search for: s16_le

...e a hand here... I need to run speex on a FPU-less platform with an audio card that only reads samples of type S32_LE (even when it is a 16bits audio card). I understand that if my platform is FPU-less then I should use speex_encode/decode_int() but then, how can I convert my S32_LE sample to S16_LE in order to pass it to speex functions ? thanks in advance...

...the same mixer with all users slave { pcm "hw:0,0" period_time 0 # needed by oss (= 0) period_size 1024 # needed by oss (= 2^n) period_size 8192 # needed by oss (= 2^n) format S16_LE rate 48000 } } pcm.!dsnoop { type dsnoop # plugin type ipc_key 5678293 # unique ipc id ipc_perm 0666 # would like to use icp_group instead ipc_key_add_uid no # share the sam...

...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 Hz, Stereo Encoding 44100 Hz audio using stereo (93 bytes per packet) paul at x61laptux:~/src/celt-0.5.2/tools$ thank you, Paul.

...below, as shown by: cat /proc/asound/card0/pcm0p/sub0/ > hw_params > slave { > # 2 for stereo, 6 for surround51, 8 for surround71 > channels 2 > pcm { > format S16_LE > #format S32_LE > > #rate 44100 > rate 48000 > > # http://www.alsa-project.org/ > alsa-doc/alsa-lib/pcm_plugins.html >...

...ng Linux (2.6) Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz if saved out 'as is' can be played with "aplay" using the following command: "aplay -c1 -r16000 -f S16_LE -t raw <fileame>" Bellow is part of some test code that I have been trying to get to work, I have removed some of the code to make it easyer to see. Any help would be wonderful, think of it as an heroic act by saving me from a mental break down. :) Thanks, David. ==== s...

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

...ctions 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 contains #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw Am running asterisk as root. When i'm playing a music file (using amarok), my music onhold is silent. Is there anything I can do? Any help of pointers will be appreciated. Regards, Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium....

...rg/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 /usr/sbin/ast-playlinein contains #!/bin/bash /usr/bin/arecord -q -c 1 -r 8000 --buffer-size=2048 -f S16_LE -t raw Am running asterisk 1.4 as root on suse 11.0 and 10.2. When I'm playing a music file (using amarok), my music onhold is silent. Is there anything I can do? Any help of pointers will be appreciated. Regards, Tim

...ibcelt (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 it using this commandline : celtenc --cbr --rate 48000 --bitrate 48 --stereo --comp 5 test.pcm test.oga The encoder just writes a 173-byte header into the *.oga file with following output at stderr: Encoding 48 kHz stereo audio in 20...

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

...----- Listen 3000 # Streaming Ogg Vorbis from stdin, using the special # OggStdin module that caches Ogg Vorbis headers <OggStdin> Path "/stream.ogg" Type "audio/ogg" </OggStdin> ----- You can run this with a shell pipeline like: $ arecord -c 2 -r 44100 -f S16_LE -t wav | oggenc -o - - | sighttpd -f examples/sighttpd-oggstdin.conf And you can connect to it as an Ogg stream, eg: $ ogg123 http://localhost:3000/stream.ogg Details ======= The following changes have been made since release 1.0.0: Conrad Parker (11): add configure check for libogg...

...----- Listen 3000 # Streaming Ogg Vorbis from stdin, using the special # OggStdin module that caches Ogg Vorbis headers <OggStdin> Path "/stream.ogg" Type "audio/ogg" </OggStdin> ----- You can run this with a shell pipeline like: $ arecord -c 2 -r 44100 -f S16_LE -t wav | oggenc -o - - | sighttpd -f examples/sighttpd-oggstdin.conf And you can connect to it as an Ogg stream, eg: $ ogg123 http://localhost:3000/stream.ogg Details ======= The following changes have been made since release 1.0.0: Conrad Parker (11): add configure check for libogg...

...d 96000 bits/sec with command ./opus_demo -e voip 48000 2 96000 music_orig.wav testcase30.opus Then decoded with same command you gave ./opus_demo -d 48000 2 testcase30.opus testcase31.wav And played using command (on my ubuntu 14.04 linux box) aplay -t raw -c 2 -r 48000 -f S16_LE testcase31.wav and was able to hear the audio clearly 2. The file that gets generated with "./opus_demo -d 48000 2 testcase30.opus testcase31.wav" (testcase31.wav) is just a raw data file.. It does not have wav header. For example, if you do $ file music_orig.wav...

...false ipc_key 1025 ipc_perm 0660 ipc_gid audio # Don't put the rate here! Otherwise it resets the rate & channels set below, as shown by: cat /proc/asound/card0/pcm0p/sub0/hw_params slave { # 2 for stereo, 6 for surround51, 8 for surround71 channels 2 pcm { format S16_LE #format S32_LE #rate 44100 rate 48000 # http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html # Maybe helps nonblock true type hw card "PCH" device 0 subdevice 0 } # mplayer2 chooses 1024 # period_size 512 with buffer_size 16384...

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

...ng Linux (2.6) Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz if saved out 'as is' can be played with "aplay" using the following command: "aplay -c1 -r16000 -f S16_LE -t raw <fileame>" Bellow is part of some test code that I have been trying to get to work, I have removed some of the code to make it easyer to see. Any help would be wonderful, think of it as an heroic act by saving me from a mental break down. :) Thanks, David. ==== s...

How are you capturing the audio-in? Jack Elliott On 8/27/2019 5:25 PM, webmaster at berean-biblechurch.org wrote: > Yes, I'm using it to capture the audio-in and pushing to Icecast. > > On 2019-08-27 10:33, Jack Elliott wrote: >> As described here in ffmpeg's documentation: >> >> http://ffmpeg.org/ffmpeg-all.html#Icecast

...Input stream: Signed 16bit Little endian, mono, PCM @ 16000Hz > if saved out 'as is' can be played with > "aplay" > using the following command: > "aplay -c1 -r16000 -f S16_LE -t raw > <fileame>" > > Bellow is part of some test code that I have been trying to get to work, > I have removed some of the code to make it easyer to see. Any help would > be wonderful, think of it as an heroic act by saving me from a mental > break down. :) > &g...

Hi all, i have just finished my installation of wine as normal user (see below). Notepad is running fine, so i decided to run something more fun, starcraft. The installation program works, even with sound and all, exept when i decide to run the previews. The previews run rine, but whithout sound. When i return to the installation program the sound is gone. But anyway, it does it's yob,