search for: hearable

I'm going to encode ~2000 CDs soon. All genres, but 90% of it has distorted guitars... Everything from punkrock to metal to industrial to goth to synthpop to classical to techno to whatever... I've heard that RC2 has some hearable artifacts, even in 192/256 kbps... There have been quite a few "bugreports" since RC2 with people sending samples that even I can differ from the original wavs. That makes me doubt if it is worth doing it now. Also I don't want to wait too long. Is RC3 anywhere near a release? Is...

...to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to "nirwana" also no indication is hearable, calling a second internal subcribes bridges them to the first. The PBX sends a SETUP message with channel identification 30 and Asterisk bridges them to Zap-30, instead of Zap-31. The configuration - Digium TE410p card, set for E1 ________in zaptel.conf________ span=1,1,1,ccs,hdb3,crc4 bchan...

...very experienced in informatics, cause i'm only an audio engineer and sorry for my broken english but i have an idea... After the filterbank the maximum aplitude in each band decides about the nessesary bits, then the psychoacoustic model decides how much of the signal can be truncated without hearable noise. What do you think about quantising the remaining bits nonlinear with a logarithmic scale? This would share 25%. Nonlinear quantisation is used for instance in a-law or for the longplay-mode of DAT-recorders. The disadvantages of nonlinear quantisation will not apply here, because the 576...

...using arbitrary sampling rate conversion. I wonder if it can provide enough performance. Because I have also designed a sampling rate converter. After tested the offset accurately, it can reduce the offset to less than 0.1Hz, then the signal after resampling is send to speex AEC. But there is still hearable echo even if it is far less than that can be heared before resampling. Does anybody have any suggestion about practical acoustic echo cancellation in low-cost soundcards? You know, most low-cost soundcards have the problem of sampling rate asynchronous. -------------- next part -------------- An HT...

...oblems registering SIP softphones, like the most excellent xTen X-Lite, try commenting out 'bindaddr' in sip.conf. That allowed me to register and get everything working. Finally, if anyone has any ideas about how to improve IAX voice quality, I'd be happy to hear them. Everything is hearable, but there are an awfull lot of clicks and pops in the background. The machine is a Celeron 500 with 128mb of RAM and Gentoo 1.4 (w/latest gentoo updates) and Asterisk 0.7.2. I'm the only one using the machine ATM and it's about 2 ft from my desk with a dedicated hub... There is no telep...

...softphones showed "REGISTERED" and can receive incoming call,but failed to make outgoing call.always showing "call failed 407,authentication reqired".In the caes ht100 made outgoing call to one softphone,ht100 received cristal clear sound,but softphone received sound bearly hearable. my pc which runing ASTERISK is athlon 1G 256g ram I am wondering if this thing really work ,anyone can give suggestion? _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url....

...command "system dahdi start/restart". When I make an outbound call over these analog lines, I'm told in the CLI " WARNING[22601]: chan_dahdi.c:2685 dahdi_enable_ec: Unable to enable echo cancellation on channel 5 (No such device)", and there is noticeable echo on the line, hearable by the caller inside the office, not by the recipient outside the office. When I make an inbound call, I get the same message, with echo still heard by the person inside the office, and not outside. I've run fxotune -i 4 -e 5 also, and still receive the same output. Below is my chan_dahdi.co...

...p-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the "cannot be completed" message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and...

...ps, voice with instruments in background, resulting ogg was 118kbps The sound was horrible. There were strong distortions in the voice of the singer which were extremely annoying. Given these results, I would conclude transcoding will give very annoying artifacts on many files which will even be hearable on bad equipment. The effects are not as bad but still easily hearable on files which were encoded with higher quality. But as we are considering users which are more concerned about diskspace than quality those will be a lot less present than lower quality encodings, which will sound simply horri...

