search for: 3db

...2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk&...

Hi The latest SVN trunk for speex has changed the SpeexPreprocessState to an opaque structure, for jolly good software engineering reasons. However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) relies on some members of this. It uses speech_prob to detect when there is enough speech to consider AAGC and then loudness2 to decide how to adjust the input mixer. We want to use

...-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: "1" # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: "avt" # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: "3" # SIP Timers timer_t1: "500" ; Default 500 msec timer_t2: "4000" ; Default 4 sec sip_retx: "10" ; Default 11 sip_invite_retx: "6&qu...

...re xx is the ASCII code for the character > in hexadecimal. In this way, any character can be used in a name value, even quotes themselves. I would add an example to clarify where the quotation marks go, since it's non-obvious. Like so: > For example, the names %"a=b", %"a\3db", and %"a\3Db" all identify the same variable. (2) LangRef.html writes: > This last way of multiplying %X by 8 illustrates several important lexical features of LLVM: > Comments are delimited with a ';' and go until the end of line. > Unnamed temporaries are...

...he speex_jitter_get(..) unless it had 1 or more packets, but would this ruin the ticking? Another reason for using this method is that I need to detect which client is saying anything as I want to merge the streams (after decode ofc). I have not exactly found out how yet, but Im considering -3db gain per stream (any hints would be appreciated =) ). If I were to use the speex_jitter_get(...) it would require checking the entire frame for 0/silence to determine if it should be merged (since there is no feedback, and I do not want to add silent streams because of the -3db gain) and...

..., have I gone wrong somewhere else? I include here my smb.conf: # [start smb.conf] [global] encrypt passwords =3D yes security =3D user netbios name =3D smbserver comment =3D Red Hat Samba Server workgroup =3D smbgroup ldap admin dn =3D "cn=3DManager,dc=3Da,dc=3Db" ldap suffix =3D "dc=3Da,dc=3Db" ldap ssl =3D off =20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20= =20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20=20 logon drive =3D U: logon path =3D \\%N\profiles\%g domain master =3D yes...

...use AAGC, and then register some callbacks that speex_preprocess() could call to query or set the input or mixer level. Further, a more intellegent implementation within speex could consider the requested changes in the rest of the preprocessor chain (i.e. it would know that if it asked for a 3dB increase in input gain, to expect that input levels would rise by 3dB within a few frames). The hacky implementation I did inside of iaxclient gave speex no such information. -SteveK

...hen register some callbacks > that speex_preprocess() could call to query or set the input or mixer > level. Further, a more intellegent implementation within speex could > consider the requested changes in the rest of the preprocessor chain > (i.e. it would know that if it asked for a 3dB increase in input gain, > to expect that input levels would rise by 3dB within a few frames). The > hacky implementation I did inside of iaxclient gave speex no such > information. This is probably things we'll want to consider one we decide on where to put the AAGC in the first plac...

...efore encoding with > `sox -v 0.5 floex.wav quiet.wav` and now I can't ABX it succesfully anymore. > So the artifact I heard was just clipping when encoding or decoding. Hi! Somewhere I read the recommendation that the peak level of input material shopuld not be 0dB, but something like -3dB. The reasoning was that encoding reconstructs the audio signal from the sampling points, and the reconsruction could actually exceed the sampling point, which my result in an overdrive, which will result in clipping. I just wonder whether the encoder (or the decoder at least) can or should warn in...

...e callbacks >> that speex_preprocess() could call to query or set the input or mixer >> level. Further, a more intellegent implementation within speex could >> consider the requested changes in the rest of the preprocessor chain >> (i.e. it would know that if it asked for a 3dB increase in input gain, >> to expect that input levels would rise by 3dB within a few frames). The >> hacky implementation I did inside of iaxclient gave speex no such >> information. >> > > This is probably things we'll want to consider one we decide on wher...

On 2/28/07, Ivo Emanuel Gon?alves <justivo@gmail.com> wrote: > On 2/28/07, Ralph Giles <giles@xiph.org> wrote: > > Well, there are todo pages at wiki.xiph.org, but I meant more in the > > community folklore sense. My point is a roadmap doesn't help much unless > > there are people committed to making things happen. That's been the > > problem with a

On 2/28/07, Ivo Emanuel Gon?alves <justivo@gmail.com> wrote: > On 2/28/07, Ralph Giles <giles@xiph.org> wrote: > > Well, there are todo pages at wiki.xiph.org, but I meant more in the > > community folklore sense. My point is a roadmap doesn't help much unless > > there are people committed to making things happen. That's been the > > problem with a

...0-Disabled (default), 1-Enabled, 2-Privileged # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: "1" # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: "avt" # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: "3" # SIP Timers timer_t1: "500" ; Default 500 msec timer_t2: "4000" ; Default 4 sec sip_retx: "10" ; Default 11 sip_invite_retx: "6&qu...

...-c "FFREPORT=file=/home/vlc/`date +%Y-%m-%d-%H-%M-%S`-mp3.log screen -A -m -d -S ffMP3 /home/vlc/ffmpeg/latest/ffmpeg -f s16le -ar 44100 -ac 2 -timeout 2000000 -i http://localhost:8500/source -reconnect_at_eof -reconnect_streamed -b:a 192k -compression_level 0 -ac 2 -ar 44100 -af "volume=3dB" -f mp3 -ice_name 'LiveFM @ 192' icecast://source:pass at localhost :8500/LiveFM_320" icecast.xml <icecast> <limits> <clients>1000</clients> <sources>3</sources> <threadpool>5</threadpool> <qu...

...file. Then I check the audio again, and normalize (like compression but only raises the whole file to where the highest peak of audio reaches the level requested, instead of raising or lowering the level on a dynamic basis by using readahead of a couple milliseconds). I usually normalize to around -3db. The end result is a WAV file that sounds good over the phone. I then put the file on the asterisk server (or another server with sox installed) and convert to gsm, ulaw, and alaw (using the original WAV, not using converted gsm or whatever). Hope this was helpful, and I wish you luck. If nothin...

...711ulaw # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec # Dialplan template (.xml format file relative to...

We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes

...dec: none # TOS bits in media stream [0-5] (Default - 5) tos_media: 5 # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 # SIP Timers timer_t1: 500 ; Default 500 msec timer_t2: 4000 ; Default 4 sec sip_retx: 10 ; Default 10 sip_invite_retx: 6 ; Default 6 timer_invite_expires: 180 ; Default 180 sec #######...

...9;ve got very little programming knowledge so a layman's description would be good. Many thanks, Helena *************************************************** Helena Rodnight Institute of Geography and Earth Sciences University of Wales, Aberystwyth Llandinam Building Aberystwyth Ceredigion SY23 3DB Tel: +44 (0)1970 622604 [[alternative HTML version deleted]]

...SIP/zaptel network ---- pri --- ASTERISK GW --- iax --- ASTERISK PBX --- PHONES w/ any codec the rx (public network to local phone) audio channel is audibly too loud and peaks are clipped as if something in the path is increasing the level of about 3dB on both SIP and zaptel phones. Prompts on ASTERISK GW are unaffected, only audio coming from the PRI is distorted. Also, ASTERISK PBX has a chan_capi device, which has good audio as well. Setting 'rxgain=-3.0' on the ASTERISK GW fixes the problem[1] but it looks to me there must be some...