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  • Audio Line-In on Ubuntu/Linux Mint

    - by hahuang65
    I'm currently on Windows, and want to switch to Linux, but some hardware issues are preventing me. Mainly, I have a sound card that supports Line-In. On Windows, anything I plug into the line-in gets outputted to the speakers. However, when I installed Linux, because there is not a control application that comes with the driver, I have no idea how to set this up. I tried going to the sound settings and it doesn't seem to be there. I also want to configure it for 2.1 sound, and do not know how to do that... Anyone here done it before? Thanks in advance for the help!

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  • 1TB HDD making strange noise (not a common one)

    - by Darkkurama
    I built a new PC some days ago, and everything seems perfect, except that the 1 TB HDD I cloned from my old 500 GB HDD is making a deep weird sound. First of all, every time I access the disk, I hear a deep sound, and when the PC is turning on, I hear some clicking (the rapid clicking is my mouse, I'm opening and closing folders to trigger the vibrating deep weird sound I'm describing). I'm using this 1TB disk for data mainly (I use a SSD as the OS). As background information, the disk is a seagate barracuda 7200 rpm which was RMAd and replaced with a refurbished one. Maybe the refurbished disks make these noises? should I worry about my data? (although the disk is working normal and passed a seagatetools short generic test? Thanks! PS: I recorded the sounds, just click on the links. Thanks

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  • Can it be harmful to grant jackd realtime priority?

    - by SuperElectric
    I am apt-get installing Ardour, a sound mixing program, just to try it out. Installing Ardour also installs JACK, a dependency. As part of the JACK installation script, I get the following dialog: If you want to run jackd with realtime priorities, the user starting jackd needs realtime permissions. Accept this option to create the file /etc/security/limits.d/audio.conf, granting realtime priority and memlock privileges to the audio group. Running jackd with realtime priority minimizes latency, but may lead to complete system lock-ups by requesting all the available physical system memory, which is unacceptable in multi-user environments. Enable realtime process priority? I'm installing on my laptop, which never has multiple simultaneous users. I still have concerns: is JACK something that'll be used by the system itself to play any sound (i.e. will it replace ALSA)? If so, does that mean that if I enable realtime priority for JACK, I'll run a slight risk of freezing the machine whenever any sound is played? Or is JACK only going to be used by Ardour for now (until I install some other JACK-dependent program)? Thanks, -- Matt

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  • External microphone not working

    - by haireefairee
    gnome-volume-control does not recognise external hardware. My headphones work nonetheless, but an external microphone does not. External microphones used to work, but at times were temperamental - I would have to login or logout with or without microphone plugged in. I am running Ubuntu 10.04 LTS (Lucid Lynx) on an mSi U100 wind notebook with one Intel soundcard and trying to use a jack microphone which has worked previously. USB microphones have also been problematic. I have done the basics: Installed upgrades. Checked nothing is muted. Looked for the device on gnome-volume-control. Tried using a different microphone that works on a friends computer. Tested my microphone works when using a different computer. Checked my soundcard can be seen (cat /proc/asound/cards). I have done more complicated things: I have tried playing around with settings in alsamixer. Nothing is muted. I can adjust "mic" and "internal mic" regardless of whether an external microphone is plugged in. I have the choice of input source from "mic", "front mic", "line" and "CD". I've played around changing this and it hasn't helped. I only have one CAPTURE option. In gnome-sound-recorder I have the choice of line, microphone 1 and microphone 2. I have played around changing this option. None of these pick up sound from the external microphone. Microphone 2 is the microphone on my laptop which is bad quality. In gnome-sound-recorder I have the choice of different profiles, and changing this has not helped either. I have looked at gstreamer-properties but none of that seemed helpful. I don't know if there a way to check if these external devices are being picked up. I would like to make an external microphone work. Please help!

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  • Bluetooth Issues Ubuntu 13.10

    - by Eduardo
    I have a bluetooth headset which works perfectly on Ubuntu 13.04. Now I update to 13.10, and here is what's happing: After installing blueman, bluetooth-suport, pulseaudio-module-bluetooth and so on, I can find my device, pair it and connect to the headset service. But the device does not appear on the Sound Settings, so I just can't select it as input/output device. In other words, it's connected but "useless". So, searching around for solutions, I found a software called stream2ip. With this I can connect the device and it appears on the Sound Settings, the sound plays on the device as well, but I microphone does not work, even when selected on the settings, also the A2DP option still not working. Stream2ip isn't a solution at all, I mean everything was working without it in the previous Ubuntu version. Maybe I'm missing something, and I hope someone could give me any hint. And formally, the question: How can I get the A2DP output option and the input working again, on the Ubuntu 13.10? Thanks!

