similar to: parts of data frames: subset vs. [-c()]

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.

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 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

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, 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

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

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

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

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

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

Is anyone running Heroes 3 with a late version of wine? Or Heroes 1 / 2 ? I cant get any of these to work. My problems are posted to wine-devel, I am just wondering if anyone else is able to run it. /Rustan Ros?n Sweden

I have a box behind a firewall that I'd like to administer. The firewall allows outgoing connections, but blocks all incoming connection requests. Thus, behind the firewall I can ssh out to my server, but I can't do the reverse. I found Sebastian Krahmer's OpenSSH Reverse [1] which looks very promising, but it is a few revisions behind. I was wondering if anyone has considered

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

Hi Everyone, My first posting to the list. Longtime Ogg fan, have created many .oggs, but this is the first time needing help. I've been transferring CDs to Ogg files using crip. I've been in the process of setting up a bunch of mules to transfer my music collection to electronic format, but I've found an oddity: >bash$ oggenc -q 5 oddity.wav >Opening with wav

hi folks, in my (unlucky) first test of ogg vs other encoders, i found a case where wma and mp3pro sound much better than ogg at 64k. can anyone suggest a setting that i haven't tried yet that can rival the wma and mp3pro samples at 64k? it's the "gravel effect" that is troublesome. the part in question is the first 15 seconds of this wave file:

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Hello, I have a program that takes blog urls that our customers enter into their profiles. In some cases they enter bad feeds (IE html pages instead of atom or rss), and it throws these two errors: execution expired and bad URI errors in the view. I need to find a graceful way to handle both. Preferably, if I could find a way to throw an error message for the execution expired

Hi all, Have trouble encoding / playing back OGG using WinAmp + plugin. I encoded the files from .WAV with Ogg drop RC2. Seems to be ok. When I try and play them back though I get very very weird sound effects (some kind of "wobbling", some bass frequencies that are usually in the background seem to get more emphasised and they seem to have some kind of strange echo / reverb to it