Static at 16k on equalizer?

Alright I use pretty high grade headphones. I have them plugged into my reciever. When I watch a movie and a quiet scene comes up I hear static. It stands out alot. I played around with equalizer and it was at 16k. I turn it down and static goes down. I own a titanium x-fi pci-e and am using the optical port.

You can use the build_static tag supplied with the demo Makefile.
make -f demo_rdbms.mk build_static .....

Similar Messages

ALC1150 16k band equalizer bug! (MSI Z87-G45)

Hi all,
I know this would be better to report on Realtek forums, but this problem applies to all of you with ALC1150 chipset and who are using equalizer - maybe also other chipsets. Maybe also MSI stuff will be interested in solving this issue as it is rather annoying issue. Here's the description of the problem I've also sent to Realtek technical support (no answer from them yet). Please let me know if you can repeat this issue. Thx
Dear Realtek support,
I have noticed a quiet annoying problem with your graphics equalizer and I would like to report the problem. Would be great if you could localize the problem and fix the issue. I suppose it would be only a trivial thing which can be fixed fast.
Here are my system specs:
OS: Windows 8 64bit
Motherboard: MSI Z87-G45 (ALC1150)
Drivers: Realtek R2.71
The problem:
There is a problem with 16k band only. Other bands works fine.
Use Case:
1. Let's say I want to use value 12 for extra heights. I set the slider to 12 and play the song.
2 Value is ignored at the beginning, instead value 0 is used even though the slider still shows 12. If I move the slider let's say to 11 (music is still playing) the change is reflected immediately and everything works fine.
3. But only until I stop the song and play it again or play another song. This is really weird behaviour So seems the band registers changes only if the sound is playing and when there is no sound input it reset the band for 16k to 0 (please note that system still show previously set value - in my case 12, but it is just ignored)
So actually 16k band is quiet unusable as it still reset its value to zero when the sound stops.
You can easily simulate the behaviour like this:
1. Reset equalizer to flat and set only 16k band to max. (12)
2. Play some music. You can immediately hear there is no sound change, even you have set 16k band to max.
3. Move slider to 11 during the music plays. You can hear the change now...
4. Stop the music, restart the music player and play the music again.
5. Flat equalizer (16k = 0) is used again, but the slider shows 11.
Could you please have a look at this issue and inform me if you can repeat it ? If there are further questions, don't hesitate to contact me. Would be great if this issue could be fixed because the equalizer is really great in all other aspects.
Thanks a lot,
Michal

New finding... If I open some music in Music player (Windows 8) and then set in on pause (still running on background) and switch to desktop and play the same song in media player, 16k band settings is remembered. This is really weird but works as a temp solution until fix is out. Maybe this issue is only Windows 8 related, I really don't have an idea but I got this idea once I've played Saints Row 4 and there were no engine sounds. Then I open some music file and set in on pause on background and suddenly, the sounds were back. I would say there is something wrong in windows or what...

Equalization and Symbol Rate

Despite my number of posts on this board...I swear that I am happy with my NI devices....
I noticed a peculiar thing today. I am using some of the example programs to do some channel characterization. The investigation is meant to be relatively simple. I am using no baseband filtering, and since my transmitter and receiver are static I desire no equalization (in fact, right now, the up and down converters are connected via a short SMA as I do some "sanity checks".)
I am interested in utilizing as much of the 20MHz real time bandwidth as possible...so I went into the transmitter code (for example, the PSK Generate.VI located in the Modulation Toolkit Example folder) and hardwired 20MHz in for the bandwidth. With no baseband filtering, the original code determines BW by scaling the symbols/sec by some number (10 I think) which would limit the symbols/sec to 1MSym/sec.
Anyway...with my 20Mhz BW...I set the symbol rate at 100kHz. Using the PSK Demod VI in the same example directory, I observe the constellation and other measurements. No problems here. 1Mhz symbol rate....OK. Increase to 5Mhz...a little noisier, but OK. Higher than that (say 10MHz symbol rate) and the MER and EVM start going way up. Note that this is independent of the scheme (BPSK, QPSK, 8-PSK...whatever).
I noticed that if I then turn on equalization (on both the Tx and Rx sides of course), that the MER and EVM goes way down....especially at the high symbol rates.
So my question....I thought equalization was only necessary if the channel response changes over time (multi-path,etc.). Why then would equalization improve things when the symbol rate gets higher? Is it possible I'm seeing the frequency/phase offset of the upconverter and downconverter LO's? Or, maybe I'm just approaching the limit of the hardware and the equalization routines actually help clean up non-idealities of the hardware at those rates? For the channel I'm characterizing, I am actually interested in seeing the effect of potential multipath, so I'd rather not use the equalization routines at this time. Maybe I'm looking at having to make a tradeoff here?
Brandon

