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  • How do I enable sound with the "linux-virtual" kernel?

    - by Ola Tuvesson
    I've been trying to enable sound for the linux-virtual kernel as I want to run an ultra slim Ubuntu server under VirtualBox but need audio. The resource usage difference between virtual and generic/server is surprisingly large, with the virtual kernel system using 80Mb less RAM after a clean boot (130Mb vs 210Mb), and I really want to squeeze every clock cycle and available byte I can out of the system. Besides, the virtual kernel has some additional optimisations enabled specifically for virtual machines (or so I am told). Now I have compiled my own kernel a few times in the past, for example to include the Intel-PHC module (for improved power management on Thinkpads), so the concept is not entirely alien to me, but I've run into a strange problem which I'm hoping someone can help explain: When I do a diff between the config files for Linux-generic and Linux-virtual there are precious few differences, and certainly none which pertain to sound support; there are really only five or six lines which differ, and they're mainly to do with i/o timing, sleep state and priorities. What gives? I expected the differences to be extensive, and that I would be able to identify the options that enabled audio by looking at them, but my problem doesn't seem to be related to the config file at all (yes, I know about the sound drivers section - it is identical between the two kernel configs). Am I looking in the wrong place? Many thanks!

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  • Sound Effects/Manipulation?

    - by Adam
    Hello, I am creating an android app that basically records an applies an "Effect" on the audio track then plays it back. I got my app to record an play back but I am stuck an not sure where do go from here. I have been Googling for days now trying to find a open source audio library or some way to change the audio after I record it. I currently have it setup to play back using SoundPool an I't lets me speed up an slow down the audio. I would like to do things like change pitch an add echo etc. I will appreciate any responses because I am totally stumped right now. Thanks Adam

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  • Help with SDL_mixer (no sound)

    - by Kaizoku
    Hello, I have this strange problem with SDL_mixer, it doesn't want to play music. It doesn't throw any error, it just skips it. Any advice? I am compiling on linux with libvorbis. audio.h #ifndef AUDIO_H #define AUDIO_H #include <string> #include <SDL/SDL_mixer.h> class Audio { private: Mix_Music *music; public: Audio(); virtual ~Audio(); public: void setMusic(std::string path); void playMusic(); }; #endif /* AUDIO_H */ audio.cpp #include "Audio.h" #include <stdexcept> Audio::Audio() { if (0 == Mix_Init(MIX_INIT_OGG)) throw std::runtime_error(Mix_GetError()); if (-1 == Mix_OpenAudio(44100, MIX_DEFAULT_FORMAT, MIX_DEFAULT_CHANNELS, 4096)) throw std::runtime_error(Mix_GetError()); } Audio::~Audio() { Mix_FreeMusic(music); Mix_Quit(); } void Audio::setMusic(std::string path) { music = Mix_LoadMUS(path.c_str()); if (NULL == music) throw std::runtime_error(Mix_GetError()); } void Audio::playMusic() { if (NULL != music) { if (-1 == Mix_PlayMusic(music, -1)) throw std::runtime_error(Mix_GetError()); } }

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  • How to sound audible bell from crontab

    - by user1526251
    The command line: /bin/echo -e "\007" in bash will ring the bell. With the line: /bin/echo -e "\007" in my crontab I expected the bell to ring every minute, but it's silent. I know crontab is working because the line: /bin/touch $HOME/jkjkjk updates the file jkjkjk every minute as it should. I found a posting some years ago suggesting that standard output should be directed to /dev/tty1 in crontab. But the line: /bin/echo "\007" /dev/tty1 Still fails. What to try next?

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  • Open Source sound engine

    - by Steph Thirion
    When I started using SoundEngine (from CrashLanding and TouchFighter), I had read about a few people recommending not to use it, for it was, according to them, not stable enough. Still it was the only solution I knew of to play sounds with pitch and position control without learning C++ and OpenAL, so I ignored the warnings and went on with it. But now I'm starting to worry. The 2.2 SDK introduced AVFoundation. Using both SoundEngine from CrashLanding (for sounds) and AVAudioPlayer (for music), I found out SoundEngine behaves strangely when the only existing AVAudioPlayer is released (all sounds stop until a new AVAudioPlayer is initiated). Around the same time as the 2.2 SDK came out, the CrashLanding sample code was mysteriously removed from the ADC site. I'm worried there are more bad surprises to come. My question is, is anyone aware of an Open Source alternative to SoundEngine? Maybe even a C++ library that uses OpenAL?

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  • Sound Manager Classes for Windows

    - by Yakov
    I need some classes for playing short wav sounds, this classes would load this wav files into memory when an instance created, play sounds in background when needed, release this wav files from memory when an instance disposed. How can I do this on C# for windows (.Net 2.0)? (Win API's sndPlaySound, OpenAL or may be any wrapper) Ideally I would love to find an exist solution that simple and able to solve my task. Do you know any solutions for this issue?

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  • Does this sound like a stack overflow?

