search for: inaudibly

Hello, I saw some conversation about this in the archives, but nothing definitive. If a call comes in over a CO line via the TDM400P, the Comedian Mail recording volume is so low it's inaudible. Calls coming in via SIP or IAX do not have this problem. Does anyone have any information on this issue? Thanks, Adam The contents of this email message and any attachments are confidential and

I quote from "Principles of Digital Audio" by Ken C. Pohlmann: "Because perceptual coders tailor the coded signal to the ear's acuity, they similarly tailor the required response of the playback system itself. Live music does not pass through amplifiers and loudspeakers, it goes directly to the ear. But recorded music must pass through the playback signal chain. Much of the

Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some calls the volume suddenly drops to inaudible o completely fades away for a time. If you hold on long enough (20 to 30 seconds) the sound will come back. My asterisk server is on a public IP, and basically acts as a VoIP bridge receiving calls from my customers (all of whom use Grandstream GXW400X

The results are at www.rational.co.za/speex.csv Each of the 11 quality settings is tested 3 times (narrow, wide and ultra wide band). Strangely narrow band quality 11 outperforms all wide band tests, but it can be due to my subsampling or some other inaudible effect like delaying. You can import it into Excel and sort it by SNR or other value. Divide the bits by 24 to get the bps. The

I'm attempting to use speex preprocess for automatic gain control in an application I'm working on and could use some help. I'm using Opus as my codec. In order to keep the number of packets down, I'm using 60msec frames. I'm sampling at 48KHz as is recommended for Opus. So, the frame length is 2880 samples and the sampling rate is 48000. The source of the data is a

Hello, I'm not a programmer or Opus developer but I tried to test the sound quality of a music file (Nick Warren - Devil's Elbow, freely available from the author's Soundcloud account) encoded with libopus 1.2 RC1. I used Windows binaries from free-codecs.com. I noticed that in the case of my selected music file (which is generally harsh on lossy codecs as it's necessary to

Jean-Mark sarkasm. Jean-Markasm. (Bonus points for providing an actual noisy WAV! ^_^) On 30/10/2017 20:28, Jean-Marc Valin wrote: Hi, Before I comment on the graphics you posted to visualize the difference between two audio signals, I'd like to ask for your help in evaluating my JPEG encoder. I've encoded an image with JPEG and then computed the difference with the original. I then

? Do you have any references for me to investigate, I am trying to understand how noise is reduced by introducing higher sampling rates. (I tried to search, but maybe it is so obvious that nobody even explains it) This is not very obvious. It requires you to understand basic signal processing theory. I will give some pointers below. Any physical signal (e.g. audio coming out of speaker, current

Hi. I've been browsing the archive on this topic and only found a few notes, all dating from a year ago (almost too precisely :) ) -- hope I haven't skipped the mails on that matter, sorry if I did. <p>Basically, I will get in the next few months a MOTU 896, that have 8 i/os in 24/96. I do pro sound recording, so it's more or less my business to have such a piece. Of

Hi Opus list, Please forgive me if this has been asked before. I find that Opus encoder created in mode OPUS_APPLICATION_AUDIO (as opposed to _VOIP) is using a lot of bits to encode silent periods of speech. This is relevant to a voip application for which good quality music is desirable, and in which I add a minimal comfort noise (order of few bits loud, e.g. MLS signal of amplitude 1 or 2)

OK, so at the link : https://uloz.to/!yyVrCY2Y8sn1/devil-s-elbow-opus-7z (https://uloz.to/%21yyVrCY2Y8sn1/devil-s-elbow-opus-7z) (change the language to English by clicking at the flag at the right upper side of the web page or just simply click at "Stáhnout pomalu" - it may take some time as the file is 55 MB+ and the download service is free) there is 7zip archive with 5 music

I've just installed RC2 and I'm very excited about the quality. It's so much better than MP3. This is the first version I've used since I just found out about Ogg Vorbis. I did notice that very high frequencies seem to be missing but since not many people can hear much above 18 KHz it's not much of an issue. I suppose this resolves the hiss problem so prevalent in MP3.

Hi, The call waiting tone is very annoying (you hear nothing while it plays the beep). I need callwaiting because of the queues (the phone has to ring as soon as you hangup) but I want to remove the beep on my dahdi channels, how can I do? Thanks, Niccol? -- http://www.linuxsystems.it

Canuck15, No, I hadn't played with the gains. But I've now done so and no difference unfortunately. Thanks for the suggestion though. I have discovered that after Asterisk has answered the call and the remote caller has hung up, if I lift the receiver on a phone connected to the line (in parallel with Asterisk), Asterisk then DOES instantly hang up. Would it be reasonable to assume the

...ot, I can usually set up an alias on OnSIP (but they don't do # in the URI) I tried Fring, but it didn't seem optimal. I'll try again since many of you have said it works well. The two apps I mentioned both work and sound fine incoming, but the outgoing audio is either noisy (iSip) or inaudibly distorted. Note these apps are NOT free. ALso, I haven't tried contacting support yet, but I will do so after the conference call today. Thanks, r _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13...

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this

...als. A good one is EncoderHell from AirWindows.com . It was the only case where I could see (not hear) some pre-echos, and some differencies between "-q 3" and the upper quality levels. Another difficult signal is the sine wave (1 KHz), that is a little wobbly at "-q 0", but inaudibly so at "-q 3". The only difference I could find between the original and the "-q 3" encoded was a static section of a complex arrangement (cannot recall which sample :^( ), that sounded a little different, maybe more "metallic", but only an immediate comparison coul...

Hello Jean-Marc, yes, it works for both 32 and 48 kb/s. Also this option marginally increases the size of the file which is understandable. I tried a few other music files and found that this stereo narrowing happens not only in the case of track I submitted (although in smaller scale) so it would be great if the final libopus 1.2 had officialy a parameter to force music "mode" of

I realize that hearing tests are far superior for calibrating codec internals, particularly the perceptual enhancement. The tester is for finding gross mistakes like overflows. The test program includes code to subsample the 32Khz source down to 8 and 18 Khz. I've broadend the tests to try all the possibilities like VBR, perceptual enhancement, complexity and quality. The results are

Hello, Merry Christmas and happy new year for everyone. I've started porting Speex 1.2rc1 on a 32-bit microcontroller, so far everything goes fine for Norrowband modes. But when I'm trying to encode in wideband mode with quality higher than 5 then decode my file back, I get inaudible "thing" (just strong noise). - All frames are decoded with correct result (decoder function