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  • Ubuntu 12.04 taking too much time to boot

    - by adarshdinesh
    Ubuntu 12.04 is taking much time for booting, Here is the system kernel message while booting .It is showing that some anacron was killed ,why ? and how to fix the problem ? [ 2.241047] scsi6 : usb-storage 2-1.6:1.0 [ 2.241501] usbcore: registered new interface driver usb-storage [ 2.241895] USB Mass Storage support registered. [ 3.240670] scsi 6:0:0:0: Direct-Access Multiple Card Reader 1.00 PQ: 0 ANSI: 0 [ 3.241791] sd 6:0:0:0: Attached scsi generic sg2 type 0 [ 3.243083] sd 6:0:0:0: [sdb] Attached SCSI removable disk [ 12.568641] Adding 4037904k swap on /dev/sda3. Priority:-1 extents:1 across:4037904k [ 12.615014] udevd[462]: starting version 175 [ 12.651334] mei: module is from the staging directory, the quality is unknown, you have been warned. [ 12.655283] [drm] Initialized drm 1.1.0 20060810 ................... [ 14.118369] init: alsa-restore main process (982) terminated with status 19 [ 14.252595] init: anacron main process (1033) killed by TERM signal [ 14.285763] HDMI status: Codec=3 Pin=5 Presence_Detect=0 ELD_Valid=0 [ 14.285841] input: HDA Intel PCH HDMI/DP,pcm=3 as /devices/pci0000:00/0000:00:1b.0/sound/card0/input8 [ 14.285925] input: HDA Intel PCH Mic as /devices/pci0000:00/0000:00:1b.0/sound/card0/input9 [ 14.285991] input: HDA Intel PCH Headphone as /devices/pci0000:00/0000:00:1b.0/sound/card0/input10 [ 14.615073] init: plymouth-stop pre-start process (1222) terminated with status 1 [ 16.447287] wlan0: authenticate with c0:8a:de:7c:60:e8 (try 1) [ 16.448858] wlan0: authenticated [ 16.453405] wlan0: associate with c0:8a:de:7c:60:e8 (try 1) [ 16.456392] wlan0: RX AssocResp from c0:8a:de:7c:60:e8 (capab=0x431 status=0 aid=2) [ 16.456398] wlan0: associated [ 16.457014] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 16.457017] ieee80211 phy0: brcmsmac: brcms_ops_bss_info_changed: associated [ 16.457019] ieee80211 phy0: changing basic rates failed: -22 [ 16.457021] ieee80211 phy0: brcms_ops_bss_info_changed: arp filtering: enabled true, count 0 (implement) [ 16.457226] ADDRCONF(NETDEV_CHANGE): wlan0: link becomes ready [ 16.654196] ieee80211 phy0: brcms_ops_bss_info_changed: arp filtering: enabled true, count 1 (implement) [ 17.823565] ieee80211 phy0: wl0: brcms_c_d11hdrs_mac80211: txop exceeded phylen 180/256 dur 1946/1504 [ 18.220865] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 26.881422] wlan0: no IPv6 routers present [ 68.228293] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 73.240133] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 76.574490] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 102.180006] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 103.100984] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement) [ 124.171624] ieee80211 phy0: brcms_ops_bss_info_changed: qos enabled: true (implement)

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  • Guide.BeginShowMessageBox wrapper

    - by Daniel Moth
    While coding for Windows Phone 7 using Silverlight, I was really disappointed with the built-in MessageBox class, so I found an alternative. My disappointment was the fact that: Display of the messagebox causes the phone to vibrate (!) Display of the messagebox causes the phone to make an annoying sound. You can only have "ok" and "cancel" buttons (no other button captions). I was using the messagebox something like this: // Produces unwanted sound and vibration. // ...plus no customization of button captions. if (MessageBox.Show("my message", "my caption", MessageBoxButton.OKCancel) == MessageBoxResult.OK) { // Do something Debug.WriteLine("OK"); } …and wanted to make minimal changes throughout my code to change it to this: // no sound or vibration // ...plus bonus of customizing button captions if (MyMessageBox.Show("my message", "my caption", "ok, got it", "that sucks") == MyMessageBoxResult.Button1) { // Do something Debug.WriteLine("OK"); } It turns out there is a much more powerful class in the XNA framework that delivered on my requirements (and offers even more features that I didn't need like choice of sounds and not blocking the caller): Guide.BeginShowMessageBox. You can use it simply by adding an assembly reference to Microsoft.Xna.Framework.GamerServices. I wrote a little wrapper for my needs and you can find it here (ready to enhance with your needs): MyMessageBox.cs.txt. Comments about this post welcome at the original blog.

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  • Multiple audio sources on a single gameObject in unity

    - by angryInsomniac
    So, I have an audio system set up wherein I have loaded all my audio clips centrally and play them on demand by passing the requesting audioSource into the sound manager. However, there is a complication wherein if I want to overlay multiple looping sounds, I need to have multiple audio sources on an object, which is fine , so I created two in my script instantiated them and played my clips on them and then the world went crazy. For some reason, when I create two audio Sources in an object only the latest one is ever used, even if I explicitly keep objects separated, playing a clip on one or the other plays the clip on the last one that was created, furthermore, either this last one is not created in the right place or somehow messes with the rolloff rules because I can hear it all across my level, havign just one source works fine, but putting a second one on it causes shit to go batshit insane. Does anyone know the reason / solution for this ? Some pseudocode : guardSoundsSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardSoundsSource.name = "Guard_Sounds_source"; // Setup this source guardThrusterSource = (AudioSource)gameObject.AddComponent("AudioSource"); guardThrusterSource.name = "Guard_Thruster_Source"; // setup this source // play using custom Sound manager soundMan.soundMgr.playOnSource(guardSoundsSource,"Guard_Idle_loop" ,true,GameManager.Manager.PlayerType); // this method prints out the name of the source the sound was to be played on and it always shows "Guard_Thruster_Source" even on the "Guard_Idle_loop" even though I clearly told it to use "Guard_Sounds_source"

