search for: reverber

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 : num 2.0 2.0 2.0 2.4 2.4

...zation: 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 ambient-miked signal. The Postfish provides a stereo reverb per input channel and a mono reverb per output post-mix. Limiting: S...

...zation: 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 ambient-miked signal. The Postfish provides a stereo reverb per input channel and a mono reverb per output post-mix. Limiting: S...

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 <robert.voigt@gmx.de> wrote: > uh, you

Hello, I have problems with encoding FLAC-Files with embedded pictures. Currently i use opus-tools 0.1.7 with libopus 1.1-rc2. After encoding opusinfo shows WARNING: Unknown picture type in METADATA_BLOCK_PICTURE comment 12 (stream 1): 1768776039 WARNING: Invalid mime type length in METADATA_BLOCK_PICTURE comment 12 (stream 1): 1697607790 bytes when 5562 are available When i play the files with

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 very good. I tried dereverb with

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

Hi Folks, I am having trouble using speex_echo_cancel. As a starting point, I am using the testecho.c source code. I compiled the 1.2beta1 version. I have not tried any other versions of speex. The document says that the order of arguments to Speex_echo_cancel is (echo_state, input_frame, echo_frame, output_frame, residue) where "input_frame" is as captured from mic and

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

Hi, Indeed I too have troubles implementing echo removal, I like ask kindly to Jean-Marc (or any other) if him can put a source code demo to show us how to use effectively echo removal API and parameters in real case scenario. A big thank you! Roberto -----Messaggio originale----- Da: speex-dev-bounces@xiph.org [mailto:speex-dev-bounces@xiph.org] Per conto di hs Inviato: marted? 11 ottobre

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

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

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

Hi. I'm wondering if anyone out there can give me a little help trying to figure something out: I presently am running CentOS4.1 with GNOME as my desktop (It runs quite well, I must say!). I just have a little curiosity: I have a few audio files (.wav, .mp3 and .ogg) and I like to "preview" them. When run my mouse over the .wav files (and mp3 files) the

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

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

At 11:08 AM -0500 7/12/04, Steven Critchfield wrote: >[snip] > >Would it be something people would like to be able to add filters to a >line? Consider normalization as a filter. Monitor could then be moved to >a filter as well. Echo cancel could be a filter. Set it up so multiple >filters could be added and chained together. This could help those with >echo chain a couple of