similar to: CPU load

Karl Heyes <karl@xiph.org> writes: > On Mon, 2004-10-18 at 18:27, Jack O'Quin wrote: > > Karl Heyes <karl@xiph.org> writes: > > > > > Are you running with realtime privileges, for this you need to start as > > > root if you want that. Even with realtime privileges there may be odd > > > cases where the scheduling latency is a bit too

the "Adv. Rec"-option in the oddcast-plugin for winamp doesn't work on my machine (winxp with winamp 2.81 and the latest oddcast version). it seams to be buggy. i can enable it (green led lights), but nothing happens. and when i disable it winamp crashes down. there is also no configuration to select a soundcard input channel (i am using rme hdsp with 8 analog, 9 digital

hello, I've just got ices-kh60 running with jack. It's working OK apart from a problem where the latency on the stream creeps up from about 15-20 secs to around 7 minutes. The stream plays OK, but the audio coming out is 7 minutes behind what goes in. I'm not sure how this can be happening, seeing as I don't think there are any buffers that long in the system, but it is.

For "include/uapi/*", excluding "linux/" sub-directory, let all files' macro prefix match the standard format, and give related stand comments for their macro suffix. The related standard format is: "_SUBDIRNAME_SUBDIRNAME[_SUBDIRNAME]_FILENAME" (1st _SUBDIRNAME is _UAPI), and use '_' instead of '.' and '-'. Signed-off-by: Chen Gang

For "include/uapi/*", excluding "linux/" sub-directory, let all files' macro prefix match the standard format, and give related stand comments for their macro suffix. The related standard format is: "_SUBDIRNAME_SUBDIRNAME[_SUBDIRNAME]_FILENAME" (1st _SUBDIRNAME is _UAPI), and use '_' instead of '.' and '-'. Signed-off-by: Chen Gang

For "include/uapi/*", excluding "linux/" sub-directory, let all files' macro prefix match the standard format, and give related stand comments for their macro suffix. The related standard format is: "_SUBDIRNAME_SUBDIRNAME[_SUBDIRNAME]_FILENAME" (1st _SUBDIRNAME is _UAPI), and use '_' instead of '.' and '-'. Signed-off-by: Chen Gang

I've been using JackWASIO and 10.5 Leopard fine, but since I switched to Snow Leo 10.6 I can't find a way to use wineasio low latency driver at all. Have tried to install some version of Wine (that comes with WineBottler) and a new CrossOver Games 8.0 but no luck: wineasio doesn't appear in the Audio preferences of winecfg. Is there a clear way (for someone who aren't in developing

Hello again, does anybody know icecast sorce client which encodes live audio in mp3 and uses ALSA drivers. I've tried darkice, but it uses too many resources for several encoding processes. Maybe the reason is because I'm using virtual ALSA devices created for each program I want to encode. But I know only two ways different channels from the same soundcard to be used simmultaneously by

>I can't really comment on jackd itself, as I haven't looked that much >into it. If running jackd as root requires client apps to also run as >root then that is a pretty bad issue. I would of thought that running >jackd as a realtime process (whether root or not) would be an important >consideration. jackd (realtime) requires various resources which it shares with its

oddcast DSP is what you want, it supports live input (called Advanced Recording) and does support streaming to Peercast...It currently does not support Vorbis metadata for live recording (since metadata for vorbis streaming is inserted into the ogg stream, as opposed to how it's done for mp3 in which the metadata is updated on the server via URL calls)... anyway, it's the closest thing to

Hi, I hit a bizare problem with dovecot 2.2.7 on debian 9 with LMTP enabled and auth/penalty disabled as documented here : https://wiki.dovecot.org/Authentication/Penalty Use case : I run a swaks command to send an email to an exim4 that tries to make a callout to dovecot-lmtp. At RCPT TO: swaks hangs 5.0<something-small> seconds then process normally (exim is waiting for callout

Hello, I am new to Asterisk, so please bear with me. I have made a success installation from source of Asterisk 14.4.0 on Debian Jessie (8.7). And I am running the Asterisk server, with several extensions and dialplans, all working well. However I am struggling to get app_jack to run. In menuselect I can see that it is XXX due to dependencies on jack and resample, however both Debian packages:

Greetings! I now found someone to test gtalk with and found out, that app_jack has a problem here. My voice gets transmitted fine, but I only get white noise from the other party. I tried to set my JACK samplerate to 8000 to make sure it's no libresample problem, the results were the same. My setup is: Linux Debian Lenny Kernel: 2.6.30.4 PREEMPT (self-built) JACKd: jackd version

Hi, I can confirm that in the bad behavior, the 5 second delay occurs at each AUTH in our case. I think the configuration we have kill auth process at each end of AUTH (and fork a new one for next next AUTH). I think the "disable" flag is local to the process that is killed / respawned. A collegue of mine, Laurent Guerby, has found a workaround for us and it's findings seems

Hello everyone! So I hacked app_jack.c today, as best I could. Whic came mostly down to inserting ast_log() messages. I discovered the following with JACK: When it starts, it tries to read 512 bytes and only gets 0. That clears up after a while. Sometimes a good time later than the reading comes the writing. And there the real strangeness might begin. Because usually the framebuffer

Karl Heyes <karl@xiph.org> writes: > Are you running with realtime privileges, for this you need to start as > root if you want that. Even with realtime privileges there may be odd > cases where the scheduling latency is a bit too high, it all depends on > the drivers and kernel version but the current state is not that bad and > getting better. There are several ways to

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hello, still searching but didn't find very precise answers yet, is there a computer-aided musical creation -oriented flavor of CentOS (like the Ubuntu Studio)? Or would it be about gathering stuff from different repos and setting up everything by hand? A repository that would provide low-latency kernel (realtime one?): I've found the CentOS-RT from the CERN + (1)). Specific tools (from

Thanks J. It didn't work. On 05/10/2017 01:57 PM, J Montoya or A J Stiles wrote: > On Wednesday 10 May 2017, andre castro wrote: >> Hello, >> I am new to Asterisk, so please bear with me. >> I have made a success installation from source of Asterisk 14.4.0 on >> Debian Jessie (8.7). And I am running the Asterisk server, with several >> extensions and

Hello all, I would like to know what kind of AAC can stream icecast 2.2.0. Is it MPEG-2 AAC or MPEG-4 AAC? How much efficiency can be gained by using this kind of AAC compared to MP3? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/icecast/attachments/20041227/c7c14910/attachment.html