Hello Brandon,
The behavior you are seeing is expected. The equalization filter is used to correct the amplitude and phase of the signal so that after it passes though the 5610 you have a flat passband. There is a section of the RFSG help called IF Response and Software Equalization and it can be found under Devices >> NI 5671 Overview >> Advanced Topics >> IF Response and Software Equalization.
I copied the typical magnitude response of the 5610 upconverter with the equalization filter disabled (upper graph) and with it enabled (lower graph) from the manual. You can see it below. I hope this answers your question.
Regards,
Jesse O.
Applications Engineering
National Instruments
Message Edited by Jesse O on 07-19-2007 02:59 PM
Jesse O. | National Instruments R&D
Attachments:
equal.jpg ‏20 KB

[Pulseaudio] equalizer problem (solve)

Hello
I installed pulseaudio and all work fine but I tried to setup a system wide equalizer with the ladspa-sink module.
I think the module is well load but there is a problem with pa_sink_input_update_max_request function.
I am not alone in this situation :
https://tango.0pointer.de/pipermail/pul … 03834.html
http://www.mail-archive.com/pulseaudio- … 02398.html
here is my default.pa :
.fail
### Automatically restore the volume of streams and devices
load-module module-device-restore
load-module module-stream-restore
load-module module-card-restore
### Automatically augment property information from .desktop files
### stored in /usr/share/application
load-module module-augment-properties
### Automatically load driver modules depending on the hardware available
.ifexists module-hal-detect.so
load-module module-hal-detect
.else
### Alternatively use the static hardware detection module (for systems that
### lack HAL support)
load-module module-detect
.endif
### Load several protocols
.ifexists module-esound-protocol-unix.so
load-module module-esound-protocol-unix
.endif
load-module module-native-protocol-unix
### Load additional modules from GConf settings. This can be configured with the paprefs tool.
### Please keep in mind that the modules configured by paprefs might conflict with manually
### loaded modules.
.ifexists module-gconf.so
.nofail
load-module module-gconf
.fail
.endif
### Automatically restore the default sink/source when changed by the user during runtime
load-module module-default-device-restore
### Automatically move streams to the default sink if the sink they are
### connected to dies, similar for sources
load-module module-rescue-streams
### Make sure we always have a sink around, even if it is a null sink.
load-module module-always-sink
### Automatically suspend sinks/sources that become idle for too long
load-module module-suspend-on-idle
### If autoexit on idle is enabled we want to make sure we only quit
### when no local session needs us anymore.
load-module module-console-kit
### Enable positioned event sounds
load-module module-position-event-sounds
### Cork music streams when a phone stream is active
load-module module-cork-music-on-phone
# equalizer
.ifexists module-ladspa-sink.so
load-module module-ladspa-sink sink_name=ladspa_out master=alsa_output.pci_10de_774_sound_card_0 plugin=mbeq_1197 label=mbeq control=-15,-15,-10,-1,-5,-1,-1,0,0,0,0,0,0,0,0
#load-module module-ladspa-sink sink_name=ladspa_out master=alsa_out plugin=mbeq_1197 label=mbeq control=-15,-15,-10,-1,-5,-1,-1,0,0,0,0,0,0,0,0
.endif
### Make some devices default
set-default-sink ladspa_out
#set-default-sink output
#set-default-source input
here the output of pulseaudio -v :
I: main.c: Nous sommes dans le groupe « pulse-rt », permettant une planification à haute priorité.
I: main.c: RLIMIT_RTPRIO is set to 65, allowing real-time scheduling.
I: main.c: RLIMIT_NICE is set to 30, allowing high-priority scheduling.
I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) a échoué : Opération non permise
I: core-util.c: Successfully gained nice level -11.
I: main.c: Abandon de CAP_NICE
I: main.c: Pulseaudio 0.9.15
I: main.c: La taille de la page est de 4096 octets
I: main.c: L'ID de la machine est 900beb14464c78bd4d354b7a4960efd5.
I: main.c: Session ID is 900beb14464c78bd4d354b7a4960efd5-1243705227.439009-752724780.
I: main.c: Utilisation du répertoire d'exécution /home/max/.pulse/900beb14464c78bd4d354b7a4960efd5:runtime.
I: main.c: Utilisation du répertoire d'état /home/max/.pulse.
I: main.c: Exécution en mode système : no
I: main.c: De nouvelles horloges à haute résolution sont disponibles ! Bon appétit !
I: module-device-restore.c: Sucessfully opened database file '/home/max/.pulse/900beb14464c78bd4d354b7a4960efd5:device-volumes.x86_64-unknown-linux-gnu.gdbm'.
I: module.c: Loaded "module-device-restore" (index: #0; argument: "").
I: module-stream-restore.c: Sucessfully opened database file '/home/max/.pulse/900beb14464c78bd4d354b7a4960efd5:stream-volumes.x86_64-unknown-linux-gnu.gdbm'.
I: module.c: Loaded "module-stream-restore" (index: #1; argument: "").
I: module-card-restore.c: Sucessfully opened database file '/home/max/.pulse/900beb14464c78bd4d354b7a4960efd5:card-database.x86_64-unknown-linux-gnu.gdbm'.
I: module.c: Loaded "module-card-restore" (index: #2; argument: "").
I: module.c: Loaded "module-augment-properties" (index: #3; argument: "").
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Analog Stereo + Input Analog Stereo'
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3c failed
I: alsa-util.c: Error opening PCM device hdmi:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround40:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround41:0: Argument invalide
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround50:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:surround51:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround71:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Analog Stereo'
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Digital Stereo (IEC958) + Input Analog Stereo'
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3c failed
I: alsa-util.c: Error opening PCM device hdmi:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround40:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround41:0: Argument invalide
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround50:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:surround51:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround71:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Digital Stereo (IEC958)'
I: alsa-util.c: Device hdmi:0 doesn't support 44100 Hz, changed to 48000 Hz.
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Digital Stereo (HDMI) + Input Analog Stereo'
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3c failed
I: alsa-util.c: Error opening PCM device hdmi:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround40:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround41:0: Argument invalide
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround50:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:surround51:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround71:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Output Digital Stereo (HDMI)'
I: alsa-util.c: Failed to set hardware parameters on plug:surround40:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround41:0: Argument invalide
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround50:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:surround51:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround71:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Argument invalide
I: module-alsa-card.c: Found output profile 'Input Analog Stereo'
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D1c failed
I: alsa-util.c: Error opening PCM device iec958:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_hw.c: open /dev/snd/pcmC0D3c failed
I: alsa-util.c: Error opening PCM device hdmi:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround40:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround41:0: Argument invalide
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: (alsa-lib)pcm_params.c: Slave PCM not usable
I: alsa-util.c: Failed to set hardware parameters on plug:surround50:0: Argument invalide
I: alsa-util.c: Failed to set hardware parameters on plug:surround51:0: Argument invalide
I: (alsa-lib)pcm.c: Unknown PCM a52:0
I: alsa-util.c: Error opening PCM device a52:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Failed to set hardware parameters on plug:surround71:0: Argument invalide
I: card.c: Created 0 "alsa_card.pci_10de_774_sound_card_0"
I: alsa-sink.c: Successfully opened device front:0.
I: alsa-sink.c: Selected configuration 'Analog Stereo' (analog-stereo).
I: alsa-sink.c: Successfully enabled mmap() mode.
I: alsa-sink.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Using mixer control "Master".
I: module-device-restore.c: Restoring volume for sink alsa_output.pci_10de_774_sound_card_0.
I: module-device-restore.c: Restoring mute state for sink alsa_output.pci_10de_774_sound_card_0.
I: sink.c: Created sink 0 "alsa_output.pci_10de_774_sound_card_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: alsa.resolution_bits = "16"
I: sink.c: alsa.mixer_element = "Master"
I: sink.c: device.api = "alsa"
I: sink.c: device.class = "sound"
I: sink.c: alsa.class = "generic"
I: sink.c: alsa.subclass = "generic-mix"
I: sink.c: alsa.name = "CONEXANT Analog"
I: sink.c: alsa.id = "CONEXANT Analog"
I: sink.c: alsa.subdevice = "0"
I: sink.c: alsa.subdevice_name = "subdevice #0"
I: sink.c: alsa.device = "0"
I: sink.c: alsa.card = "0"
I: sink.c: alsa.card_name = "HDA NVidia"
I: sink.c: alsa.long_card_name = "HDA NVidia at 0xc0000000 irq 20"
I: sink.c: alsa.driver_name = "snd_hda_intel"
I: sink.c: device.bus_path = "/devices/pci0000:00/0000:00:07.0/sound/card0"
I: sink.c: hal.udi = "/org/freedesktop/Hal/devices/pci_10de_774_sound_card_0"
I: sink.c: hal.product = "HDA NVidia Sound Card"
I: sink.c: hal.card_id = "HDA NVidia"
I: sink.c: device.string = "front:0"
I: sink.c: device.buffering.buffer_size = "65536"
I: sink.c: device.buffering.fragment_size = "32768"
I: sink.c: device.access_mode = "mmap+timer"
I: sink.c: device.profile.name = "analog-stereo"
I: sink.c: device.profile.description = "Analog Stereo"
I: sink.c: device.description = "HDA NVidia"
I: sink.c: device.icon_name = "audio-card"
I: module-device-restore.c: Restoring volume for source alsa_output.pci_10de_774_sound_card_0.monitor.
I: module-device-restore.c: Restoring mute state for source alsa_output.pci_10de_774_sound_card_0.monitor.
I: source.c: Created source 0 "alsa_output.pci_10de_774_sound_card_0.monitor" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of HDA NVidia"
I: source.c: device.class = "monitor"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "HDA NVidia"
I: source.c: alsa.long_card_name = "HDA NVidia at 0xc0000000 irq 20"
I: source.c: alsa.driver_name = "snd_hda_intel"
I: source.c: device.bus_path = "/devices/pci0000:00/0000:00:07.0/sound/card0"
I: source.c: hal.udi = "/org/freedesktop/Hal/devices/pci_10de_774_sound_card_0"
I: source.c: hal.product = "HDA NVidia Sound Card"
I: source.c: hal.card_id = "HDA NVidia"
I: source.c: device.string = "0"
I: source.c: device.icon_name = "audio-card"
I: alsa-sink.c: Using 2 fragments of size 32768 bytes, buffer time is 371,52ms
I: alsa-sink.c: Time scheduling watermark is 20,00ms
