similar to: reducing latency?

Hi, What kind of latency is expected using 8,16,32 khz? I am trying to do a realtime stream server, and I am having latency above >500ms ( I capture the sound using the mic, encode it and send it to the client). I am using ALSA system, a SB128 PCI and a 800Mhz P3 . What can I do to lower the latency? I tried a test : arecord -t raw | speexenc - - | speexdec - and I found that I also have a 500

My problem is this. I am connected via fwd's iax protocol. I try to call the echo test, sometimes I get an answer, sometimes not. When I do, sound is about one second lagged - I still hear the ring about a second after I get the answering on console message, and the echo test takes about a second or second and a half to get back the answer. When I use asterisk with dmix and dsnoop, calling

Hi u, On Thu, 22 Jan 2015, unosonic wrote: > well, sort of, with a PI, Ices2 as a client, and an USB audio device as > input, e.g. a "class compliant" audio recoder like Olympus LS5, Zoom H2 > etc., but also a little mixer like the Behringer Xenyx 302 USB. > The problem i've encountered is with USB class1 devices, i.e. so called > "full speed" USB (which is

Hi, I'm trying to record something for about 24 hours but oggenc tells me that I have 203m remaining. Quote: durand at Deuterium ~> arecord -f cd -t wav | oggenc -b 192 -o file.ogg - Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Opening with wav module: WAV file reader Encoding standard input to "file.ogg" at approximate bitrate

Hi all, Just found out about this codec and I'm really impressed. I compiled celt-0.5.2.tar.gz on my desktop and tried out a few tests. I then did a native fixed point arm compile on my beagleboard which also worked a treat. Before I get started with the library I was trying to see if I could grab some real time audio, encode it and write to a file using arecord in conjunction with

On 01/04/2008, Erwan A <mout551 at hotmail.fr> wrote: > > Hi, > > Yes i agree with you. You don't have to delete these files. > > But if i cross compile with ogg header files, i have the following error : > > > /usr/lib/libogg.so: could not read symbols: Invalid operation > collect2: ld returned 1 exit status > make[2]: *** [speexenc] Erreur 1

Hi all, I try to compile speex for ARM. I follow the README.arm and i always have the same errors : make[2]: entrant dans le r?pertoire ? /usr/src/linphone/arm/speex-1.1.6/src ? source='speexenc.c' object='speexenc.o' libtool=no \ depfile='.deps/speexenc.Po' tmpdepfile='.deps/speexenc.TPo' \ depmode=gcc3 /bin/sh ../depcomp \

I have a Debian Sarge box that I am trying to get icecast2 and ices2 up and running on. I'm having problems, but do not know where to look to identify the problem. I have my audio working with ALSA - microphone in plays through my audio out. I'm running a basic configuration using the defaults provided, and the ALSA example config file for ices. When I try to connect to the server using

On Fri, 5 Jun 2015, Michael Hennebry wrote: > On Thu, 4 Jun 2015, Frank Cox wrote: > >> On Fri, 5 Jun 2015 00:54:24 -0500 (CDT) >> Michael Hennebry wrote: >> >>> Actually, I mistook black for green. >>> Green was occupied by my external speakers. >>> Putting the headset's green plug in the green spot >>> lets VLC and aplay make

Hi, this problem seems to be coming up a lot lately -- the README.arm in linphone tells people to remove libogg headers, but speexenc and speexdec require them to build. Perhaps we need a README.arm in libspeex telling people not to remove the libogg headers. K. On 01/04/2008, Erwan A <mout551 at hotmail.fr> wrote: > > Hi all, > > I am following the README.arm from Simon

Hi apols as not really speex dev but really struggling to find this info that sure you will know in an instant. I am trying to compile alsa-plugins so I get the speexdsp as well as the rate conversion and just don’t know how. pi at raspberrypi:~/alsa-plugins-1.2.2 $ arecord -D mic ALSA lib dlmisc.c:287:(snd1_dlobj_cache_get) Cannot open shared library

I am having problems getting Ices to work with ALSA. If I use arecord (arecord -D plughw:0,0 output.wav) I can record the input from my microphone into a file. The playback works perfectly. However, if I use Ices with the same values I hear nothing on the receiving end of my broadcast (although I can connect to it). In order to do a further test, I setup a <savefile> tag in my

Hi all, New to list and Opus Codec. I've been using it on a beagleboneblack with 'arecord'. One of the things I've noticed while encoding is that Opus reports a float with an 'x' along with "realtime". Where can I look up where this calculation is derived? It got me looking and I've noticed that when I record something from a known source (in testing)

Hi, Earlier this day I started a thread about Skype not working with outgoing sound. Now I wonder if it's not a problem with my soundcard. I had a vague suspicion about this, but since it's way off the original topic, I decided to start a new thread about this. Here's what the soundcard looks like: [root at grossebertha etc]# lspci | grep -i audio 00:1b.0 Audio device: Intel

What sould be the capture and playback buffer size in 8,16,32 khz for the Alsa system? Can this also causing latency? In the server side I have: -A thread that reads the input from mic (capture) ands copies to main buffer. -The main loop encodes and sends it to the client ( it's read the data from the main buffer) Client: -A thread for

I am using usb-audio for Console/Dsp with asterisk. it is crashing 1.4.21 and also svn. During the brief times its working the audio is choppy but understandable. I have used aplay and arecord at the same time on the same wave file and they work fine every time and I have done it MANY times. Asterisk failes after 1 or 2 times. Any ideas on something I can try? Jerry

On Aug 23, 2004, at 9:35 PM, Ralph Giles wrote: > On Mon, Aug 23, 2004 at 11:49:27AM -0700, Eric Smith wrote: > >> I've tracked the problem down to abuse of the C type size_t in >> aplay.c. So I'm pretty sure it's not a problem in oggenc. > > Hmm. Wouldn't have expected that from alsa, though if it's saving > as 'wav' you have to sort of

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

Hello, I have a computer with this sound card (from lspci): 0000:01:04.0 Multimedia audio controller: Creative Labs SB Audigy LS Alsa is installed (maybe not properly...) # cat /proc/asound/cards 0 [CA0106 ]: CA0106 - CA0106 Live! 7.1 24bit [SB0410] at 0xc800 irq 10 But I have the impression alsa is ok : :~# aplay -l **** List of PLAYBACK Hardware Devices **** card

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >