[asterisk-users] voice detection during playback
Is it possible to detect voice while playing back a message? I am using AMD (Answering Machine Detect application) and it seems to work pretty well but some outgoing messages (on my Sprint cell phone for example) have silence in them. After the initial message of about 20 seconds it says press or say one. Then there is a pause of two or three seconds, followed by leave a message ofter the tone. That pause breaks AMD. If I could detect voice while playing a message I could stop playing the message, wait for silence and restart playing the message from the beginning. Regards, David Koski ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice detection
You can also use the AGI interface function RECORD FILE and specify a max record duration of 5s and silence detection of 1s. Time the duration of the call to asterisk - if its longer than 1 second you know you've got voice. If you need to check for voice over a longer period of time - repeat the call x times. This way you'll wait for approx 'x' seconds for voice or silence. I've done this using some C code and works very well as a grunt detector - timing out after 5 seconds of silence or returning immediately when voice is received. Paul - Original Message - From: Christian Hecimovic [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, October 04, 2003 10:39 AM Subject: Re: [Asterisk-Users] Voice detection dsp.c has silence detection that works quite well for detecting end-of-voice silence. It is used to allow only a certain amount of silence at the end of voicemails, for instance. See app_voicemail2.c on how to use it, specifically the function play_and_record(). Note that the silence threshold (how sensitive you are to silence) is read in from the voicemail.conf file. Since the silence detection stuff has a nice public API, you can use it for any app you write. See app_skel.c for a basic shell, and follow something like app_voicemail2.c. Read in the acceptable values for threshold and so forth from a configuration file (there is a nice Asterisk API for this, also), and you're set. For your purposes (playing a file after detecting silence in a remote voice stream), such an app should be quite simple. Christian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice detection
Steve, I don't have any real experience in DSP methodologies, although I have picked up on the high-level theories in my research. However, I am *very* strong-willed in the Where there's a will, there's a way category. :) Here's my current thought: Sphinx is an open source STT library that can work in real-time (specifically sphinx2). Could we not pipe the called party's audio into it and then look for a given period of time with no text output? I also found this site, http://www.cs.wpi.edu/~claypool/courses/525-S01/projects/proj1/ where a prof's got a project for students that fits perfectly with what I'd like to do. He mentions Rabiner and Sambur's algorithm (from 1975) for detecting isolated speech endpoints. Brad Steve Underwood wrote: Hi Brad, If you want to detect that a sound is voice, rather than something else, it isn't easy. There is information around on the Internet about methods, but I have never tried them and don't know how well they work. Unless you have some understanding of DSP I wouldn't bother trying. On the other hand, if you do have some DSP expertise it might be a fun thing to try. Regards, Steve Brad Waite wrote: Does anyone know if there's public voice detection algorithms available? I've scoured the net for the last hour or so, and I can't come up with anything except a few proprietary or embedded solutions. I know dsp.c uses goertzel algorithms for DTMF detection, but how does one detect voice? I dunno, maybe detecting voice isn't the way to go. I want to begin playback of a file after a phone/answering machine has answered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice detection
dsp.c has silence detection that works quite well for detecting end-of-voice silence. It is used to allow only a certain amount of silence at the end of voicemails, for instance. See app_voicemail2.c on how to use it, specifically the function play_and_record(). Note that the silence threshold (how sensitive you are to silence) is read in from the voicemail.conf file. Since the silence detection stuff has a nice public API, you can use it for any app you write. See app_skel.c for a basic shell, and follow something like app_voicemail2.c. Read in the acceptable values for threshold and so forth from a configuration file (there is a nice Asterisk API for this, also), and you're set. For your purposes (playing a file after detecting silence in a remote voice stream), such an app should be quite simple. Christian On Friday 03 October 2003 10:25, Brad Waite wrote: Steve, I don't have any real experience in DSP methodologies, although I have picked up on the high-level theories in my research. However, I am *very* strong-willed in the Where there's a will, there's a way category. :) Here's my current thought: Sphinx is an open source STT library that can work in real-time (specifically sphinx2). Could we not pipe the called party's audio into it and then look for a given period of time with no text output? I also found this site, http://www.cs.wpi.edu/~claypool/courses/525-S01/projects/proj1/ where a prof's got a project for students that fits perfectly with what I'd like to do. He mentions Rabiner and Sambur's algorithm (from 1975) for detecting isolated speech endpoints. Brad Steve Underwood wrote: Hi Brad, If you want to detect that a sound is voice, rather than something else, it isn't easy. There is information around on the Internet about methods, but I have never tried them and don't know how well they work. Unless you have some understanding of DSP I wouldn't bother trying. On the other hand, if you do have some DSP expertise it might be a fun thing to try. Regards, Steve Brad Waite wrote: Does anyone know if there's public voice detection algorithms available? I've scoured the net for the last hour or so, and I can't come up with anything except a few proprietary or embedded solutions. I know dsp.c uses goertzel algorithms for DTMF detection, but how does one detect voice? I dunno, maybe detecting voice isn't the way to go. I want to begin playback of a file after a phone/answering machine has answered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice detection
Does anyone know if there's public voice detection algorithms available? I've scoured the net for the last hour or so, and I can't come up with anything except a few proprietary or embedded solutions. I know dsp.c uses goertzel algorithms for DTMF detection, but how does one detect voice? I dunno, maybe detecting voice isn't the way to go. I want to begin playback of a file after a phone/answering machine has answered. Suggestions? Brad Waite ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