I: alsa-sink.c: Volume ranges from 0 to 74.
I: alsa-sink.c: Volume ranges from -74,00 dB to 0,00 dB.
I: alsa-sink.c: No particular base volume set, fixing to 0 dB
I: alsa-util.c: All 2 channels can be mapped to mixer channels.
I: alsa-sink.c: Using hardware volume control. Hardware dB scale supported.
I: alsa-sink.c: Starting playback.
I: alsa-sink.c: Underrun!
N: alsa-sink.c: Increasing wakeup watermark to 30,00 ms
I: alsa-source.c: Successfully opened device front:0.
I: alsa-source.c: Selected configuration 'Analog Stereo' (analog-stereo).
I: alsa-source.c: Successfully enabled mmap() mode.
I: alsa-source.c: Successfully enabled timer-based scheduling mode.
I: (alsa-lib)control.c: Invalid CTL front:0
I: alsa-util.c: Unable to attach to mixer front:0: Aucun fichier ou dossier de ce type
I: alsa-util.c: Successfully attached to mixer 'hw:0'
I: alsa-util.c: Cannot find mixer control "Capture" or mixer control is no combination of switch/volume.
I: alsa-util.c: Cannot find fallback mixer control "Mic" or mixer control is no combination of switch/volume.
I: module-device-restore.c: Restoring volume for source alsa_input.pci_10de_774_sound_card_0.
I: module-device-restore.c: Restoring mute state for source alsa_input.pci_10de_774_sound_card_0.
I: source.c: Created source 1 "alsa_input.pci_10de_774_sound_card_0" with sample spec s16le 2ch 44100Hz and channel map front-left,front-right
I: source.c: alsa.resolution_bits = "16"
I: source.c: device.api = "alsa"
I: source.c: device.class = "sound"
I: source.c: alsa.class = "generic"
I: source.c: alsa.subclass = "generic-mix"
I: source.c: alsa.name = "CONEXANT Analog"
I: source.c: alsa.id = "CONEXANT Analog"
I: source.c: alsa.subdevice = "0"
I: source.c: alsa.subdevice_name = "subdevice #0"
I: source.c: alsa.device = "0"
I: source.c: alsa.card = "0"
I: source.c: alsa.card_name = "HDA NVidia"
I: source.c: alsa.long_card_name = "HDA NVidia at 0xc0000000 irq 20"
I: source.c: alsa.driver_name = "snd_hda_intel"
I: source.c: device.bus_path = "/devices/pci0000:00/0000:00:07.0/sound/card0"
I: source.c: hal.udi = "/org/freedesktop/Hal/devices/pci_10de_774_sound_card_0"
I: source.c: hal.product = "HDA NVidia Sound Card"
I: source.c: hal.card_id = "HDA NVidia"
I: source.c: device.string = "front:0"
I: source.c: device.buffering.buffer_size = "65536"
I: source.c: device.buffering.fragment_size = "32768"
I: source.c: device.access_mode = "mmap+timer"
I: source.c: device.profile.name = "analog-stereo"
I: source.c: device.profile.description = "Analog Stereo"
I: source.c: device.description = "HDA NVidia"
I: source.c: device.icon_name = "audio-card"
I: alsa-source.c: Using 2 fragments of size 32768 bytes, buffer time is 371,52ms
I: alsa-source.c: Time scheduling watermark is 20,00ms
I: module.c: Loaded "module-alsa-card" (index: #4; argument: "device_id=0 name=pci_10de_774_sound_card_0 card_name=alsa_card.pci_10de_774_sound_card_0 tsched=1").
I: module-hal-detect.c: Loaded 1 modules.
I: module.c: Loaded "module-hal-detect" (index: #5; argument: "").
I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: "").
I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: "").
I: module.c: Loaded "module-gconf" (index: #8; argument: "").
I: module-default-device-restore.c: No previous default sink setting, ignoring.
I: module-default-device-restore.c: No previous default source setting, ignoring.
I: module.c: Loaded "module-default-device-restore" (index: #9; argument: "").
I: module.c: Loaded "module-rescue-streams" (index: #10; argument: "").
I: module.c: Loaded "module-always-sink" (index: #11; argument: "").
I: module.c: Loaded "module-suspend-on-idle" (index: #12; argument: "").
I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1"
I: module.c: Loaded "module-console-kit" (index: #13; argument: "").
I: module.c: Loaded "module-position-event-sounds" (index: #14; argument: "").
I: module.c: Loaded "module-cork-music-on-phone" (index: #15; argument: "").
I: sink.c: Created sink 1 "ladspa_out" with sample spec float32le 2ch 44100Hz and channel map front-left,front-right
I: sink.c: device.description = "LADSPA Plugin mbeq on HDA NVidia"
I: sink.c: device.master_device = "alsa_output.pci_10de_774_sound_card_0"
I: sink.c: device.class = "filter"
I: sink.c: device.ladspa.module = "mbeq_1197"
I: sink.c: device.ladspa.label = "mbeq"
I: sink.c: device.ladspa.name = "Multiband EQ"
I: sink.c: device.ladspa.maker = "Steve Harris <[email protected]>"
I: sink.c: device.ladspa.copyright = "GPL"
I: sink.c: device.ladspa.unique_id = "1197"
I: sink.c: device.icon_name = "audio-card"
I: source.c: Created source 2 "ladspa_out.monitor" with sample spec float32le 2ch 44100Hz and channel map front-left,front-right
I: source.c: device.description = "Monitor of LADSPA Plugin mbeq on HDA NVidia"
I: source.c: device.class = "monitor"
I: source.c: device.icon_name = "audio-input-microphone"
I: resampler.c: Forcing resampler 'copy', because of fixed, identical sample rates.
I: resampler.c: Using resampler 'copy'
I: resampler.c: Using float32le as working format.
I: sink-input.c: Created input 0 "LADSPA Stream" on alsa_output.pci_10de_774_sound_card_0 with sample spec float32le 2ch 44100Hz and channel map front-left,front-right
E: sink-input.c: Assertion 'PA_SINK_INPUT_IS_LINKED(i->thread_info.state)' failed at pulsecore/sink-input.c:807, function pa_sink_input_update_max_request(). Aborting.
zsh: abort pulseaudio -v
I wonder if someone could help me...
I try to install pulseaudio-git from aur because a solution proposed (in the seconde link) is to simply remove the line 807 from the sink_input.c file but compilation fail.
I use a 64bits architecture.
Last edited by akira86 (2009-05-31 00:29:27)

