Displaying 20 results from an estimated 43 matches for "3db".
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2005 Jun 16
1
Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
...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&...
2007 May 03
3
iaxclient & speex
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
2005 Jul 26
2
7960 SIP Firmware Upgrade Strange Problem
...-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...
2012 Nov 13
0
[LLVMdev] Errata for "LLVM Assembly Language" docs (llvm.org/docs/LangRef.html)
...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...
2009 Mar 03
0
speex_jitter_buffer and DTX (and multiple streams)
...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...
2003 Dec 01
0
No subject
..., 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...
2007 May 03
0
Re: [Iaxclient-devel] iaxclient & speex
...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
2007 May 03
2
Re: [Iaxclient-devel] iaxclient & speex
...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...
2017 Apr 18
1
Antw: Re: 133 kbps stereo killer sample
...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...
2007 May 03
0
Re: [Iaxclient-devel] iaxclient & speex
...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...
2007 Apr 14
13
Ambisonics in Ogg Vorbis
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
2007 Apr 14
13
Ambisonics in Ogg Vorbis
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
2004 Dec 16
0
FW: Cisco 7960 (SIP) hold problems
...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...
2016 Aug 02
2
fallen too far behind, removing
...-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...
2005 Sep 29
1
Audio Files, Filtering, and Formats for Asterisk
...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...
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
...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...
2005 Mar 26
4
Cisco's description of echo
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
2004 Oct 04
1
Cisco 7960G w/ SIP not working properly
...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
#######...
2004 Jan 20
0
nlminb function
...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]]
2003 Oct 17
0
zaptel: [rx|tx]gain on E1/PRI/isdn audio quality problems
...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...