    - by Jordan S
    I think I might be having a stack overflow problem or something similar in my embedded firmware code. I am a new programmer and have never dealt with a SO so I'm not sure if that is what's happening or not. The firmware controls a device with a wheel that has magnets evenly spaced around it and the board has a hall effect sensor that senses when magnet is over it. My firmware operates the stepper and also count steps while monitoring the magnet sensor in order to detect if the wheel has stalled. I am using a timer interrupt on my chip (8 bit, 8057 acrh.) to set output ports to control the motor and for the stall detection. The stall detection code looks like this... // Enter ISR // Change the ports to the appropriate value for the next step // ... StallDetector++; // Increment the stall detector if(PosSensor != LastPosMagState) { StallDetector = 0; LastPosMagState = PosSensor; } else { if (PosSensor == ON) { if (StallDetector > (MagnetSize + 10)) { HandleStallEvent(); } } else if (PosSensor == OFF) { if (StallDetector > (GapSize + 10)) { HandleStallEvent(); } } } this code is called every time the ISR is triggered. PosSensor is the magnet sensor. MagnetSize is the number of stepper steps that it takes to get through the magnet field. GapSize is the number of steps between two magnets. So I want to detect if the wheel gets stuck either with the sensor over a magnet or not over a magnet. This works great for a long time but then after a while the first stall event will occur because 'StallDetector (MagnetSize + 10)' but when I look at the value of StallDetector it is always around 220! This doesn't make sense because MagnetSize is always around 35. So the stall event should have been triggered at like 46 but somehow it got all the way up to 220? And I don't set the value of stall detector anywhere else in my code. Do you have any advice on how I can track down the root of this problem? The ISR looks like this void Timer3_ISR(void) interrupt 14 { OperateStepper(); // This is the function shown above TMR3CN &= ~0x80; // Clear Timer3 interrupt flag } HandleStallEvent just sets a few variable back to their default values so that it can attempt another move... #pragma save #pragma nooverlay void HandleStallEvent() { ///* PulseMotor = 0; //Stop the wheel from moving SetMotorPower(0); //Set motor power low MotorSpeed = LOW_SPEED; SetSpeedHz(); ERROR_STATE = 2; DEVICE_IS_HOMED = FALSE; DEVICE_IS_HOMING = FALSE; DEVICE_IS_MOVING = FALSE; HOMING_STATE = 0; MOVING_STATE = 0; CURRENT_POSITION = 0; StallDetector = 0; return; //*/ } #pragma restore

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  • Simple sound effect loop using AudioToolKit

    - by Typeoneerror
    I've created a few sounds for use in my game. I can play them at certain events without issue: // create sounds CFBundleRef mainBundle; mainBundle = CFBundleGetMainBundle(); _soundFileShake = CFBundleCopyResourceURL(mainBundle, CFSTR("shake"), CFSTR("wav"), NULL); AudioServicesCreateSystemSoundID(_soundFileShake, &_soundIdShake); // later... AudioServicesPlaySystemSound(_soundIdShake); The game has a mechanism which allows you to shake the device to activate some functionality. I've got the shaking code done so I get get a "shaking started" and "shaking ended" message to my game. What I need to have happen is start playing "shave.wav" when shaking starts and loop it until it stops. Is there a way to do this with AudioToolbox/AudioServices? How could I do this if not?

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  • save and play recorded sound

    - by blacksheep
    i'd like to save and play again this recorded sounds: @interface Recorder : NSObject { NSMutableArray *times; NSMutableArray *samples; } @end @implementation Recorder – (id) init { [super init]; times = [[NSMutableArray alloc] init]; samples = [[NSMutableArray alloc] init]; return self; } – (void) recordSound: (id) someSound { CFAbsoluteTime now = CFAbsoluteTimeGetCurrent(); NSNumber *wrappedTime = [NSNumber numberWithDouble:now]; [times addObject:wrappedTime]; [samples addObject:someSound]; } @end thanx blacksheep

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  • Can I install 12.04 packages on 11.10?

    - by Jason R
    I'm running 11.10 and am trying to apply the fix to this bug in Empathy, shown at the very bottom. There is an updated package for the offending component available in the Precise repository, and someone even posted a backported .deb package for use on Oneiric. However, when I try to install that package, it seems to have a dependency on a package that isn't available for Oneiric: (Reading database ... 254452 files and directories currently installed.) Preparing to replace telepathy-indicator 0.0.7-0ubuntu1 (using telepathy-indicator_0.1.1-0ubuntu1_amd64.deb) ... Unpacking replacement telepathy-indicator ... dpkg: dependency problems prevent configuration of telepathy-indicator: telepathy-indicator depends on libunity9 (>= 3.4.6); however: Package libunity9 is not installed. dpkg: error processing telepathy-indicator (--install): dependency problems - leaving unconfigured Errors were encountered while processing: telepathy-indicator The person who posted the backported telepathy-indicator package indicated that it depends upon libunity-dev-5.0; the latest version in the Oneiric repositories is a 4.0 vintage. I also can't find a libunity9 available for Oneiric, so I'm wondering: is it possible to just add the Precise repository to my list and pull the updated packages from there, or should I not expect that they would operate correctly?

