similar to: RE: Playback 0.5% Too Fast?

Hi All I have a problem with IVR scripts which consist mainly of Playback of audio files, driven from an AGI application. There are clicks every few seconds or more frequently that is audible on the remote end (PSTN), but not on the Asterisk recording of the call. If I record the remote end and compare it to the local recording, it appears to be about 5%-7% too fast - i.e. if I synchronise the

I'm collaborating with and electronic musician to experiment on the production of music from number sequences. As I'm an R user I started playing around with the sound library and I found it very useful. However there are several things I do not understand (I'm not an expert in acustic nor audio signal treatment). The first thing I'd like to understand is: let s be a normalized

Jamie, I hear what you're saying. I don't believe this *should* be a plugin's responsibility, though it sounds like with XMMS it is. But I don't know how to fix it. Probably with enough archaeology into the XMMS source and other plugins I could find out. I'll file it in the feature requests and hope someone can get to it. Josh --- Matt Zimmerman <mdz@debian.org>

I am forwarding your request to the FLAC development mailing list. ----- Forwarded message from Jamie Heilman <jamie@audible.transient.net> ----- Date: Sun, 8 Sep 2002 16:13:32 -0700 From: Jamie Heilman <jamie@audible.transient.net> Resent-From: Jamie Heilman <jamie@audible.transient.net> To: submit@bugs.debian.org Subject: Bug#160155: gapless playback Package: xmms-flac

>>> Jan Stary <hans at stare.cz> schrieb am 05.11.2018 um 11:05 in Nachricht <20181105100534.GB44329 at www.stare.cz>: > (Are we off‑list now by intention?) No, just fooled by the list defaults (some need just reply, others need reply to all) > >> Did you also try to listen at the beginning, shortly before the real tone > appears in the audible spectrum?

Novex NUPS-650 USB information $ lsusb -s 002:027 -v Bus 002 Device 027: ID 0001:0000 Fry's Electronics Device Descriptor: ??bLength ???????????????18 ??bDescriptorType ????????1 ??bcdUSB ??????????????1.00 ??bDeviceClass ???????????0 (Defined at Interface level) ??bDeviceSubClass ????????0 ??bDeviceProtocol ????????0 ??bMaxPacketSize0 ????????8 ??idVendor ??????????0x0001 Fry's

Hi Jim, First of all - thanks, turning the highpass filter off was what I needed, and the waveforms match now. But, when i did the PESQ tests again I found an interesting result : version 1.0.5 still got a slightly better average score, but the standard deviation on version 1.2 beta1 was much smaller. The cause for that is this - on some samples versions 1.0.5 and 1.2beta2 produced a single

Hi! I'm totally new to audio programming. I managed to create a VoIP with waveform (PCM). The problem was the really high traffic, so I desided to use speex for compressing the data. I'm recording with mono, 16bit/sample, 8000 samples per second. I tried to add the speex compression, but I always get a crash when I try to decode the data. Here is my encode and decode function: int

Hello, I have an p2p voice chat application using WAVE PCM (winmm). Now i am trying to add opus encoding to it to send it over the TCP/IP and then decoding it back to play - but without success (without opus it works ok) Here is an example of my code. I get message from input device then encode with opus then decode it back to output wave header and play. Doing it i hear only noise in my

I'm having a problem trying to write a simple vorbis file player using DirectSound. The decoding portion is pretty much straight from the vorbisfile sample code. The pcm data gets put into one half of a DirectSound buffer, and as that half is playing, the next half gets filled. The problem is, there's skipping and some noise when the file is being played (although some of the music is

Does Asterisk need a sound card or functional Console/dsp to answer inbound DID number from PRI and playback .gsm files? I can call from any of the SIP extensions on Asterisk and hear audio from Playback(), MeetMe(), or MOH. The problem I am having with calls from my PRI is as follows: I have an Asterisk (CVS-HEAD-08/25/04-20:28:51) currently interfacing a NEAX 2400 IPX with PRI. I have a

I am developing apps using ARI which need suppression of DTMF tones in the audio, and I have been told (back in December) that asterisk depends on SIP providers to suppress DTMF tones in the audio stream. Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that SIP providers are not very good at doing that suppression (leaving audible clicks, or failing to suppress the tones

Sounds like either your machine is too slow or you have some other buffer issue. A decode thread is usually ideal so that data can be decoded before it is needed by the buffer fill operation. You might be able to get a sasifactory solution by increasing the number of buffers and using them in a round-robin fashion 1-2-3-4-1-2-3-4. Wouldn't do any more than 1/4 sec per buffer. Cheers, Chris

-----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]En nombre de asterisk-users-request@lists.digium.com Enviado el: jueves, 02 de febrero de 2006 10:15 Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 19, Issue 15 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To

I am specifically using Celt Wideband (48kHz) over WiFi multicast that naturally leads to lost packets and am trying to minimize the impact to the audio. I implemented PLC but the audio it produces is robotic. Have I implemented PLC correctly? Checking the waveform it is using the previous received waveform to fill in a missing packet but not the full waveform so it has to repeat. Basically,

I have been looking at the medium-rate codecs in Asterisk - ADPCM and G.726. Both of these are adaptive PCM codecs - the G.726 one is a little more expensive in processing power, however both are 32k bit-rate. I am experiencing problems using G.726 where the audio level is high. It produces loud clicks as if clipping. For quiet audio however, it seems fine. ADPCM (Digilogic VOX?) seems to be

Hello, I am trying to install R packages under linux, some of the packages can not be installed and I got the following error, could anybody give me suggestion on where the problem is and how to fix it? Thanks-e > .libPaths() [1] "/usr/lib64/R/library" [2] "/usr/share/R/library" [3]

I have three time series, x, y, and z, and I want to analyse the relations between them. However, they have vastly different resolutions. I am writing to ask for advice on how to handle this situation in R. x is a stimulus, and y and z are responses. x is a rectangular pulse 4 sec long. Its onset and offset are known with sub-millisecond precision. The onset varies irregularly -- it doesn't

I am puzzled by a filtering problem using fft(). I don't blame R. I have a waveform y consisting of the sum of 2 sinewaves having freqs f1 and f2. I do s = fft() of y. Remove s's spike at freq=f2 Do inverse fft on s. The resulting waveform still has a lot of f2 in it! But the filtering should have removed it all. What is going on, and how to fix?? Thanks very much for any help. Bill

What do you mean by "implement"? You're just using the Opus built-in PLC (passing NULL), right? The PLC generally attempts to find periodicity and replicate it. I guess if your signal isn't periodic it can lead to a repetition that isn't great. It's something that could probably be improved in the PLC. Cheers, Jean-Marc On 08/25/2015 01:21 PM, Scott Boekweg wrote: