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  • Microphone crackling in Ubuntu 12.04

    - by Tomas
    I am not very experienced in linux and have this problem with microphone which works fine on Windows. Microphone is external one and when listening to recorded sound (both Sound Recorder and Skype Echo Test) you can hear crackling noise. I fixed output crackling by replacing load-module module-hal-detect with load-module module-hal-detect tsched=0 in /etc/pulse/default.pa but I have no idea how to fix input. Hardware info: Card: HDA ATI SB Chip: Realtek ALC272X Thanks for help!

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  • Volume control doesn't work for headphones

    - by Hendekagon
    Fresh install of 11.10 and the gnome panel's volume control doesn't work for headphones but does work for speakers on my laptop. In sound settings I have Internal Audio Analogue Stereo selected for output (Output tab) and Analogue Speakers selected for Connector. When I plug headphones in, the volume control doesn't work and the volume is maximum through the headphones. On a side note, only one of my two headphone sockets produces sound too (whereas both worked in 10.04)

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  • Audio not working for Gnome Screencast Ctrl + Alt + Shift + R

    - by Costa
    I can do screencasts by pressing ctrl+alt+shift+r, but I get no sound when I view the videos, I've check my built in mic and headset, they both work on skype and such, I just can't record sound with the gnome built in screencasting. Also, when I open the videos in movieplayer they are in mute by default and there's nothing the the preferences I can find to change that. Any help would be awesome!

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  • Why doesn't Python's `re.split()` split on zero-length matches?

    - by Tim Pietzcker
    One particular quirk of the (otherwise quite powerful) re module in Python is that re.split() will never split a string on a zero-length match, for example if I want to split a string along word boundaries: >>> re.split(r"\s+|\b", "Split along words, preserve punctuation!") ['Split', 'along', 'words,', 'preserve', 'punctuation!'] instead of ['', 'Split', 'along', 'words', ',', 'preserve', 'punctuation', '!'] Why does it have this limitation? Is it by design? Do other regex flavors behave like this?

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  • How to disable Power off/Power on sound on Android phone? [closed]

    - by yvolk
    My Android phone (Android v.2.1) produces loud sound both before Power off and after Power on (boot) is being completed. I don't want to turn the sound off before reboot/powering it off and to turn sound on after reboot every time I'm restarting my phone. I just want to get rid of these sounds forever :-) How can I do this (maybe executing some script, changing some properties... preferably without rooting my phone)? Update: Thank you Christopher Orr! I've also found some information that partially allows to achieve what I'm looking for (on androidforums.com): "if you are going into SettingsSound & displayNotification ringtone that sets your notification tone for app updates, power on (rebooting) etc. If you want to change notification for messages you would have to go into messaging and then hit the menu key for notification option. You can do the same for email accounts as well." So I disabled Power on/off sounds (plus something else... but I didn't notice undesired side effect yet...) with built-in settings.

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  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

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  • Excel or Access: how to group several lines in a table and insert contents in columns? ("split column")

    - by Martin
    I have a table containing data of sold products (shown in the example on the left): Columns: Number of the order Product Name Attribute - specifies what is given in the following field "value", e. g. Customer Name or Product Variant Value - is the value of the Attribute Count - is the number of products of this variant sold in the order That means: Product B has 2 variants "c" and "d" Note that in Order 1 Product B was sold in Variant d only, because the letter "N" in field "D4" means "none". Note, that in OrdnerNo 3 Product B was sold only in Variant c, because for Variant d field "D9" is "N"!! This is confusing, but it is the structure of the original data (which I can not change). I need a way to convert the table on the left in a table like that on the right: one line for each product type Order Number Product Name Customer Name Count (number of products sold in this order) Variant - this is the problem, as it has to be filled with the So all rows with the same OrderNo and same product have to be grouped in to one, and I hope it is clear what I need. I tried to do it with Pivot Tables, but that fails, as the Count is always in each line, no matter if it has Value "N" or not and for the products without variants there is only one line for each order, however for products with variants there are several... So how could I create the right table with a VBA macro in MS Excel or maybe there is a trick in MS Access to do it directly or with an SQL query?

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  • Deletes that Split Pages and Forwarded Ghosts

    - by Paul White
    Can DELETE operations cause pages to split? Yes. It sounds counter-intuitive on the face of it; deleting rows frees up space on a page, and page splitting occurs when a page needs additional space. Nevertheless, there are circumstances when deleting rows causes them to expand before they can be deleted. The mechanism at work here is row versioning (extract from Books Online below): Isolation Levels The relationship between row-versioning isolation levels (the first bullet point) and page splits is...(read more)

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  • How to Enable or Disable the Startup Sound in Windows 8

    - by Taylor Gibb
    We have all been in that awkward situation where you sit down in a meeting room only to have your Windows laptop play the Startup sound. Here’s how to disable it or even enable it if you so choose. HTG Explains: What is the Windows Page File and Should You Disable It? How To Get a Better Wireless Signal and Reduce Wireless Network Interference How To Troubleshoot Internet Connection Problems

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  • YouTube, no video or sound

    - by Cautious1
    I tried the answers from previous posts without luck. I'm using ubuntu 10.04.4 and youtube shows a black screen, no video and no sound . I uninstalled adobe flash player closed down and reinstalled but it didn’t help. I have run Mint 13 on the same computer and it will play youtube without a problem. I'm not familier with linux language . Using comands in terminal might make everything terminal if I try!

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  • Split Body and Sprite

    - by Siddharth
    I want to split the body and sprite into multiple pieces as like following link suggest: http://www.raywenderlich.com/14302/how-to-make-a-game-like-fruit-ninja-with-box2d-and-cocos2d-part-1 I try to convert the cocos2d program into AndEngine and almost done but they use PRKit named external library that type of implementation I didn't find in AndEngine. So my basic requirement you know so please suggest me some suitable answer. My current working platform is AndEngine with Box2D.

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  • No USB 04b4:0307 mike input after 10.04

    - by Papou
    I am using an USB phone that is in fact a so-called "sound card" based on the 04b4:0307 chip. In fact, I have two different phones using 04b4:0307 and in fact I have a sound USB key too. This, I believe, is the start of why they call 04b4:0307 "ubiquitous" (instead of oh four bee...). But not "eternal". The mike worked in Ubuntu 9.10 and 10.04 but not later ([email protected]). "not working" means that 04b4:0307 shows in Sound Preference but that its vu-meter is mute. I have posted the full lsusb and the result of tests in various systems here: http://www.papou.byethost9.com/tmp/1043601.html Note: Tests done on VirtualBox (thankfully). But UbuntUnity no longer works on VB, so I used the LinuxMint equivalent. ?hat's fate. I could not find any problem report close enough. What should be my next step? I believe the problem occurs in module snd-usb-audio. One thing I might try if I knew how is hacking a DEB with its 10.04's source. I can hack DEBs. Any hint welcome on how to make a DEB overriding a kernel packed module. I mean that the newly installed module should have precedence, be loaded instead, the module installed with the kernel. TIA !

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  • Restore audio settings - cannot open mixer: No such file or directory

    - by Alfred M.
    The internal speaker of my laptop never functionned under Ubuntu. I tried to follow indication on the web and now the jack audio does not work either. The graphic interface for audio management now displays a 'dummy output' instead of the three possible outputs I used to have (one of them was working for the jack output). In a terminal alsamixer raises an error: cannot open mixer: No such file or directory I did try to remove and reinstall alsa-utils but it did not change anything. This happened after a failed atempt to install alsa-driver-linuxant_1.0.23.1_all.deb from here. My sound card seems to be not recognised anymore. After reboot I have no more the sound icon in menu bar the upper right corner. I think I have removed my sound card driver. Indeed, the command sudo lshw -class multimedia indicated audi device as unclaimed. Any idea how I could revert to a better situation (that is jack support and alsa working)? EDIT: The command lspci -nnk | grep -iEA3 audio gives lspci -nnk | grep -iEA3 audio 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 03) Subsystem: ASUSTeK Computer Inc. Device [1043:1893] 00:1c.0 PCI bridge [0604]: Intel Corporation 82801I (ICH9 Family) PCI Express Port 1 [8086:2940] (rev 03)

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  • Audio comes out of both headphone and speaker at the same time.. Ubuntu 12.04LTS [closed]

    - by pst007x
    I have the same issue on an Aspire. Ubuntu 12.04LTS 64bit realtek audio sound chip onboard If I plug in a headset, audio does not switch from internal speaker to headset, instead plays out of both at the same time. I have looked at the alsamixer setting, all on. I installed gnome-alsamixer, and I noticed headphone was ticked, if I untick the main audio mutes, and the headphone no longer works. Headset only works with internal speaker. Audio works fine on my other desktop and laptop running this release 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) salvatore@salvatore-Aspire-7730:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. salvatore@salvatore-Aspire-7730:~$ head -n 1 /proc/asound/card*/codec#* ==> /proc/asound/card0/codec#0 <== Codec: Realtek ALC888 ==> /proc/asound/card0/codec#1 <== Codec: LSI ID 1040 ==> /proc/asound/card0/codec#2 <== Codec: Intel Cantiga HDMI salvatore@salvatore-Aspire-7730:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC888 Analog [ALC888 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC888 Digital [ALC888 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 salvatore@salvatore-Aspire-7730:~$ uname -a Linux salvatore-Aspire-7730 3.2.0-23-generic #36-Ubuntu SMP Tue Apr 10 20:39:51 UTC 2012 x86_64 x86_64 x86_64 GNU/Linux salvatore@salvatore-Aspire-7730:~$ The alsa-base.conf does not exist Tried this: sudo apt-get remove --purge alsa-base sudo apt-get remove --purge pulseaudio sudo apt-get install alsa-base sudo apt-get install pulseaudio sudo alsa force-reload Then: sudo apt-get purge pulseaudio gstreamer0.10-pulseaudio sudo apt-get install pulseaudio gstreamer0.10-pulseaudio indicator-sound Tred this. sudo gedit Then open terminal: sudo /etc/modprobe.d/alsa-base.conf At the end of the file add a new line: options snd-hda-intel model=generic Save and then reboot But alsa-base.conf does not exist

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  • Issues in pulse audio in Ubuntu 11.10

    - by Kamal
    Good Morning All. I am a new ubuntu user. So please forgive me if this question is too basic. I have installed Ubuntu 11.10 in my machine. I have logged in as USER_A. My external audio device is a Headset and I was able to hear the audio properly. I need to join my Ubuntu machine to a window's domain (my office server). I followed the steps explained in http://www.ghacks.net/2010/04/21/join-a-ubuntu-machine-to-a-windows-domain/ and was successful in joining my ubuntu machine to the windows domain. sudo apt-get install likewise-open5 sudo domainjoin-cli join DOMAIN USER_B Now when I logged in as USER_B, there is no audio for this user in the same machine. I crossed check with my User_A account. There is no issues with the sound for User_A. Only for User_B, there is no audio. When I checked the sound settings of User_B, there is no device listed in Hardware, Input and Output. Whereas for User A, my Headset is listed in Input and Output. Can anyone please help me on this. Why there is no sound for User_B? Thank you.

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  • Can't configure 5.1 audio with 12.04

    - by xster
    I have an Intel ALC892 and a Nvidia GT 520m connected to speakers via HDMI. On lspci, I see 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: ZOTAC International (MCO) Ltd. Device a218 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at db400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel 02:00.1 Audio device: NVIDIA Corporation HDMI Audio stub (rev a1) Subsystem: ZOTAC International (MCO) Ltd. Device 2180 Flags: bus master, fast devsel, latency 0, IRQ 18 Memory at db080000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Kernel driver in use: snd_hda_intel My alsamixer looks like I enabled pulseaudio configuration file to have 6 channels. My sound setting looks like When I use the test dialog, only front left and right have sounds. If I use alsa in XBMC on a 5.1 video, there's no sound. If I use pulseaudio, only front right and left have sound. I can barely hear any speech since I'm guessing it's mapped to front center. Any clues?

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  • Why does my mic boost automatically go to 100 on every boot?

    - by Ben
    When my computer turns on, it automatically sets the "mic boost" sound setting to 100. This causes a loud static sound in my speakers. I can manually go to alsamixer and turn the mic boost down manually, but I would prefer it if I didn't have to do this every time I turn the computer on. I've tried running sudo alsactl store after fixing the settings, and this does save them, but I have to run sudo alsactl restore to restore the settings. This means that I have to manually fix the sound every time I start the computer anyway, so it isn't really a fix. I tried putting sudo alsactl restore in my startup programs, but that didn't seem to fix anything. I'm running Ubuntu 12.04, but I started having this problem before upgrading from 11.10. I'm using a Sony Vaio laptop. I'm not really sure what made it start; it seemed like I just started having the problem randomly one day. Any help would be appreciated! Edit: here is the output from running amixer: http://paste.ubuntu.com/1060080/

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  • HP dv7 Beats Audio Subwoofer on while headphones plugged in

    - by msilis
    I have looked for answers to this but have not found anything. People have similar problems but mostly with getting the subwoofer to work at all. I got the subwoofer to work by editing the /etc/modprobe.d/alsa-base.conf file and adding options snd-hda-intel model=ref. When I plug in my headphones, the main speakers turn off but the subwoofer still has sound coming through it. The output is selected automatically in sound preferences and is set to headphones when they are plugged in. I also have tried muting the main speakers, which in turn mutes the subwoofer, before plugging in the headphones as they seem to have two different volume controls, but as soon as the headphones are plugged in, the subwoofer has sound coming out of it. I am running Ubuntu 11.10 64 bit on HP dv7- 4285dx. I am able to just not have the subwoofer on but since I have gotten it to work I would like to keep it around and not have to change a config file every time I want to plug in my headphones. Any ideas or suggestions would be greatly appreciated. Thanks

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  • M-Audio Delta 1010LT on 12.04

    - by user74039
    I have 12.04 64bit installed, my soundcard is a Delta 1010LT, it seems to be partially detected, I've been following steps here: https://help.ubuntu.com/community/SoundTroubleshooting/ lspci -v | grep -A7 -i "audio" shows this: 04:07.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. M-Audio Delta 1010LT Flags: bus master, medium devsel, latency 64, IRQ 22 I/O ports at ec00 [size=32] I/O ports at e880 [size=16] I/O ports at e800 [size=16] I/O ports at e480 [size=64] Capabilities: <access denied> Kernel driver in use: snd_ice1712 aplay shows this: **** List of PLAYBACK Hardware Devices **** card 0: M1010LT [M Audio Delta 1010LT], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 In the sound settings on the desktop all I see is the ICE1712 S/PDIF, which I don't use, I want to use the individual outputs on the card, I'm not so bothered about inputs, I just want the playback for now. If I open alsamixer in the console, I see all of the output and input channels, i've raised the volume on them but I don't get anything in the sound settings on the desktop and when I play any sound, I hear nothing. Can someone help?

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  • PulpCore music playback - loop sound and animate volume

    - by Peter Perhác
    I have been experimenting with PulpCore, trying to create my own tower defence game (not-playable yet), and I am enjoying it very much I ran into a problem that I can't quite figure out. I extended PulpCore with the JOrbis thing to allow OGG files to be played. Works fine. However, pulpCore seems to have a problem with looping the sound WHILE animating the volume level. I tried this with wav file too, to make sure it isn't jOrbis that breaks it. The code is like this: Sound bgMusic = Sound.load("music/music.ogg"); Playback musicPlayback; ... musicVolume = new Fixed(0.75); musicPlayback = bgMusic.loop(musicVolume); //TODO figure out why it's NOT looping when volume is animated // musicVolume.animate(0, musicVolume.get(), FADE_IN_TIME); This code, for as long as the last line is commented out, plays the music.ogg again and again in an endless loop (which I can stop by calling stop on the Playback object returned from loop(). However, I would like the music to fade in smoothly, so following the advice of the PulpCore API docs, I added the last line which will create the fade-in but the music will only play once and then stop. I wonder why is that? Here is a bit of the documentation: Playback pulpcore.sound.Sound.loop(Fixed level) Loops this sound clip with the specified volume level (0.0 to 1.0). The level may have a property animation attached. Parameters: level Returns: a Playback object for this unique sound playback (one Sound can have many simultaneous Playback objects) or null if the sound could not be played. So what could be the problem? I repeat, with the last line, the sound fades in but doesn't loop, without it it loops but starts with the specified 0.75 volume level. Why can't I animate the volume of the looped music playback? What am I doing wrong? Anyone has any experience with pulpCore and has come across this problem? Anyone could please download PulpCore and try to loop music which fades-in (out)? note: I need to keep a reference to the Playback object returned so I can kill music later.

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  • Sound Complete Not Firing (AS3)

    - by JasonMc92
    I have a bit of a quandary. I need to call a function inside a MovieClip once a particular sound has finished playing. The sound is played via a sound channel in an external class which I have imported. Playback is working perfectly. Here is the relevent code from my external class, Sonus. public var SFXPRChannel:SoundChannel = new SoundChannel; var SFXPRfishbeg:Sound = new sfxpr_fishbeg(); var SFXPRfishmid:Sound = new sfxpr_fishmid(); var SFXPRfishend3:Sound = new sfxpr_fishend3(); var SFXPRfishend4:Sound = new sfxpr_fishend4() public function PlayPrompt(promptname:String):void { var sound:String = "SFXPR" + promptname; SFXPRChannel = this[sound].play(); } This is called via an import in the document class "osr", thus I access it in my project via "osr.Sonus.---" In my project, I have the following line of code. osr.Sonus.SFXPRChannel.addEventListener(Event.SOUND_COMPLETE, promptIsFinished); function prompt():void { var level = osr.Gradua.Fetch("fish", "arr_con_level"); Wait(true); switch(level) { case 1: osr.Sonus.PlayPrompt("fishbeg"); break; case 2: osr.Sonus.PlayPrompt("fishmid"); break; case 3: osr.Sonus.PlayPrompt("fishend3"); break; case 4: osr.Sonus.PlayPrompt("fishend4"); break; } } function Wait(yesno):void { gui.Wait(yesno); } function promptIsFinished(evt:Event):void { Wait(false); } osr.Sonus.PlayPrompt(...) and gui.Wait(...) both work perfectly, as I use them in other contexts in this part of the project without error. Basically, after the sound finishes playing, I need Wait(false); to be called, but the event listener does not appear to be "hearing" the SOUND_COMPLETE event. Did I make a mistake somewhere? For the record, due to my project structure, I cannot call the appropriate Wait(...) function from within Sonus. Help?

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