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  • An unexpected pleasure from Windows 8

    - by eddraper
    This post is certainly on the more nuanced side of all the goodness that is Windows 8, but it’s about something that’s really changed my PC usage experience for the better. Besides being a geek and the enjoying all the techno-thrills and chills that go along with sitting in front of a keyboard all day, I really love the forest.  Trees have always been special to me.  The feeling of being in the forest with all the sounds and ambiance, the broken light, the fragrance of the air… it’s paradise to me. As I can’t get there often, due to work, and quite often the heat here in Texas, I’ve found something that can at least partially fill the gap…  When you install Windows 8, you’ll have an app called “Naturespace” from http://www.naturespace.com/ .  It boasts a number of predefined loops in what they call “holographic audio.”  They’re essentially high-tech 3D sound fields recorded in natural environments. After checking them out, I really liked the sound of the “Daybreak” selection: A great benefit is that you don’t have to be in Metro/Modern/Windows App Store mode, in order to keep the sound playing.  To start the day, I click on Daybreak, start it, then go back to the desktop and fire up VS, Chrome, etc. As I work and play, I’m surrounded by this delightful background ambiance which relaxes me and puts my mind at ease. Give it a try.  I think you’ll like it.  And no, you don’t need ear buds or headphones to get the benefit.

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  • Reproducible freezes with on an AMD fusion (e350) sony vaio

    - by doycho
    So a week ago I bought it and I've been struggling to make the Ubuntu which I installed stable. There's one thing that makes my life miserable, though. There's this easily reproducible freeze when I start some kind of video. So here is what happens: Everything works fine for some time I start vlc/mplayer/flashplayer/totem with something to watch In few minutes time I lose the sound (nothing in the logs at this point) At that time the video app instantly allocates all the memory and its CPU usage skyrockets. Total freeze. I can move the cursor around for few seconds and sometimes even switch to another app. But ultimately there comes the time I can't do anything - can't kill X with ctrl+alt+backspace (I have it enabled), can't switch to any other console (ctrl+alt+f1-6), can't connect to the machine via ssh. The only way to restart it is the ctrl+alt+SysRq+UABI magic :) What discourages me most is the fact I can't see anything in the logs. The only error I've noticed is Jun 19 17:00:37 serenity kernel: [ 1506.350676] software-center[17581]: segfault at 30 ip 00007fd3631b814c sp 00007fff18a6fa10 error 4 in libgtk-x11-2.0.so.0.2400.4[7fd362f7d000+436000]. I've been searching through the Xorg log, kernel logs, syslog. If you have any idea how I can get more debug info I'll be glad to try them. Things I've tried: Changing drivers - the open source one, the proprietary driver xorg-edgers' ppa - https://launchpad.net/~xorg-edgers/+archive/ppa changing to the last stable kernel (2.6.39) Some notes: It my be irrelevant but the sound is constantly stuttering. This probably is a separate issue though I've found that if I start more video/sound apps the freeze happens faster.

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  • How to make pulseaudio and ubuntu detect the same audio device as alsa driver

    - by Kiwy
    I use Ubuntu 14.04 x64 and I use gnome-shell on my laptop. I have a Bose companion 5 (which is basically a USB sound system) and a HDMI port, both does work perfectly when I just boot with the cable plugin. However, when my laptop go to sleep or get unplugged from those two outputs, if I plug back the device, I end up without any hardware detection (only the built-in speakers) from pulse and gnome-shell sound output selector while if I use alsamixer, the device look up and ready. gstreamer-properties allow me to select and test effectively any device but while alsa recognize any device on the run, pulse is not capable of handling things correctly, my question is then: How can I make pulse detect and use the same hardware as alsa, or how to remove completely and gracefully pulseaudio (meaning volume applet running in gnome shell) I don't mind if the project implies to recompile half gnome shell if it implies those audio outputs work all the time. Pulse does not list my soundcard when I use command pactl list cards while the module plug&play for sound card is loaded in pactl list modules. I really don't know what to do, the behavior seems pretty random.

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  • SEHException throw using Microsoft XACT Audio Framework (XACT3)

    - by Sweta Dwivedi
    I have been developing a game using Kinect + XNA and using Microsoft Audio Creation tool (XACT3) for managing my sound files and music, however in the code an SEHException is thrown whenever it tries to get the wave file from the wave Bank . . Sometimes the code works magically and all of a sudden it will start throwing this exception randomly ..I need a help on solving this exception /*Declaring Audio Engine for music*/ AudioEngine engine; SoundBank soundBank; WaveBank waveBank; Cue cue; /*Declaring Audio engine for sound effects*/ AudioEngine engine1; SoundBank soundbank; WaveBank wavebank; Cue effect; engine = new AudioEngine(@"Content\therapy.xgs"); soundBank = new SoundBank(engine, @"Content\Sound Bank.xsb"); **waveBank = new WaveBank(engine, @"Content\Wave Bank.xwb");** cue = null; engine1 = new AudioEngine(@"Content\Music_Manager\Sound_effects.xgs"); soundbank = new SoundBank(engine1, @"Content\Music_Manager\Sound1.xsb"); **wavebank = new WaveBank(engine1, @"Content\Music_Manager\Wave1.xwb");** effect = null; cue = soundBank.GetCue("hypnotizing"); cue.Play();

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  • Dell xps 15z fan issue in ubuntu 12.04

    - by Paxinum
    I just updated to ubuntu 12.04 on my Dell laptop xps 15z. The trouble is that I hear a slight ticking sound every 3rd second, probably from a fan. This is a new issue in this ubuntu version. I use the recommended boot options for grub, i.e. acpi_backlight=vendor, but I do not use any acpi=off or acpi=noirq. Is there a way to fix this issue from ubuntu, by maybe controlling the fans somehow? EDIT: Notice, the sound goes away as the fan speeds up, (when doing calculations or such), so it is really a fan issue. EDIT2: I have located the issue: If I use conky 1.9, together with the command execpi for a python application, then the sound appears, and the noise syncs with the update interval for conky (NOT for the update interval for execpi!). The noise seems to be proportional to the complexity of the drawing that is needed. This is very strange, as this issue was not in the prev. version of conky I used. The solution was to increase the update interval for conky from 0.5 to 3, i.e. update every 3rd second instead of twice a second.

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  • Ubuntu 12.10: Installing proprietary Nvidia driver causes freeze at boot

    - by Greg
    Ok, so I just installed Ubuntu on my laptop, and I immediately encountered an issue: the HDMI audio output won't work. Yes, I know about the sound settings thing where you have to select the HDMI option, but even when it's selected I get no sound out of the TV I'm hooking it up to. This is a dealbreaker for me, because my laptop speakers are terrible, it's one of the big reasons I use my TV monitor. So I decided to work on solving the problem by upgrading my Nvidia drivers. I switched to one of the propriety drivers offered in that software updating utility that comes with the OS, the one option that said (tested). Viola, sound over the HDMI is now working. Unfortunately, this now brings me to my next problem: when I reboot Ubuntu with this or any other proprietary driver installed, it freezes when it tries to load my desktop. As in I can see my wallpaper, but no icons or options of any kind. The system is totally frozen, and gives me one of those "we've experienced an error, do you want to report it messages." So there's my bind. I need HDMI audio out, that's a total dealbreaker for me, but installing the drivers that give me that capability crash the system. Does anyone have any idea what's causing this

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  • ActionScript 2: Event doesn't fire?

    - by Pascal Schuster
    So I have a soundHandler class that's supposed to play sounds and then point back to a function on the timeline when the sound has completed playing. But somehow, only one of the sounds plays when I try it out. EDIT: After that sound plays, nothing happens, even though I have EventHandlers set up that are supposed to do something. Here's the code: import mx.events.EventDispatcher; class soundHandler { private var dispatchEvent:Function; public var addEventListener:Function; public var removeEventListener:Function; var soundToPlay; var soundpath:String; var soundtype:String; var prefix:String; var mcname:String; public function soundHandler(soundpath:String, prefix:String, soundtype:String, mcname:String) { EventDispatcher.initialize(this); _root.createEmptyMovieClip(mcname, 1); this.soundpath = soundpath; this.soundtype = soundtype; this.prefix = prefix; this.mcname = mcname; } function playSound(file, callbackfunc) { _root.soundToPlay = new Sound(_root.mcname); _global.soundCallbackfunc = callbackfunc; _root.soundToPlay.onLoad = function(success:Boolean) { if (success) { _root.soundToPlay.start(); } }; _root.soundToPlay.onSoundComplete = function():Void { trace("Sound Complete: "+this.soundtype+this.prefix+this.file+".mp3"); trace(arguments.caller); dispatchEvent({type:_global.soundCallbackfunc}); trace(this.toString()); trace(this.callbackfunction); }; _root.soundToPlay.loadSound("../sound/"+soundpath+"/"+soundtype+prefix+file+".mp3", true); _root.soundToPlay.stop(); } } Here's the code from the .fla file: var playSounds:soundHandler = new soundHandler("signup", "su", "s", "mcs1"); var file = "000"; playSounds.addEventListener("sixtyseconds", this); playSounds.addEventListener("transition", this); function sixtyseconds() { trace("I am being called! Sixtyseconds"); var phase = 1; var file = random(6); if (file == 0) { file = 1; } if (file<10) { file = "0"+file; } file = phase+file; playSounds.playSound(file, "transition"); } function transition() { trace("this works"); } playSounds.playSound(file, "sixtyseconds"); I'm at a total loss for this one. Have been wasting hours to figure it out already. Any help will be deeply appreciated.

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  • Release Management as Orchestra

    - by ericajanine
    I read an excellent, concise article (http://www.buildmeister.com/articles/software_release_management_best_practices) on the basics of release management practices. In the article, it states "Release Management is often likened to the conductor of an orchestra, with the individual changes to be implemented the various instruments within it." I played in music ensembles for years, so this is especially close to my heart as example. I learned most of my discipline from hours and hours of practice at the hand of a very skilled conductor and leader. I also learned that the true magic in symphonic performance is one where everyone involved is focused on one sound, one goal. In turn, that solid focus creates a sound and experience bigger than just mechanics alone accomplish. In symphony, a conductor's true purpose is to make you, a performer, better so the overall sound and end product is better. The big picture (the performance of the composition) is the end-game, and all musicians in the orchestra know without question their part makes up an important but incomplete piece of that performance. A good conductor works with each section (e.g. group) to ensure their individual pieces are solid. Let's restate: The conductor leads and is responsible for ensuring those pieces are solid. While the performers themselves are doing the work, the conductor is the final authority on when the pieces are ready or not. If not, the conductor initiates the efforts to get them ready or makes the decision to scrap their parts altogether for the sake of an overall performance. Let it sink in, because it's clear--It is not the performer's call if they play their part as agreed, it's the conductor's final call to allow it. In comparison, if a software release manager is a conductor, the only way for that manager to be effective is to drive the overarching process and execution of individual pieces of a software development lifecycle. It does not mean the release manager performs each and every piece, it means the release manager has oversight and influence because the end-game is a successful software enhancin a useable environment. It means the release manager, not the developer or development manager, has the final call if something goes into a software release. Of course, this is not a process of autocracy or dictation of absolute rule, it's cooperative effort. But the release manager must have the final authority to make a decision if something is ready to be added to the bigger piece, the overall symphony of software changes being considered for package and release. It also goes without saying a release manager, like a conductor, must have full autonomy and isolation from other software groups. A conductor is the one on the podium waving a little stick at the each section and cueing them for their parts, not yelling from the back of the room while also playing a tuba and taking direction from the horn section. I have personally seen where release managers are relegated to being considered little more than coordinators, red-tapers to "satisfy" the demands of an audit group without being bothered to actually respect all that a release manager gives a group willing to employ them fully. In this dysfunctional scenario, development managers, project managers, business users, and other stakeholders have been given nearly full clearance to demand and push their agendas forward, causing a tail-wagging-the-dog scenario where an inherent conflict will ensue. Depending on the strength, determination for peace, and willingness to overlook a built-in expectation that is wrong, the release manager here must face the crafted conflict head-on and diffuse it as quickly as possible. Then, the release manager must clearly make a case why a change cannot be released without negative impact to all parties involved. If a political agenda is solely driving a software release, there IS no symphony, there is no "software lifecycle". It's just out-of-tune noise. More importantly, there is no real conductor. Sometimes, just wanting to make a beautiful sound is not enough. If you are a release manager, are you freed up enough to move, to conduct the sections of software creation to ensure a solid release performance is possible? If not, it's time to take stock in what your role actually is and see if that is what you truly want to achieve in your position. If you are, then you can successfully build your career and that of the people in your groups to create truly beautiful software (music) together.

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  • No external microphone Acer AO722

    - by Leeghwater
    The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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  • Why is this beat detection code failing to register some beats properly?

    - by Quincy
    I made this SoundAnalyzer class to detect beats in songs: class SoundAnalyzer { public SoundBuffer soundData; public Sound sound; public List<double> beatMarkers = new List<double>(); public SoundAnalyzer(string path) { soundData = new SoundBuffer(path); sound = new Sound(soundData); } // C = threshold, N = size of history buffer / 1024 B = bands public void PlaceBeatMarkers(float C, int N, int B) { List<double>[] instantEnergyList = new List<double>[B]; GetEnergyList(B, ref instantEnergyList); for (int i = 0; i < B; i++) { PlaceMarkers(instantEnergyList[i], N, C); } beatMarkers.Sort(); } private short[] getRange(int begin, int end, short[] array) { short[] result = new short[end - begin]; for (int i = 0; i < end - begin; i++) { result[i] = array[begin + i]; } return result; } // get a array of with a list of energy for each band private void GetEnergyList(int B, ref List<double>[] instantEnergyList) { for (int i = 0; i < B; i++) { instantEnergyList[i] = new List<double>(); } short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; int samplesPerBand = nextSamples / B; // for the whole song while (sampleIndex + nextSamples < samples.Length) { complex[] FFT = FastFourier.Calculate(getRange(sampleIndex, nextSamples + sampleIndex, samples)); // foreach band for (int i = 0; i < B; i++) { double energy = 0; for (int j = 0; j < samplesPerBand; j++) energy += FFT[i * samplesPerBand + j].GetMagnitude(); energy /= samplesPerBand; instantEnergyList[i].Add(energy); } if (sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; samplesPerBand = nextSamples / B; } } // place the actual markers private void PlaceMarkers(List<double> instantEnergyList, int N, float C) { double timePerSample = 1 / (double)soundData.SampleRate; int index = N; int numInBuffer = index; double historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } } } For some reason it's only detecting beats from 637 sec to around 641 sec, and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates, and it seems that it's assigning a beat to each instant energy value in between those values. It's modeled after this: http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats register properly?

