similar to: Suitability of speex for use with noisy, non-voice source material?

Hello Jean-Marc, I have completed some very basic testing with unexpectedly excellent results. I am posting this to the reflector to encourage others into similar experiments. My experiment consisted of processing a 200-second long sample taken from a ham radio shortwave receiver, with a variety of signals (some strong, some weak - that is, weaker signals have more noise, and the noise is

I am totally new to icecast, and would appreciate a pointer in the right direction. My application is to encode audio (at the linux server) at a low bit rate, say ~16 kbps, for transmission to a decoder at another location. The audio source will be bandlimited (basically voice grade circuit such as you'd find in a telephone application.) The decoder end (connected via a TCP/IP session)

Hello Thomas, You are correct - the application is basically for secure remote access to ham CW (morse code) station. I believe the solution will be based on low bit rate MP3 streaming for receiver side (MP3 allows use with existing CODEC in client side, which could include Win CE or Linux palmtop type form factor, and the audio source would just be "voice grade" material on a mono,

A while ago someone asked about a low-pass filter for oggenc and was told to get AFsp and filter outside of Oggenc. Well, I got it, and am totally lost (It's way more complicated than SOX) so now can anyone briefly describe what type of filter I should set up (FIR, IIR, all-pole), why one is better than the other, and if you have filter coefficient files lying around (lowpass, 19 or 20 kHz

Doing a little research before Friday's Voip Users Conference call with Dan Behringer. Are any of the newer Polycom wideband codecs implemented in v1.6? Specifically, G.722.1 or G.722.2? Thanks, Michael Graves mgraves <at> mstvp.com o(713) 861-4005 c(713) 201-1262 sip:mjgraves at mstvp.onsip.com skype mjgraves

Hi Folks, I'm trying to use my mobile as a trunk via bluetooth - calls done in a softphone go thru GSM network and calls destinated to my mobile are answered at the softphone. I have asterisk configured to do so but I'm facing an issue - Audio is audible but it?s not intelligible. I feel like the audio is breaking. Below is the asterisk log. I also get lots of ?hci_scodata_packet: hci0

Has anyone ever thought configuring asterisk on a pair of pc's to act as remote broadcast terminals for the broadcast radio industry? Seems like a stripped down asterisk on a laptop with a PCMCIA ISDN modem connecting to another asterisk instance on a PC at a radio station would work nicely. Use one of the higher quality codecs, interface the remote mixer to the sound card on the

I’m using a Windows development component which uses vorbis.dll, ogg.dll, vorbisenc.dll for encoding an Ogg Vorbis file. It's all working well except for one user occasionally has a 1 hour file appear as 2 hours and it plays at half speed. It is being converted from stereo to mono before feeding the encoder with a channels=1 configuration. Here is an example file which will be available for

The following creates a point process version of a sinewave (maybe there's a better way): p<-amp*cos(2*pi*freq*(1:n)/n ) + 0.5 as.numeric(runif(1:n)<p) I get something like this: 1 o oo oo o o ooooo o 0 ooo o o o o o ooo ooo In case it's not obvious, this is a noisy version of 1 ooooooo ooooooo 0 ooooooo

Hello, When I plot y=f(x) from the file xy.txt ( http://www.nabble.com/file/p18773387/xy.txt xy.txt ), I can clearly see a trend. Is there a function or a package able to take the median value of y for an interval of x (x +/- a defined value) to plot nice graph (at least a better one) ? Thanks in advance, Have a nice week-end, Ptit Bleu. -- View this message in context:

Hi, I have a set of observations (x,y), derived from a previous estimation. For each observation I also have an estimated variance s(y) derived from the first stage. The problem is that I need to smooth the data (x,y) while taking into account the fact that the y's have been estimated at a previous stage and thus already come with a variance. So, if I smooth the data I somehow need to

https://bugzilla.mindrot.org/show_bug.cgi?id=1431 Summary: ssh-copy-id -i .ssh/*.pub hostname is noisy Classification: Unclassified Product: Portable OpenSSH Version: 4.7p1 Platform: Other OS/Version: Linux Status: NEW Severity: minor Priority: P2 Component: Miscellaneous AssignedTo: bitbucket

When I call in from an outside POTS line to a Zap channel, and the call ends, it seems like the hangups are very "sloppy." I see Asterisk give the hangup command, but on my phone there's lots of clicks and the line acts like it's staying open for several seconds, then I hear a phone ringing sound followed by "If you'd like to make a call, please hang up and try

When a Snom 200 (v2.1l) calls a C7960 (v4.4), both using g711u as default, the conversation is extremely noisy from the Snom to the Cisco, but clear in the reverse direction. Using a sniffer, I see packets from the Snom to the Cisco of 87 bytes and Cisco to Snom of 214 bytes. Asterisk is CVS from Saturday. The communications between the two was working fine on Saturday, however something has

Dear all, Currently I am using asterisk 1.4.23.1. . Over the period of 1 week, the memory in use starts off at 50% and continues to climb until it hits 99%. When memory usage ratio become 50% or more, the quality of calls become extremely noisy. The call quality goes back to being perfect once I reboot the machine, but I was to try and avoid having to reboot the machine every week. the following

On 11/22/2017 05:07 PM, Martin Peres wrote: > Hey, > > Thanks for your answer, Andy! > > On 22/11/17 04:06, Ilia Mirkin wrote: >> On Tue, Nov 21, 2017 at 8:29 PM, Andy Ritger <aritger at nvidia.com> wrote: >> Martin's question was very long, but it boils down to this: >> >> How do we compute the correct values to write into the e114/e118 pwm

For those of you using Dovecot on Fedora, the noisy syslog is fixed: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=124979 I'd suggest getting the pam SRPM from Rawhide (Fedora development tree) and rebuilding, to minimize dependency issues. (I haven't tried this yet. I'm just forwarding the bug-closing report.)

Hi Bernhard, Can you confirm I'm understanding everything correctly? You encode with the same encoder and then decode with either A) blackfin assembly and fixed-point or B) fixed-point only on Blackfin. Then A) sounds bad and B) sounds good. If you do the same in narrowband, it sounds OK. Is that correct? If that's the case, it's *probably* some kind of bug and/or invalid assumption

On 11/23/2017 02:48 PM, Martin Peres wrote: > On 23/11/17 10:06, John Hubbard wrote: >> On 11/22/2017 05:07 PM, Martin Peres wrote: >>> Hey, >>> >>> Thanks for your answer, Andy! >>> >>> On 22/11/17 04:06, Ilia Mirkin wrote: >>>> On Tue, Nov 21, 2017 at 8:29 PM, Andy Ritger <aritger at nvidia.com> wrote: >>>>

Hi, I have data of the form: t x y trip t1 x1+e y1+e A t2 x2+e y2+e A t3 x3+e y3+e B t4 x4+e y4+e B t5 x5+e y5+e B ... ... ... ... where t is time and x/y are positions in space and e is a random error term. Trips A/B/C/etc are entirely independent