search for: arecord

...sed. 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 celtenc. I tried this using the below command but unfortunately the encoder just exists straight away with no errors, anybody else tried this? arecord -t wav -c 2 -r 44100 -f S16_LE | celtenc - test.oga Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100...

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 192 kbps (VBR encoding enabled) [ 0.1%] [203m23s...

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

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 /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_speex.so ((null): /usr/lib/arm-linux-gnueabihf/alsa-lib/libasound_module_pcm_speex.so: cannot open shared object file: No such file or directory) arecord: main:8...

...y 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 ms latency. <p>--- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'speex-dev-request@xiph.org' containing on...

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 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

How can I reduce latency on the following? arecord -r22050 -c2 -fs16_le | speexenc - - | speexdec - I have no coding experience in C, so can't write my own stdin/stdout wrapper. this is just an echo test, but it's severely lagged. the mailing list posts on this subject didn't give a solution. I used this over ssh, and managed to have a...

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 <s...

How do I make my headset work. Before I try it with anything else, I'd like to make it work with arecord and aplay. VLC and movie player can make sounds with my external speakers. though not in my headset yet. I've got an insignia NS-PAH5101 headset. As directed, the red connector is in the pink spot and the green connector in green spot in the back of my computer. arecord folloed by aplay just ge...

...ther recording, but went for an apt-get update / dist-upgrade which installed (among others) a new rpi-update, and running rpi-update gave me kernel 3.18.3+ After a reboot, playback worked like a charm, no pops, clicks, ticks, just perfect playback. No messages in syslog either. Then I tried: arecord -D hw:1 -f cd testrec.wav But as soon as something other than silence was sent to the device, lines like these started to show up: Jan 22 23:17:48 testpi kernel: [ 9190.974009] Transfer to device 4 endpoint 0x4 frame 1697 failed - FIQ reported NYET. Data may have been lost. Jan 22 23:17:48 testp...

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) li...

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

...ayer, VLC and the likes without any problem. On the other hand, when I try to record sound over the soundcard, the result is quite fuzzy, full of distortions. I tried to fiddle with alsamixer to get rid of the distortions, but to no avail. Here's what I did to record sound over the mic: $ arecord > ~/test.wav Now in the past I configured a laptop with an ICH sound card on it, a small ASUS. I spent like whole afternoons googling and searching through forums, until some day I stumbled over the solution. I had to add something to /etc/modprobe.conf, something like: alias snd-card-0 snd...

...[repeated indefinitely until killed] Hmmm. Since these are occurring when reading a channel, it indicates that it's the attempt to read audio from pulseaudio which isn't working. Have you confirmed that pulseaudio is in fact working correctly in both directions (using e.g. both aplay and arecord)?

...e says "Input volume" 100%. >> There is also a row of 15 rectangles labeled "Input level". >> None of them are highighted. > > Also, I tried different settings for connector. > All gave me static except for Line-in. > That one gave me silence. When I run arecord the applications tab tells me that an alsa plug-in, aplay, is running. I'm aware that aplay is an alias for arecord. I need a nap. -- Michael hennebry at web.cs.ndsu.NoDak.edu "SCSI is NOT magic. There are *fundamental technical reasons* why it is necessary to sacrifice a young goat...

...erver, 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 ms latency. > > There are likely 3 problems with your setup: > 1) arecord might have a big latency (probably reads the audio big chunks > at a time) > 2) speexenc/speexdec has a larger latency than...

...gup the following error messages show up: NOTICE[]: channel.c:1683 ast_set_read_format: Unable to find a path from SLINR to UNKN WARNING[]: app_record.c:287 record_exec: Unable to restore read format on 'ALSA/default' Kernel 2.6.5 and Asterisk CVS 04-97-23 are installed. Some test using arecord and aplay shows that microphone and speakers are working. Any help would be appreciated. -- Stefan Tichy <asterisk@pi4tel.de>

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