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  • AVAudioPlayer making noise when playing multiple sounds at the same time

    - by Rob
    I am having an issue where AVAudioPlayer is introducing noise into playback ONLY when I play multiple sound files at the same time. If I play them each individually, they all sound perfect. But, if I play sound clip B while sound clip A is still playing, the speakers start crackling like there is noise. I have tried both m4a files AND caf files and both make the same noise, so it has to be something with how I am implementing this method or a quirk with AVAudioPlayer. Any insights? code I am using: UITouch* touch = [[event allTouches] anyObject]; NSString* filename = [soundArray objectAtIndex:[touch view].tag]; NSString *path = [[NSBundle mainBundle] pathForResource:filename ofType:@"m4a"]; AVAudioPlayer * newAudio=[[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:path] error:NULL]; self.theAudio = newAudio; // automatically retain audio and dealloc old file if new m4a file is loaded [newAudio release]; // release the audio safely theAudio.delegate = self; [theAudio prepareToPlay]; [theAudio setNumberOfLoops:0]; [theAudio setVolume: volumeLevel]; [theAudio play];

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  • flash as3 document class and event listeners

    - by Lee
    I think i have this document class concept entirly wrong now, i was wondering if someone mind explaining it.. I assumed that the above class would be instantiated within the first frame on scene one of a movie. I also assumed that when changing scenes the state of the class would remain constant so any event listeners would still be running.. Scene 1: I have a movieclip named ui_mc, that has a button in for muting sound. Scene 2: I have the same movie clip with the same button. Now the eventListener picks it up in the first scene, however it does not in the second. I am wondering for every scene do the event listeners need to be resetup? If that is the case if their an event listener to listen for the change in scene, so i can set them back up again lol.. Thanks in advance.. package { import flash.display.MovieClip; import flash.events.MouseEvent; import flash.media.Sound; import flash.media.SoundChannel; public class game extends MovieClip { public var snd_state:Boolean = true; public function game() { ui_setup(); } public function ui_setup():void { ui_mc.toggleMute_mc.addEventListener(MouseEvent.CLICK, snd_toggle); } private function snd_toggle(event:MouseEvent):void { // 0 = No Sound, 1 = Full Sound trace("Toggle"); } } }

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  • Cocos2d shake/accelerometer issue.

    - by Ryan Poolos
    So I a little backstory. I wanted to implement a particle effect and sound effect that both last about 3 sec or so when the user shakes their iDevice. But first issue arrived when the build in UIEvent for shakes refused to work. So I took the advice of a few Cocos veterans to just use some script to get "violent" accelerometer inputs as shakes. Worked great until now. The problem is that if you keep shaking it just stacks the particle and sounds over and over. Now this wouldn't be that big of a deal except it happens even if you are careful to try and not do so. So what I am hoping to do is disable the accelerometer when the particle effect/sound effect start and then reenable it as soon as they finish. Now I don't know if I should do this by schedule, NStimer, or some other function. I am open to ALL suggestions. here is my current "shake" code. - (void)accelerometer:(UIAccelerometer *)accelerometer didAccelerate:(UIAcceleration *)acceleration { const float violence = 1; static BOOL beenhere; BOOL shake = FALSE; if (beenhere) return; beenhere = TRUE; if (acceleration.x > violence * 1.5 || acceleration.x < (-1.5* violence)) shake = TRUE; if (acceleration.y > violence * 2 || acceleration.y < (-2 * violence)) shake = TRUE; if (acceleration.z > violence * 3 || acceleration.z < (-3 * violence)) shake = TRUE; if (shake) { id particleSystem = [CCParticleSystemQuad particleWithFile:@"particle.plist"]; [self addChild: particleSystem]; // Super simple Audio playback for sound effects! [[SimpleAudioEngine sharedEngine] playEffect:@"Sound.mp3"]; shake = FALSE; } beenhere = FALSE; }

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  • SoundChannel, removeEventHandler, AS3

    - by pixelGreaser
    Is there a better way to use the sound channel is AS3? This works, but I hate it when I tap the play button twice and it starts doubling. Please advise. var mySound:Sound = new Sound(); playButton.addEventListener (MouseEvent.CLICK, myPlayButtonHandler); var myChannel:SoundChannel = new SoundChannel(); function myPlayButtonHandler (e:MouseEvent):void { myChannel = mySound.play(); } stopButton.addEventListener(MouseEvent.CLICK, onClickStop); function onClickStop(e:MouseEvent):void{ myChannel.stop(); } /*-----------------------------------------------------------------*/ //global sound buttons, add instance of 'killswitch' and 'onswitch' to stage killswitch.addEventListener(MouseEvent.CLICK, clipKillSwitch); function clipKillSwitch(e:MouseEvent):void{ var transform1:SoundTransform=new SoundTransform(); transform1.volume=0; flash.media.SoundMixer.soundTransform=transform1; } onswitch.addEventListener(MouseEvent.CLICK, clipOnSwitch); function clipOnSwitch(e:MouseEvent):void{ var transform1_:SoundTransform=new SoundTransform(); transform1_.volume=1; flash.media.SoundMixer.soundTransform=transform1_; }

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  • SoundChannel, Flash AS3

    - by pixelGreaser
    Is there a better way to use the sound channel is AS3? This works, but I hate it when I tap the play button twice and it starts doubling. Please advise. var mySound:Sound = new Sound(); playButton.addEventListener (MouseEvent.CLICK, myPlayButtonHandler); var myChannel:SoundChannel = new SoundChannel(); function myPlayButtonHandler (e:MouseEvent):void { myChannel = mySound.play(); } stopButton.addEventListener(MouseEvent.CLICK, onClickStop); function onClickStop(e:MouseEvent):void{ myChannel.stop(); } /*-----------------------------------------------------------------*/ //global sound buttons, add instance of 'killswitch' and 'onswitch' to stage killswitch.addEventListener(MouseEvent.CLICK, clipKillSwitch); function clipKillSwitch(e:MouseEvent):void{ var transform1:SoundTransform=new SoundTransform(); transform1.volume=0; flash.media.SoundMixer.soundTransform=transform1; } onswitch.addEventListener(MouseEvent.CLICK, clipOnSwitch); function clipOnSwitch(e:MouseEvent):void{ var transform1_:SoundTransform=new SoundTransform(); transform1_.volume=1; flash.media.SoundMixer.soundTransform=transform1_; }

