similar to: S32_LE to S16_LE

Hi, I found the main reason for sound distortions on my system is guest timer configuration. The working one is: <clock offset="localtime"> <timer name="hypervclock" present="yes"/> <timer name="hpet" present="no"/> <timer name="rtc" present="yes" track="guest"

Hi, I can’t seem to find a solution for my last VM issue. I have distorted sound, when I play the Windows 10 device test sound. Only the Windows sound is affected. Mpd of the host is playing just fine meanwhile. I am using ich9 as emulated card, which is detected and installed correctly. The VM is using vfio-igd passthrough, so vnc or spice are not used. I tried to match the Windows driver

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

hello, can I get out of wine more than 16 bit output? sound_file -> player -> wine_out, which is 24/32 to linux? somethink like that, I am audiophile :( thank you

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

Hi All, I'm very new to speex and in fact handling audio at all, it seems I have run in to a problem I seem unable to fix. I'm trying to take audio from a microphone using alsa, then encode it as speex and save to disk. I have been wondering if it has something to do with endian type, but speexenc and speexdec works fine. Currently I have the following setup: Platform:

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's

I have searched all around and combined items in an /etc/asound.conf file to have by default a capture and multiple playback for alsa. I am looking for a way to: something like "aplay --nonblock test.wav" having at least 2 active one time and at that same time do "arecord --nonblock -d 1 input.wav" I have something like: pcm.!dmixer { type dmix

From: Sergei Golovan <sgolovan@gmail.com> Replace the use of liberl_interface, which is removed in Erlang 23, by libei. The implementation uses the ei_decode_iodata() function which has been introduces only for Erlang 23, so it doesnt work with earlier Erlang versions. --- erlang/Makefile.am | 1 - erlang/main.c | 312 +++++++++++++++++++++++++-------------------

Maybe what Centos needs is a bridal registry. Here in the US, an engaged couple can tell their friends what they'd like to be given as wedding presents. They do this by listing items in a registry, in various stores around town. Anyway, the idea is, post stuff you need in a list on your site. Say you need 20 hard drives, or a particular power supply, or whatever items that get consumed in

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Greetings, In looking for inexpensive JBOD and/or RAID solutions to use with ZFS, I''ve run across the recent "VTrak" SAS/SATA systems from Promise Technologies, specifically their E-class and J-class series: E310f FC-connected RAID: http://www.promise.com/product/product_detail_eng.asp?product_id=175 E310s SAS-connected RAID:

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

Hi, I'm currently testing libcelt (celt-0.9.1) on my Openwrt/Kamikaze RDC-3210 platform and although I'm able to Crosscompile on my Ubuntu 10.10 machine without errors, the resulting celtenc doesn't seem to work properly. Ran ./configure via the Openwrt makefile first (activating fixed-point) ; I first produce a S16_LE Raw PCM file (48Khz, stereo) : test.pcm Then I try to encode

Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no success. i have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and =/usr/sbin/ast-playlinein

Hi, I would like to get musiconhold from a sound card. This is because I want to kind of be a DJ and easily change the music playing, etc. However, I followed the instructions at http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and other tutorials on the net but no success. I have [mycustom] mode=custom directory=/var/lib/asterisk/mohmp3 application=/usr/sbin/ast-playlinein and

sighttpd 1.1.0 ============== Sighttpd is an HTTP streaming server designed for distributing realtime input. It is particularly useful for making camera streams available to multiple clients, and has been designed for embedded systems use. This release is available as a source tarball from: http://www.kfish.org/software/sighttpd/ New in this release =================== This release

sighttpd 1.1.0 ============== Sighttpd is an HTTP streaming server designed for distributing realtime input. It is particularly useful for making camera streams available to multiple clients, and has been designed for embedded systems use. This release is available as a source tarball from: http://www.kfish.org/software/sighttpd/ New in this release =================== This release

Hi Sakharam, I see 2 potential issues with what you are doing. 1. ./opus_demo -e voip 48000 2 16 music_orig.wav testcase30.opus in above command, "16" for bits/sec seems too low. I'm no audio expert, but just cant convince myself you can get any reasonable audio data with 16 bits/sec. FYI, I was able to encode and decode with 16 bits/sec, but when I played the decode file with