similar to: how to play multiple speex audio streams simultaneously

Another approach that I have come across is to decode all speex audio streams and add their 16 bit signed PCM values clipping values above 2^15-1 and below -2^15. Is this a reliable appproach? On Thu, Oct 14, 2010 at 3:54 PM, Vishal Jain <cse.vishal at gmail.com> wrote: > Hello, > > In a way similar to speexclient voip example, I want to have a voip > application where 3 or

Hello, In a way similar to speexclient voip example, I want to have a voip application where 3 or more users can interact. This creates the following problems "1. How to playback 2 or more audio streams such that they appear overlayed (parallel)". The audio streams are speex encoded. I have few approaches in mind. 1. Create multiple decoder states and multiple ALSA audio playback

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hi So in your case if you don't want to use speex_echo_cancellation than you should do next way: Save the playback frames (clean echo) within separate playback thread with speex_echo_playback func. Then use speex_echo_capture - it will remove the clean echo frame (saved with speex_echo_playback func) from const spx_int16_t *rec buffer (saved with speex_echo_capture) and you will get

Well, I'm not a professional in AEC theory, but what I've mentioned is: speex_echo_state_init(20, 320*10) - frame size should correspond to 20ms. At your sampling rate (16000hz) is should be 16K*0.02 = 320. The same I can notice about echo tail. 100ms: 16000*0.3 = 4800, not 3200 as you has. But that's not crucial I think. Just wanna you get the point. "Internally,

Hi Anton Thanks for the info... What if I use this function do I still need to sync the playback and capture? Assuming that the captured voice (*rec) already contains the echo which I need to cancel out... -------------------------------- void speex_echo_capture (SpeexEchoState *st, const spx_int16_t *rec, spx_int16_t *out) Parameters: st Echo canceller state rec signal

Dear useR's I have a small basic problem which I am hoping to get some help with. I have a data frame, testSeq_df, with 1 row and 500 columns. Each column is a character (a,c,g or t). I want this sequence to have 4 factors (a,c,g,t). When I try the following: for(i in 1:500){ if (length(levels(testSeq_df[,i]))==1) levels(testSeq_df[,i]) <-

Anton A. Shpakovsky <saa <at> tomsksoft.com> writes: > > As for me - speex_echo_cancellation is a better choise. Try using it in > capture thread instead > of those speex_echo_capture and speex_echo_playback functions. > > And please, describe your problem in details. Cause the fact that you > "didn get echo cancellation" > doesn't mean you are

On Sat, Jan 22, 2005 at 07:20:46PM -0800, Jeff Simmons wrote: > I've been experimenting with running an internet audio stream to the remote > site. Our ISP has a direct route from the local studio to the remote > transmitter (over their backbone and then to the remote via high speed > wireless, about 5 hops). I've set up ice2 and icecast, and I'm getting a very >

Hello list members. I would like to know how to playback an audio file to the caller, and while it's played asterisk to continue executing the next priorities on extensions.conf That's not the case when using "playback" command, because the next priority is executed until the audio file ends playing. I want to evaluate some variables while caller hears the audio file. Any

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

On Fri, Aug 04, 2023 at 09:03:20PM +0000, Verma, Vishal L wrote: > On Fri, 2023-08-04 at 20:39 +0200, Pankaj Gupta wrote: > > Gentle ping! > > > > Dan, Vishal for suggestion/review on this patch and request for merging. > > +Cc Michael for awareness, as virtio-pmem device is currently broken. > > Looks good to me, > > Reviewed-by: Vishal Verma

On Fri, Aug 04, 2023 at 09:03:20PM +0000, Verma, Vishal L wrote: > On Fri, 2023-08-04 at 20:39 +0200, Pankaj Gupta wrote: > > Gentle ping! > > > > Dan, Vishal for suggestion/review on this patch and request for merging. > > +Cc Michael for awareness, as virtio-pmem device is currently broken. > > Looks good to me, > > Reviewed-by: Vishal Verma

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

As for me - speex_echo_cancellation is a better choise. Try using it in capture thread instead of those speex_echo_capture and speex_echo_playback functions. And please, describe your problem in details. Cause the fact that you "didn get echo cancellation" doesn't mean you are doing smth wrong. Regards, Anton A. Shpakovsky -----Original Message----- From: speex-dev-bounces at

Hello, I have dozens of video files that were encoded using the speex format of audio and can not seem to find any video playback devices that support speex lol. Literally every playback device I've encountered (mac, 10.10.2) plays the videos w no sound. Do you know of any playback devices for the mac format that support speex? -------------- next part -------------- An HTML attachment was

I found this information elsewhere.... making it /root/.asoundrc helped. # This is .asoundrc # # this makes legacy OSS apps use alsa software mixing dmix pcm.dsp0 { type plug slave.pcm "dmix" } # mixer0 can stay unchanged, because it isn't used anyway, I guess ? ;) ctl.mixer0 { type hw card 0 } # this makes native ALSA apps default to using dmix pcm.!default {

Whoops, left this message in my outbox. I managed to fix the problem. Apparently I was only copying 160 bytes (Frame Size) back into the audio stream when I should have been copying 320 (chars <-> shorts confused me there). Hence why I could hear myself yet it was distorted. Half the wav was missing =) To answer some of the other questions here, for any insight into what I'm doing:

Hi useR's, Has anyone used the "Lars" package in R before? If so, is there any tutorial (not manual) or worked out example online for this R package that one can go through to figure out how one can use this package with lasso regression? I appreciate any help I can get in this direction. Sincerely, Vishal -- *Vishal Thapar, Ph.D.* *Scientific informatics Analyst Cold Spring

hi all, We going to implement a music service which enable user to playback a song by dialing to a service number. The problem is that the amount of data is huge so we have to plae it on an different server which is connected to the asterisk's via internet. Does asterisk support playing a audio file from an resource locate in a remote host? Please help, Thanks, Quyps