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  • Tips for XNA WP7 Developers

    - by Michael B. McLaughlin
    There are several things any XNA developer should know/consider when coming to the Windows Phone 7 platform. This post assumes you are familiar with the XNA Framework and with the changes between XNA 3.1 and XNA 4.0. It’s not exhaustive; it’s simply a list of things I’ve gathered over time. I may come back and add to it over time, and I’m happy to add anything anyone else has experienced or learned as well. Display · The screen is either 800x480 or 480x800. · But you aren’t required to use only those resolutions. · The hardware scaler on the phone will scale up from 240x240. · One dimension will be capped at 800 and the other at 480; which depends on your code, but you cannot have, e.g., an 800x600 back buffer – that will be created as 800x480. · The hardware scaler will not normally change aspect ratio, though, so no unintended stretching. · Any dimension (width, height, or both) below 240 will be adjusted to 240 (without any aspect ratio adjustment such that, e.g. 200x240 will be treated as 240x240). · Dimensions below 240 will be honored in terms of calculating whether to use portrait or landscape. · If dimensions are exactly equal or if height is greater than width then game will be in portrait. · If width is greater than height, the game will be in landscape. · Landscape games will automatically flip if the user turns the phone 180°; no code required. · Default landscape is top = left. In other words a user holding a phone who starts a landscape game will see the first image presented so that the “top” of the screen is along the right edge of his/her phone, such that the natural behavior would be to turn the phone 90° so that the top of the phone will be held in the user’s left hand and the bottom would be held in the user’s right hand. · The status bar (where the clock, battery power, etc., are found) is hidden when the Game-derived class sets GraphicsDeviceManager.IsFullScreen = true. It is shown when IsFullScreen = false. The default value is false (i.e. the status bar is shown). · You should have a good reason for hiding the status bar. Users find it helpful to know what time it is, how much charge their battery has left, and whether or not their phone is in service range. This is especially true for casual games that you expect someone to play for a few minutes at a time, e.g. while waiting for some event to start, for a phone call to come in, or for a train, bus, or subway to arrive. · In portrait mode, the status bar occupies 32 pixels of space. This means that a game with a back buffer of 480x800 will be scaled down to occupy approximately 461x768 screen pixels. Setting the back buffer to 480x768 (or some resolution with the same 0.625 aspect ratio) will avoid this scaling. · In landscape mode, the status bar occupies 72 pixels of space. This means that a game with a back buffer of 800x480 will be scaled down to occupy approximately 728x437 screen pixels. Setting the back buffer to 728x480 (or some resolution with the same 1.51666667 aspect ratio) will avoid this scaling. Input · Touch input is scaled with screen size. · So if your back buffer is 600x360, a tap in the bottom right corner will come in as (599,359). You don’t need to do anything special to get this automatic scaling of touch behavior. · If you do not use full area of the screen, any touch input outside the area you use will still register as a touch input. For example, if you set a portrait resolution of 240x240, it would be scaled up to occupy a 480x480 area, centered in the screen. If you touch anywhere above this area, you will get a touch input of (X,0) where X is a number from 0 to 239 (in accordance with your 240 pixel wide back buffer). Any touch below this area will give a touch input of (X,239). · If you keep the status bar visible, touches within its area will not be passed to your game. · In general, a screen measurement is the diagonal. So a 3.5” screen is 3.5” long from the bottom right corner to the top left corner. With an aspect ratio of 0.6 (480/800 = 0.6), this means that a phone with a 3.5” screen is only approximately 1.8” wide by 3” tall. So there are approximately 267 pixels in an inch on a 3.5” screen. · Again, this time in metric! 3.5 inches is approximately 8.89 cm. So an 8.89 cm screen is 8.89 cm long from the bottom right corner to the top left corner. With an aspect ratio of 0.6, this means that a phone with an 8.89 cm screen is only approximately 4.57 cm wide by 7.62 cm tall. So there are approximately 105 pixels in a centimeter on an 8.89 cm screen. · Think about the size of your finger tip. If you do not have large hands, think about the size of the fingertip of someone with large hands. Consider that when you are sizing your touch input. Especially consider that when you are spacing two touch targets near one another. You need to judge it for yourself, but items that are next to each other and are each 100x100 should be fine when it comes to selecting items individually. Smaller targets than that are ok provided that you leave space between them. · You want your users to have a pleasant experience. Making touch controls too small or too close to one another will make them nervous about whether they will touch the right target. Take this into account when you plan out your game initially. If possible, do some quick size mockups on an actual phone using colored rectangles that you position and size where you plan to have your game controls. Adjust as necessary. · People do not have transparent hands! Nor are their hands the size of a mouse pointer icon. Consider leaving a dedicated space for input rather than forcing the user to cover up to one-third of the screen with a finger just to play the game. · Another benefit of designing your controls to use a dedicated area is that you’re less likely to have players moving their finger(s) so frantically that they accidentally hit the back button, start button, or search button (many phones have one or more of these on the screen itself – it’s easy to hit one by accident and really annoying if you hit, e.g., the search button and then quickly tap back only to find out that the game didn’t save your progress such that you just wasted all the time you spent