search for: 0db

...clock timezone GMT 0 dspint DSPfarm1/0 ! ip subnet-zero no ip routing ! ! ip name-server XXX.XXX.XXX.XXX ! no ip cef isdn switch-type primary-ni call rsvp-sync voice call send-alert voice rtp send-recv ! voice service voip ! ! ! ! ! ! controller T1 1/0 framing esf linecode b8zs cablelength long 0db pri-group timeslots 1-24 ! controller T1 1/1 framing esf linecode b8zs cablelength long 0db pri-group timeslots 1-24 ! controller T1 2/0 framing esf service-type ccs-voice linecode b8zs cablelength long 0db pri-group timeslots 1-24 ! controller T1 2/1 framing esf service-type ccs-voice...

...the poor reliability during near end talk is what we >> need to find a solution to now. > > OK, I'll have to think about it. At this point I'm not sure what to do > here. If you find anything, I'm definitely interested. The one idea I have is to have a default gain2 of 0dB and only when the residual echo goes high we reduce the gain2. This could be implemented similar to eq. (3) and (4) in this paper: http://research.microsoft.com/pubs/69504/diegobenderskyhdres.pdf The idea would be to use your estimation of the residual echo and noise But to modify the calculatio...

...ately. It says: Adit 600> show a:1 SLOT A: Settings for DS1 1: Circuit ID: CAC DS1# A:1 Up/Down: UP Framing: ESF Line Coding: B8ZS Line Build Out: DSX-1 EQUALIZATION FOR 0-133 ft. (CSU 0dB) Loop Code Detection: ON Loopback: OFF FDL Type: None Can anyone familiar with the Adit 600 and/or TE405P see any obvious errors here? -Darren -- Darren Nickerson Senior Sales & Support Engineer iFAX Solutions, Inc. www.ifax.com darren....

...lems when I set the format to constrain the bitrate so it doesn't go any higher than 128KBPS, for example. This problem only occurs with stereo files, not mono. Let's say I set the format to use bitrates up to 45KBPS. Well, the low-pass filter has this problem, the closer the volume gets to 0DB, the lower the cutoff, by 0DB I think it's around 4KHZ cutoff but when the volume gets quiet it sounds like the cutoff is where it should be. The higher I let the bitrate go, the less the lowpass filter acts up, by 80KBPS I can't hear anything wrong with it. On the "dare to compare&quo...

...d to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15 is-slave = off DS0 Provisioning: analog-begin = 1 analog-end = 16 data-begin = 17 data-end = 24 alignment = same (The following is what our Vina T1 Integrator currently has in its settings. Our Linux box...

...termination service which routes the call to a PSTN number. Everything sounds great on the SIP phone, but the sound on the other end of the line is distant and missing bass, most especially so on the SPA-941 (which is the phone we really want to use). If I use the default handset mic gain value of 0db, the sound is so loud for the other person they have to hold the phone away from their ear. If I set it to -6db, it is still too quiet. The Aastra 9113i sounds a little better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm pretty sure my network setup is capable of transmitt...

...c/wanpipe/wanpipe7.conf: [wanpipe7] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 9 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 3 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 7 TDMV_DCHAN = 0 TDMV_HW_DTMF = YES [w7g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES Here is /etc/dahdi/system.conf loadzone=us defaultzone=us #Sangoma A104...

...st wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo cancelleris not really actively cancelling the echo in such condition? But without the echo sppression/NLP, echo is still noticable. I used the same onboard sound card for the speaker and microphone. As far as I can tell, the delay between echo_frame and input_frame is consiste...

...AN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 4 PCIBUS = 1 FE_MEDIA = T1 FE_LCODE = B8ZS FE_FRAME = ESF FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_HIGHIMPEDANCE = NO LBO = 0DB FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 0 [w1g1] ACTIVE_CH = ALL TDMV_ECHO_OFF = NO TDMV_HWEC = YES wanpipe2.conf: [devices] wanpipe2 = WAN_AFT_TE1, Comment [interfaces] w2g1 = wanpipe2, , TDM_VOICE, Comment [wanpipe2] CARD_TY...

...he volume of the sample before 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 lea...

...retreive the voicemail. My location creates another xx db of loss between myself and asterisk, and voicemail can hardly be heard. Actual Measured Values: 1. Asterisk is 5.6 db from the central office. Called from one pstn line, through the central office, to asterisk and sending a 1004 hz tone at 0db. Recorded the tone into voicemail. (Tone should have been recorded at about 11.2db, two times the cable loss) 2. Called into asterisk again, this time to retreive the voicemail and measured the 1004 hz tone from voicemail. It was -36db "actual". This retreival added another 11.2db of los...

...st wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo cancelleris not really actively cancelling the echo in such condition? But without the echo sppression/NLP, echo is still noticable. I used the same onboard sound card for the speaker and microphone. As far as I can tell, the delay between echo_frame and input_frame is consiste...

...wtih my recording file offline through the testecho. I got the same result(no prepocessor echo suppression) as my recording file for that live call. Basically I don't see the difference between input_frame and output_frame. As I mentioned in my original post, the echo return loss is about 20db. I guess the echo canceller is not really actively cancelling the echo in such condition? But without the echo sppression/NLP, echo is still noticable. I used the same onboard sound card for the speaker and microphone. As far as I can tell, the delay between echo_frame and input_frame is consi...

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

...I have this thought, and then I forget about it. I just got done on a call that was really bad, so.... Is there anyway (with SIP only - no ZAP members) to do auto gain control on conference calls? Even if I have to buy a module or something. I just need all call members mixed to approximately 0dB so people aren't having to constantly adjust their volume control on the speaker phones. Does such a beast exist? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150217/1a910994/attachment.html&g...

...uot;); to fprintf(stderr, "Usage: ztmonitor <channel num> [-v | -f FILE]\n"); Now run ztmonitor in verbose mode /ztmonitor 1 -vv Note: verbose mode is 2 v's not a "w" Now you will see the RX and TX numbers on the right hand side. If you are doing the 1004 Hz 0db test to set your levels. Try get a value around 14500 which means the RX graphical display is at maximum NOT midway! I could not find these details anywhere so that is why I am posting this here. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists...

...opusinfo is strange: $ opusinfo demo01.opus Processing file "demo01.opus"... New logical stream (#1, serial: 0000456a): type opus Encoded with libopus 1.0.1-rc3 User comments section follows... ENCODER=opusenc from opus-tools 0.1.5 Opus stream 1: Pre-skip: 356 Playback gain: 0dB Channels: 2 Original sample rate: 44100Hz Packet duration: 20.0ms (max), 20.0ms (avg), 20.0ms (min) Page duration: 1000.0ms (max), 980.0ms (avg), 20.0ms (min) Total data length: 579914 bytes (overhead: 0.758%) Playback length: 0m:48.000s Average bitrate: 96.65...

...b than ZIP! The second problem would be that no other tool would read them correctly anyway. The second idea is to truncate & use it as a lossy encoder, which can be "audibly lossless" anyway. But what's a common practice here? I would be tempted to leave 1 bit of headroom above 0dB, or maybe 2 bits. Normalizing before encoding could be an option, but the gain would then have to be inserted in a tag. & finally, what bit depth? 24? 25, 26? 32? I know there's no right answer, just asking if there are common practices. Thanks. -------------- next part --------------...

...: medium Component: Driver/nouveau Assignee: nouveau at lists.freedesktop.org Reporter: wolfgang.kde at rohdewald.de QA Contact: xorg-team at lists.x.org Sound goes out as Surround 5.1 over the HDMI cable on an Nvidia GT 630. Volume amplification is at minimum 0dB - if I increase it, the screen freezes. lspci: 01:00.0 VGA compatible controller: NVIDIA Corporation GK208 [GeForce GT 630 Rev. 2] (rev a1) 01:00.1 Audio device: NVIDIA Corporation GK208 HDMI/DP Audio Controller (rev a1) Kubuntu 16.04.1 with self compiled standard kernel 4.8.6 kodi version 15.2...

Hi all, I routinesly rips my CDs to WAV and then convert to ogg vorbis format for use in my car and portable player. I don't usually notice anything amiss, but on the last track of Mike Oldfield's "Music of the Spheres" album ("Musica Universalis", at the very end crescendo), the converted .ogg file exhibits terrible distortion (sounds like digital clipping). This