble with speex_echo_cancel(). When i try testecho.c with clean speech for far-end input and same speech with attenuation, a bit of reverb and 50-150 ms delay, all this done in sound editor, for mic input, i get 5-8 db attenuation. But when i use the same speech played and recorded for mic input, i see about 5 db of attenuation only if i remove the delay (150ms) at the beginning of mic input file to make far-end and mic input file...

...I can still hear the echo signal (althought > the signal is a little more silent. when using the delay 3 buffers the > echo is still not cancelled and after few seconds I don't hear anything. when using about 10 buffers for delay between these signals the echo signal is heared as it was reverbed, everything else is ok and the noise is removed. how to completely remove the echo signal? regards hs _______________________________________________ Speex-dev mailing list Speex-dev@xiph.org http://lists.xiph.org/mailman/listinfo/speex-dev

...g sticky-sheet to the inside of the case. Problem gone - no more echo. So, once again, you may be having a headset/handset-related problem! > I noticed, something, though: While I only kept G11u on both the XLite > and Siemens, I noticed that sound coming from the Siemens contains > some reverb when running Asterisk (1.4.4) on an Atcom appliance > (www.atcom.cn/IP01.html), while the reverb is gone when running > Asterisk (1.6.2.5) on a regular PC with Ubuntu. I guess codecs sound a > bit different depending on what hardware they're running on. That's odd... the U-law cod...

Hi! I want to use speex for echo cancellation in my program, but I have bad results. I will explain what my program does. it is a client-server application. I run a server in room A and a client in room B. the client sends some voice to the server and the server plays it on loudspeakers. I run another server in room B and connects to it from room A using the same application that runs

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

I've replaced 2XTE110 with an A102 with echo cancellation specifically to deal with echo problems. However, user feedback has indicated to me that on some calls (not a lot, but some) the call is unusable, with audio artifiacts, described by one user, as: "very bad phasing reverb & feedback (from my rock & roll days)". This is quite intermittent, as in most cases, the user says, it's a one-time thing, they hang up, the problem caller calls back, everythings good. It's as if the Sangoma is trying too hard? I personally have not heard this, but I have to...

Hello, I'm new on this list. First of all, thank you to all the developers of this great project! At the moment there is only an application that keeps me on both macos and windows, its name is PWGL a free environment for computer assisted composition in openGL. (http://www2.siba.fi/PWGL/) I'm running Ubuntu 8.04 and wine 0.9.59. I have to say that I also installed vcrun2005 and

...e -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al au auto avr cdr cvs dat vms gsm hcom la lu maud mp3 nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve which let's me know that my versi...

Richard Lee wrote: > But there is a caveat. Does Vorbis coupling preserve "phase" relations? Any references explaining this in simple detail? Sebastian : >"coupling" is a rather loose term in the sense that it doesn't exactly specify HOW it's done. It can be understood as the opposite of coding channels independently. Obviously there are many ways to do

...opts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Which package contains libmad and libm...

...- voice disguise/modulation. Think about how many customers you'd get with a module that sounds like they're Mickey Mouse. You think: 'That's really stupid!' but then look at how many people pay for Usher ringtones. Look at how many crazy truckers buy CB microphones with reverb. - voice stress analysis. If you're dumping the audio through a filter, there's no reason you can't simply extract data from it instead of alter the audio path. A one-way background audio carrier tone to the listener might change pitch during stress events. - customized backg...