similar to: No sound, no information why, where to start?

Hi all, I am trying to extract from the aplay -l output the Card and Device numbers. I had a script that was working but found another motherboard that has an EXTRA word. HDA Nvidia and HDA ATI HDMI is what I am talking about below. how do I search for a line that has HDMI (which I know how to do), then ignore everything upto the device work and take the next field after that? Thanks, Jerry

>> 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 >

Hi, i want send sound to my yamaha amplifier trough digital output from my audio-card. When I make: speaker-test -Dplughw:0,1 -c2 i can hear the sound. If I set the DevicePCM1 to default i can hear sounds from analog output trying to set hw:0 hw:0,1 plughw:0,1 didn't produced any sound (analog nor digital). How should I setup Alsa Driver in the wine registry? asound.conf and .asoundrc is

I am currently using a gentoo, (so no pulseaudio installed) and I can't get any sound with wine. I 'm trying to use alsa because winecfg deleted the entry of oss. My wine version is 1.3.24 The sound worked one time but doesn't still work...I uninstalled wine, deleted the .wine directory and re-installed but it didn't fix the problem. These are details about my soundcard : >

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"

I am trying to connect audio through HDMI on a config. aplay - l gives: **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: VT1708B Analog [VT1708B Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 So I change my

I have this line: ALSA=`aplay --version` in a script. when I execute the script I get the message line 187: --version: command not found when I do "aplay --version" on the command line it works just fine. What is happening here, --version is a valid command line option? Thanks, jerry

* EvilOverlord (eviloverlord@kucs.net) écrivait : > In whatever you use as mixer control for the soundcard (alsamixer, > aumix, etc) what is set as the "capture" channel? Your soundcard may > not support capturing what is being played. If it helps : my soundcard is a ES1988 Allegro-1, the module is "maestro3". I use kmix as mixer, but there is nothing in it about a

* Enrico Minack (enrico.minack@informatik.tu-chemnitz.de) écrivait : > > I use kmix as mixer, but there is nothing in it about a > > "capture channel". How could I find where it is defined ? > then try alsamixer or amixer and watch out for capture and unmute and apply > this for the according channel (mic, line-in, pcm or master) Hum... the problem is that RH 9 uses

I have tried broadcasting from both SuSE Linux and Slackware and with various programs being the last one Tim`s Mixice (which is a perl script that works along with a program called catdsp). Looking at the source I saw that this programs, both of them, try to open /dev/dsp. Now, my question is if /dev/dsp is the standard device for the line-in of the audio card? I have tried, as suggested in

* John McHarry (jmcharry@whqr.org) écrivait : > Have you tried aumix from the command line? I have used it in RH 9 and > Fedora, although not with ices. Yes, I did : in fact, aumix from the command line behaves exactly like the KMix GUI : when I change the PCM or Mic level with aumix, KMix shows the new settings. But even with aumix set to the good values (PCM to 100%), ices get no sound.

I am running pulseaudio on centOS 6.5 audio works fine as the user. running the command aplay /usr/share/sounds/alsa/Front_Center.wav works fine. when I login as root and run the command su user -c "aplay /usr/share/sounds/alsa/Front_Center.wav" it does not work. I thought su actually runs as that user... is there some "magic" to su I don't know about? How might I get

https://bugs.freedesktop.org/show_bug.cgi?id=67051 Priority: medium Bug ID: 67051 Assignee: nouveau at lists.freedesktop.org Summary: No nouveau HDMI sound on NVIDIA GT430 Severity: normal Classification: Unclassified OS: Linux (All) Reporter: alupu01 at gmail.com Hardware: x86 (IA32) Status:

Hey guys, I have a mic plugged into my sound card. It worked late last night but not today. When I try to select the mic using System/Preferences/Sound I'm provided with several options to choose from. On the Hardware tab I choose the Internal Audio device and set it to Analog Stereo Duplex. On the Input tab I choose Internal Audio Analog Stereo there are two Analog Microphone Inputs,

Hi all, I'm fairly new to icecast / ices - been using icecast (2.2.0) to stream audio files for a few weeks, it works very well, now I'm trying to add live feeds via ices (v2.0.1). I've built and installed (on SuSE 9.3 platform) with no apparent errors, copied ices-alsa.xml to user area and modified paths, server password etc. When I run ices (using "ices

On 03/29/2017 06:43 AM, ken wrote: > On 03/28/2017 08:53 PM, ken wrote: >> The www has failed me with this, so I'm trying you guys. Sound worked >> great out of the box when I installed 7.2... Yay! I could watch all >> kinds of videos, like on facebook and youtube. And I could listen to >> most podcasts too. But then something happened. It was either a >>

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

Ulrich Windl wrote: > Hi! > > Recently I made some recordings (44.1kH/24bit WAV), and after > post-processing (normalization, noide reduction) I saved them as flac (still > 44.1kHz/24bit). > I noticed that ogg123 can decode these, but it will output 24 bit only. > Specifically there's no option to transform the output to 16bit. > When trying to pipe the 24bit WAV output

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

When I use pulse audio and Asterisk 11.4.0 on the Console/Dsp port I get a motorboating sound or warble - or - just not clear audio. When I switch that to ALSA direct it sounds just fine. What might be happening with pulse audio that it does not sound clear??? asound.conf below. Thanks, Jerry more /etc/asound.conf # # Place your global alsa-lib configuration here... # @hooks [ {