very interesting post! thanks a lot for sharing it with us..
commission de surendettement - commission de surendettement, vous pouvez demander un dossier de surendettement.

Static when turning up or down volume on Klipsch 2.1

just hooked up my "Klipsch 2.1" speakers on my iMac (late '13). I used these on my mid '11 mac mini with no problem. but so far I've tried listening to K-Love radio and I'm getting static when I turn the volume up or down. I've tried turning down the volume down in sound preferences to compensate. Is there anything else I can do to eliminate this? I didn't have the problem using "Bose companion 2" speakers prior to hooking up the Klipsch 2.1. Is there an app that might help with this? Some kind of equalizer or something? Or any other equalizer within Yosemite?

System Preferences > Mission Control > uncheck "Group windows by application"

DB_BUFFER_CACHE is static or not in Oracle10g v.2....?????

Hi ,
I would like you to confirm if the database's buffer size area in SGA is static or dynamic according to the system needs....
I have somehow been blocked as :
SQL> select * from v$sga;
NAME                      VALUE
Fixed Size              1248600
Variable Size         125829800
Database Buffers 159383552
Redo Buffers            7139328In the above sql statement the database buffers are not included in the variable size of the sga.... so can it be assumed that it is a static component ..????
The above values are confirmed (they cannot be otherwise....) in the results of the following statement:
SQL> select * from v$sgainfo;
NAME                                  BYTES RES
Fixed SGA Size                      1248600 No
Redo Buffers                        7139328 No
Buffer Cache Size 159383552 Yes
Shared Pool Size                  109051904 Yes
Large Pool Size                     4194304 Yes
Java Pool Size                     12582912 Yes
Streams Pool Size                         0 Yes
Granule Size                        4194304 No
Maximum SGA Size                  293601280 No
Startup overhead in Shared Pool    37748736 No
Free SGA Memory Available                 0However , there is a table in the Oracle Admininistation e-book which titled "Automatically Sized SGA Components and Corresponding Parameters" and includes , among others , the Buffer Cache component......!!!!!
So , what is true......?????
They seem to be one contrary to the other....!!!!!
Thanks , a lot
Simon

Hi ,
I have just been blocked because in the followng sql statement:
SQL> select * from v$sga;
NAME                      VALUE
Fixed Size              1248600
Variable Size         130024104
Database Buffers      155189248
Redo Buffers            7139328The database buffers is not included in the variable size but in an individual section.....
However , as you have pointed out right executing the statement:
SQL> select * from v$sga_dynamic_components
2 /
COMPONENT                                                        CURRENT_SIZE   MIN_SIZE   MAX_SIZE USER_SPECIFIED_SIZE OPER_COUNT LAST_OPER_TYPE LAST_OPER_MODE LAST_OPER_TIME GRANULE_SIZE
shared pool                                                         113246208 109051904          0                   0          1 GROW           IMMEDIATE      26/03/2007 18:      4194304
large pool                                                            4194304    4194304          0                   0          0 STATIC                                            4194304
java pool                                                            12582912   12582912          0                   0          0 STATIC                                            4194304
streams pool                                                                0          0          0                   0          0 STATIC                                            4194304
DEFAULT buffer cache                                                155189248 155189248          0                   0          1 SHRINK         IMMEDIATE      26/03/2007 18:      4194304
KEEP buffer cache                                                           0          0          0                   0          0 STATIC                                            4194304
RECYCLE buffer cache                                                        0          0          0                   0          0 STATIC                                            4194304
DEFAULT 2K buffer cache                                                     0          0          0                   0          0 STATIC                                            4194304
DEFAULT 4K buffer cache                                                     0          0          0                   0          0 STATIC                                            4194304
DEFAULT 8K buffer cache                                                     0          0          0                   0          0 STATIC                                            4194304
DEFAULT 16K buffer cache                                                    0          0          0                   0          0 STATIC                                            4194304
DEFAULT 32K buffer cache                                                    0          0          0                   0          0 STATIC                                            4194304
ASM Buffer Cache                                                            0          0          0                   0          0 STATIC                                            4194304The database buffers is a dynamic parameter.........if of course the automatic shared memory management is active.....
Thanks,
Simon

What is the best iTunes Equalizer settings for Bose QC15 Headphones?

What is the best iTunes Equalizer settings for the Bose QC15 Headphones? An image of those settings would be great.

I'm a bit of an audio geek, so I can tell you that YES whatever sounds right is what you want, but NO those settings are not very good. The general setting is "OK" but the values are extremely out of range. Take that setting and reduce the distance between the sliders and you're getting there. Look for example at how the other iTunes presets are made, the don't go that far from the horizontal middle, and for good reason.
When you play a recorded song, the song has usually been put through an extensive process of mixing, EQ and mastering before you got it, in order to get the best balance of all frequencies, to ensure the best sound accross a wide range of speakers and headphones, and to ensure the most important details and instruments are audible when played together.
All speakers are different, and that's where the idea of an EQ for the listener is introduced. The idea is to tweak the overall sound from your music player (iTunes) to compensate for the strengths and weaknesses of your speakers. Which to be honest, is a bit much to expect form a typical music consumer.
The iTunes EQ has a set of horizontal lines representing 3dB (volume steps) each. In my experience you should not boost or cut more than 6 db at the most - and that's a lot! I stay at around 0-4dB. The extreme options are there to cut noise from bad recordings or emphasize a detail you can't hear very well.
So here's basic EQ lesson:
The EQ represents the range frequencies in sound that the human ear can hear. Actually they go beyond that - most people cannot hear the far left and right frequencies. The range of frequencis start at "deep" or "low" frequencies at the left (bass) passing through the Midrange (midtones) in the center, to the (treble) at the right.
Your speaker has a limit to how many frequencies it can play at once before it starts to lose detail - meaning one sound will drown out in order to make room for another. So don't over-EQ your sound or it'll ruin it.
How to tweak an EQ:
1) Start by playing a well-produced piece of modern music - it'll most likely be well mixed and employ a wide range of frequencies. I'm into heavy metal but for this I'd probably go with some average pop music, because metal is so noisy it gets hard to pick out the details.
2) Set you EQ to FLAT - zero all accross the board. It'll sound pretty flat and boring unless your speakers are very "flavoured" or "colored" already.
Sliding any of the EQ bar up, will increase or boost the volume of the frequency the bar represents. You can't really add frequencies that aren't there already - you're just adding noise in that freqeuncy and that will "fill up" the available frequencies on your speaker, causing that "drowned out" effect I was talking about.
Instead, slide them down from the center, and you're cutting the volume instead. This is better because it'll leave more "room" for the other frequencies. You can always use the Preamp slider at the end if you're worried about loudness (but I wouldn't).
3) Start by tweaking the 500 and 1K slider. This represents the middle of your ears' range of hearing and the most noticable change. It should most likely go downwards - this will take the "canned" sound out of your music and that'll be apparent quickly. Go ahead and slide that down as far as -6dB.
4) Now work outwards from the center forming a "v" shape, where the top tips of the V end at the 64 slider and the 8k slider. The very far left and right sliders (32 and 16K) should dip down again to about 0bB. Trust me, these are almost non-important in most music. You can tweak thos last if you feel it's needed.
5) now it's time to start playing round with the V shape. Go ahead and try sliding the sliders a little up or down between 125-250 and 1K to 4K. You may find that some need to move a lot out of the average "V" shape before the sound is good. This is what my iTunes EQ settings ended up looking like for my Motörheadphones "Motorizer" headphones:
Hope it helps.

Windows 7 Audigy 2 Value Equalizer?

I've had a look at the Creative support website, and installed the latest drivers. This works perfectly fine however, I only have "ALChemy" and "Audio Console" applications available.
Is there some way I can install the Equalizer application for Audigy 2 Value on Windows 7?
Thanks.

&Re: Windows 7 - Audigy 2 ZS Mic Issues? The support pack does absolutely nothing to fix the mic issues.
Does this driver improve sound quality or performance? Are there any tweaks?
No, this driver does not improve sound quality nor increase performance.
No, there are no tweaks.
If someone states that its driver improves sound quality and performance, you are being fooled, cheated.
It is even worse that the so called tweaks come from a person who is not EVEN technically acknowledged, does NOT know anything about Windows internals, software engineering and?sound?fidelity that all audiophiles always look for.
If you "feel" the sound better, that is the infamous placebo effect and there is no real proof that any tweaks work.
In fact, I have yet to see a driver that states in its release notes that it improves sound quality.
I've disassembled about every single .SYS driver, .DLL libraries and .EXE executables and found nothing.
There are some undocumented registry values, but nothing related to sound quality or performance.
Would the so called tweaks even exist, they would be in my Support Packs, you can bet that.
and
Microphone issues on Windows 7/Vista 64-bit (x64) with 4GB?of RAM or more
Audigy driver does not support 64-bit addressing, resulting in random issues with the Microphone (static, robotic or distorted voice, white noise).
Workarounds:
- install the 32-bit (x86) version of Windows
- limit the RAM available to Windows to less than 4GB
- enable Memore Hole in BIOS Setup, if?your motherboard supports the setting

CD-R has static/distortion

I burned a CD-R in iTunes v.10.1 but there is static/distortion toward the end of the CD-R. This is a first. Any suggestions as to what went wrong? Thanks so much.

n m u wrote:
Thanks so much. After posting my question I found a Forum suggestion to undo any iTunes Equalizer settings, so I did; and also lower the burn speed, so I burned a new CD-R at 6x. It seems to have worked.
Glad it worked.
If I use iTunes and iVolume to adjust the volume levels before I burn a Playlist, would listening with headphones be the best way to make sure I am satisfied with the volume of each song in that particular Playlist?
If the choice is computer speakers or high quality headphones, you will hear better over the headphones. iVolume is quite good at equalizing, but for best quality you should listen and evaluate.

Light static on DAT tape

I recently shot a band and used a DAT recording from the soundboard. during soundcheck i set my levels, recorded a bit and everything sounded perfect. during the bands set, the soundguy did something and now i'm getting a light static sound, but only on the highs...like feedback and some on vocals. i tried using the equalizer and it helped a bit, but it's still kinda noticable. is there anyway i can reduce the static or eliminate it without harming the quality of the recording. i use a sony tcd-d8 DAT recorder and my levels were set at about 10db or so. i'm still kinda new to recording live sound like that so any suggestions are greatly appreciated

You will probably want to experiment with a noise gate filter - but if you want to preserve the quality of the recording, you'll have to go easy on it or else your audio will sound muddled. You may only be able to remove the worst of it.
I've had some success using the Emagic filter that comes with Sountrack for noise gate. It takes some trial and error, and I suggest loading the Wikipedia site while you mess with it, so you can look up some of the terminology Anyway, click the little asterisks-looking button on the upper far right of the track properties area, above the volume value entry field and to the left of the reveal triangle for sound envelopes . That will bring up the filter window where you can add the Noise Gate.
There may be third party filters out there that are mean for this specifically. But the key is experimenting, because everybody's problem file is going to be different.

I hear a static(garbled )sound on the left speaker when i use iTune?

This started only when i updated to OS.X version 10.9.I tis a STATIC sound that becomes more evident at higher volume levels!
I have tried:
1) i have tried the equalizer solution.problem still there.
2) tried the Repair Disk permissions route.problem still there.
3) deleting the system preference(.plist files).problem still there.
HELP please.I had a perfect itunes before this upgrade w Mavericks.

MusicByLaurie wrote:
how can I pan the left side to play through both speakers (in mono)?
you can not, you'll need to create a mono version of the left channel:
http://www.bulletsandbones.com/GB/GBFAQ.html#stereotomono
(Let the page FULLY load. The link to your answer is at the top of your screen)

Problem with Threads and a static variable

I have a problem with the code below. I am yet to make sure that I understand the problem. Correct me if I am wrong please.
Code functionality:
A timer calls SetState every second. It sets the state and sets boolean variable "changed" to true. Then notifies a main process thread to check if the state changed to send a message.
The problem as far I understand is:
Assume the timer Thread calls SetState twice before the main process Thread runs. As a result, "changed" is set to true twice. However, since the main process is blocked twice during the two calls to SetState, when it runs it would have the two SetState timer threads blocked on its synchronized body. It will pass the first one, send the message and set "changed" to false since it was true. Now, it will pass the second thread, but here is the problem, "changed" is already set to false. As a result, it won't send the message even though it is supposed to.
Would you please let me know if my understanding is correct? If so, what would you propose to resolve the problem? Should I call wait some other or should I notify in a different way?
Thanks,
B.D.
Code:
private static volatile boolean bChanged = false;
private static Thread objMainProcess;
   protected static void Init(){
        objMainProcess = new Thread() {
            public void run() {
                while( objMainProcess == Thread.currentThread() ) {
                   GetState();
        objMainProcess.setDaemon( true );
        objMainProcess.start();
    public static void initStatusTimer(){
        if(objTimer == null)
             objTimer = new javax.swing.Timer( 1000, new java.awt.event.ActionListener(){
                public void actionPerformed( java.awt.event.ActionEvent evt){
                          SetState();
    private static void SetState(){
        if( objMainProcess == null ) return;
        synchronized( objMainProcess ) {
            bChanged = true;
            try{
                objMainProcess.notify();
            }catch( IllegalMonitorStateException e ) {}
    private static boolean GetState() {
        if( objMainProcess == null ) return false;
        synchronized( objMainProcess ) {
            if( bChanged) {
                SendMessage();
                bChanged = false;
                return true;
            try {
                objMainProcess.wait();
            }catch( InterruptedException e ) {}
            return false;
    }

Thanks DrClap for your reply. Everything you said is right. It is not easy to make them alternate since SetState() could be called from different places where the state could be anything else but a status message. Like a GREETING message for example. It is a handshaking message but not a status message.
Again as you said, There is a reason I can't call sendMessage() inside setState().
The only way I was able to do it is by having a counter of the number of notifies that have been called. Every time notify() is called a counter is incremented. Now instead of just checking if "changed" flag is true, I also check if notify counter is greater than zero. If both true, I send the message. If "changed" flag is false, I check again if the notify counter is greater than zero, I send the message. This way it works, but it is kind of a patch than a good design fix. I am yet to find a good solution.
Thanks,
B.D.

Can I use same IP to static to internal server from external dedicated ip?

M question is would it cause my problems if I had the outside interface external ip mapped to vlan2 with the same ip address as I am trying to create static route to my server? Or is this OK? Keep in mind these are imaginary ips.
I was trying to use this command but it was not letting me:
static (inside,outside) tcp 99.89.69.333 3389 192.168.6.2 3389 netmask 255.255.255.255
Here is the front end of my config
ASA Version 7.2(4)
hostname ciscoasa
domain-name default.domain.invalid
enable password DowJbZ7jrm5Nkm5B encrypted
passwd 2KFQnbNIdI.2KYOU encrypted
names
interface Vlan1
nameif inside
security-level 100
ip address 192.168.6.254 255.255.255.0
interface Vlan2
nameif outside
security-level 0
ip address 99.89.69.333 255.255.255.248
interface Ethernet0/0
switchport access vlan 2
interface Ethernet0/1

Since you are using the outside interface ip address in your static PAT, you should use the keyword "interface" as follows:
static (inside,outside) tcp interface 3389 192.168.6.2 3389 netmask 255.255.255.255

Setting Equalizer for more than one song at a time

Is there any way to set the equalizer setting for more than one song at a time. For example, if I have a classical music album with 10 songs, can I set the equalizer for "classical" for all 10 songs at once? It's a pain to have to do it for each one individually. I tried using the "shift" key, but to no avail.
Thanks,
Joe The Author

highlight all the songs you want then right click and go to get info. in the options tab at the top there is an equalizer preset option.

Compilation error while calling static method from another class

Hi,
I am new to Java Programming. I have written two class files Dummy1 and Dummy2.java in the same package Test.
In Dummy1.java I have declared a static final variable and a static method as you can see it below.
package Test;
import java.io.*;
public class Dummy1
public static final int var1= 10;
public static int varDisp(int var2)
return(var1+var2);
This is program is compiling fine.
I have called the static method varDisp from the class Dummy2 and it is as follows
package Test;
import java.io.*;
public class Dummy2
public int var3=15;
public int test=0;
test+=Dummy1.varDisp(var3);
and when i compile Dummy2.java, there is a compilation error <identifier > expected.
Please help me in this program.

public class Dummy2
public int var3=15;
public int test=0;
test+=Dummy1.varDisp(var3);
}test+=Dummy1.varDisplay(var3);
must be in a method, it cannot just be out somewhere in the class!

Static at 16k on equalizer?

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