playing). · People do not like doing somersaults in order to move something forward with accelerometer-based controls. Test your accelerometer-based controls extensively and get a lot of feedback. Very well-known games from noted publishers have created really bad accelerometer controls and been virtually unplayable as a result. Also be wary of exceptions and other possible failures that the documentation warns about. · When done properly, the accelerometer can add a nice touch to your game (see, e.g. ilomilo where the accelerometer was used to move the background; it added a nice touch without frustrating the user; I also think CarniVale does direct accelerometer controls very well). However, if done poorly, it will make your game an abomination unto the Marketplace. Days, weeks, perhaps even months of development time that you will never get back. I won’t name names; you can search the marketplace for games with terrible reviews and you’ll find them. Graphics · The maximum frame rate is 30 frames per second. This was set as a compromise between battery life and quality. · At least one model of phone is known to have a screen refresh rate that is between 59 and 60 hertz. Because of this, using a fixed time step with a target frame rate of 30 will cause a slight internal delay to build up as the framework is forced to wait slightly for the next refresh. Eventually the delay will get to the point where a draw is skipped in order to recover from the delay. (See Nick's comment below for clarification.) · To deal with that delay, you can either stay with a fixed time step and set the frame rate slightly lower or else you can go to a variable time step and make sure to adjust all of your update data (e.g. player movement distance) to take into account the elapsed time from the last update. A variable time step makes your update logic slightly more complicated but will avoid frame skips entirely. · Currently there are no custom shaders. This might change in the future (there is no hardware limitation preventing it; it simply wasn’t a feature that could be implemented in the time available before launch). · There are five built-in shaders. You can create a lot of nice effects with the built-in shaders. · There is more power on the CPU than there is on the GPU so things you might typically off-load to the GPU will instead make sense to do on the CPU side. · This is a phone. It is not a PC. It is not an Xbox 360. The emulator runs on a PC and uses the full power of your PC. It is very good for testing your code for bugs and doing early prototyping and layout. You should not use it to measure performance. Use actual phone hardware instead. · There are many phone models, each of which has slightly different performance levels for I/O, screen blitting, CPU performance, etc. Do not take your game right to the performance limit on your phone since for some other phones you might be crossing their limits and leaving players with a bad experience. Leave a cushion to account for hardware differences. · Smaller screened phones will have slightly more dots per inch (dpi). Larger screened phones will have slightly less. Either way, the dpi will be much higher than the typical 96 found on most computer screens. Make sure that whoever is doing art for your game takes this into account. · Screens are only required to have 16 bit color (65,536 colors). This is common among smart phones. Using gradients on a 16 bit display can produce an ugly artifact known as banding. Banding is when, rather than a smooth transition from one color to another, you instead see distinct lines. Be careful to avoid this when possible. Banding can be avoided through careful art creation. Its effects can be minimized and even unnoticeable when the texture in question is always moving. You should be careful not to rely on “looks good on my phone” since some phones do have 32-bit displays and thus you’ll find yourself wondering why you’re getting bad reviews that complain about the graphics. Avoid gradients; if you can’t, make sure they are 16-bit safe. Audio · Never rely on sounds as your sole signal to the player that something is happening in the game. They might have the sound off. They might be playing somewhere loud. Etc. · You have to provide controls to disable sound & music. These should be separate. · On at least one model of phone, the volume control API currently has no effect. Players can adjust sound with their hardware volume buttons, but in game selectors simply won’t work. As such, it may not be worth the effort of providing anything beyond on/off switches for sound and music. · MediaPlayer.GameHasControl will return true when a game is hooked up to a PC running Zune. When Zune is running, any attempts to do anything (beyond check GameHasControl) with MediaPlayer will cause an exception to be thrown. If this exception is thrown, catch it and disable music. Exceptions take time to propagate; you don’t want one popping up in every single run of your game’s Update method. · Remember that players can already be listening to music or using the FM radio. In this case GameHasControl will be false and you should handle this appropriately. You can, alternately, ask the player for permission to stop their current music and play your music instead, but the (current) requirement that you restore their music when done is very hard (if not impossible) to deal with. · You can still play sound effects even when the game doesn’t have control of the music, but don’t think this is a backdoor to playing music. Your game will fail certification if your “sound effect” seems to be more like music in scope and length.

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  • Free space on Dedi' in CentOS

    - by Trance84
    It will sound stupid but i need to figure out how much disk space i have in my dedicated server, it runs CentOS6...the last command i issued was this [root@ks34900 ~]# df -h Filesystem Size Used Avail Use% Mounted on rootfs 9.7G 6.4G 2.9G 69% / /dev/root 9.7G 6.4G 2.9G 69% / none 1000M 288K 1000M 1% /dev /dev/sda2 914G 200M 868G 1% /home But again, stupid as it may sound... i cant figure out how much space i have in "/" folder (root) And is it possible that "/usr" have a different space (partition)?

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