Hi! I have a problem. I hope, you can help me. I use a Delphi conversion (from Aleksandr Shamray), but it doesn't work when I'd like to convert a *.RAW into a mono *.ogg file. vorbis_encode_init_vbr(vi, 1, 44100, 0.5); //because of the mono the program stops at line: //* uninterleave samples */ . . buffer[1][i] := smallInt((pArray(@readbuffer)[i shl 2 + 3] shl 8) or

Hi: Saw this in Edupage today. Grrrrrr. Geoff. From: EDUCAUSE <educause@EDUCAUSE.EDU> COPY-PROTECTED CDS QUIETLY SLIP INTO STORES Macrovision, in coordination with several major recording labels, has for several months been piloting new technology to prevent music consumers from copying CDs onto their PCs. The technology distorts CD recordings with a series of audible pops and clicks

.... I wonder if it can provide >> enough performance. Because I have also designed a sampling rate >> converter. After tested the offset accurately, it can reduce the >> offset to less than 0.1Hz, then the signal after resampling is send to >> speex AEC. But there is still hearable echo even if it is far less >> than that can be heared before resampling. >> >> Does anybody have any suggestion about practical acoustic echo >> cancellation in low-cost soundcards? You know, most low-cost >> soundcards have the problem of sampling rate asynchron...

...ap channels. But actually it is already the voice volume is low and I was looking to increase the gain (currently it is 0.0), so I do not know if eric was mean to reduce it less than 0.0, but I can not do that due to the low volume that is already existed, so any more reduce will make the voice not hearable well, even if the DTMF problem resolved. I can share u one thing, the main problem in the Background is the duplication in the first digit detection, so for example if I entered 150, it will detect it 115 (and will not continue to detect the 0 as the digit length completed). Any advise? Regards...

I was pretty shure that W3 works great in wine, but in recent version 1.1.4 there is no image in cinematics but only in full screen - itrs weird, because when turn into windowed mode cinematics play well - in opposite when in fullscreen mode, only sound is hearable. Whats up with that??

...rate conversion. I wonder if it can provide > enough performance. Because I have also designed a sampling rate > converter. After tested the offset accurately, it can reduce the > offset to less than 0.1Hz, then the signal after resampling is send to > speex AEC. But there is still hearable echo even if it is far less > than that can be heared before resampling. > > Does anybody have any suggestion about practical acoustic echo > cancellation in low-cost soundcards? You know, most low-cost > soundcards have the problem of sampling rate asynchronous. > That one sou...

...sion. I wonder if it can provide >> enough performance. Because I have also designed a sampling rate >> converter. After tested the offset accurately, it can reduce the >> offset to less than 0.1Hz, then the signal after resampling is send to >> speex AEC. But there is still hearable echo even if it is far less >> than that can be heared before resampling. >> >> Does anybody have any suggestion about practical acoustic echo >> cancellation in low-cost soundcards? You know, most low-cost >> soundcards have the problem of sampling rate asynchronous....

Hi i have seen that the dbPowerAmp ripping and encoding software supports a new so-called "FLAC uncompressed" format, e.g. http://www.audiostream.com/content/dbpoweramps-flac-lossless-uncompressed-wish-come-true i know only the normal flac compression levels from 0 to 8. have i missed an option on the flac comamnd line tool or how could i achieve that on the linux command line flac

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Hi guys. IMO now it's definitely time for finalizing Vorbis and throwing it in the arena. Two reasons: - RC2 sounds wonderfully well, I'm really amazed how good it is.. and I personally think there's little room for further improvement in sound quality.. - portable music devices are starting to take off (see Waitec, Creative, Apple and many other vendors). Wouldn't be good

...SIP server to sip01.sipphone.com of course) - when the SIP server is Asterisk, can be dialed from ISDN without any problem (maybe a slight delay), quality is good both directions. - can dial to Asterisk, in that case Asterisk's debug shows the call, but fails. Nothing is hearable on the BudgetTone except a busy tone. Software: Program--1.0.3.81 Bootloader--1.0.0.7 HTML--1.0.0.18 Call examples: (this time with `sip debug' I just found about) SIP phone dials '2' Sip read: INVITE sip:2@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.190 From: &quo...