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  • Sound Manager Classes for Windows (C# or C++ .Net 2.0)

    - by Yakov
    Hi guys! I need some classes for playing short wav sounds, this classes would load this wav files into memory when an instance created, play sounds in background when needed, release this wav files from memory when an instance disposed. How can I do this on C# for windows (.Net 2.0)? (Win API's sndPlaySound, OpenAL or may be any wrapper) Ideally I would love to find an exist solution that simple and able to solve my task. Do you know any solutions for this issue? Thankx for your time.

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  • change sound effect in asp.net

    - by beaso_88
    in fact i have educational sites for small students , this sites contains hundered of educational MP3 files , our aim is to convert these MP3 files to funny sounds . i have search on the net and i found great example on C#.net . http://channel9.msdn.com/coding4fun/articles/Skype-Voice-Changer but my problem , i want to do that in asp.net not windows form application. who can help me?? assume that i have mp3 file and i want to make some change on it then save these changes to this file.

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  • Using python to play two sine tones at once

    - by Alex
    I'm using python to play a sine tone. The tone is based off the computer's internal time in minutes, but I'd like to simultaneously play one based off the second for a harmonized or dualing sound. This is what I have so far; can someone point me in the right direction? from struct import pack from math import sin, pi import time def au_file(name, freq, dur, vol): fout = open(name, 'wb') # header needs size, encoding=2, sampling_rate=8000, channel=1 fout.write('.snd' + pack('>5L', 24, 8*dur, 2, 8000, 1)) factor = 2 * pi * freq/8000 # write data for seg in range(8 * dur): # sine wave calculations sin_seg = sin(seg * factor) fout.write(pack('b', vol * 127 * sin_seg)) fout.close() t = time.strftime("%S", time.localtime()) ti = time.strftime("%M", time.localtime()) tis = float(t) tis = tis * 100 tim = float(ti) tim = tim * 100 if __name__ == '__main__': au_file(name='timeSound1.au', freq = tim, dur=1000, vol=1.0) import os os.startfile('timeSound1.au')

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  • Normalize FFT magnitude to imitate WMP

    - by Bevin
    So, I've been working on a little visualizer for sound files, just for fun. I basically wanted to imitate the "Scope" and "Ocean Mist" visualizers in Windows Media Player. Scope was easy enough, but I'm having problems with Ocean Mist. I'm pretty sure that it is some kind of frequency spectrum, but when I do an FFT on my waveform data, I'm not getting the data that corresponds to what Ocean Mist displays. The spectrum actually looks correct, so I knew there was nothing wrong with the FFT. I'm assuming that the visualizer runs the spectrum through some kind of filter, but I have no idea what it might be. Any ideas?

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  • Using python to play two sin tones at once

    - by Alex
    Im using python to a sine tone. the tone is based off the computers internal time in minutes, but id like to simultaneously play one based off the second for a harmonized or dualing sound. This is what I have so far can someone point me in the right direction. from struct import pack from math import sin, pi import time def au_file(name, freq, dur, vol): fout = open(name, 'wb') # header needs size, encoding=2, sampling_rate=8000, channel=1 fout.write('.snd' + pack('>5L', 24, 8*dur, 2, 8000, 1)) factor = 2 * pi * freq/8000 # write data for seg in range(8 * dur): # sine wave calculations sin_seg = sin(seg * factor) fout.write(pack('b', vol * 127 * sin_seg)) fout.close() t = time.strftime("%S", time.localtime()) ti = time.strftime("%M", time.localtime()) tis = float(t) tis = tis * 100 tim = float(ti) tim = tim * 100 if name == 'main': au_file(name='timeSound1.au', freq = tim, dur=1000, vol=1.0) import os os.startfile('timeSound1.au')

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  • SoundChannel object plays small portion after being stopped and played again

    - by gok
    SoundChannel object is stopped and played again. When played again it plays small portion from the previous position and suddenly jumps back to the beginning. It doesn't play the whole sound before looping. This happens only once, then it loops normally. It happens again if I stop and play. public function play():void { channel = clip.play(trimIn); volume(currentVolume); isPlaying = true; timer.start(); channel.addEventListener(Event.SOUND_COMPLETE, loopMusic); } public function loopMusic(e:Event=null):void { if (channel != null) { timer.stop(); channel.removeEventListener(Event.SOUND_COMPLETE, loopMusic); play(); } } Do I need to somehow reset the soundChannel?

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  • How do I get a mp3 file's total time in Java?

    - by Tom Brito
    The answers provided in How do I get a sound file’s total time in Java? work well for wav files, but not for mp3 files. They are (given a file): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(file); AudioFormat format = audioInputStream.getFormat(); long frames = audioInputStream.getFrameLength(); double durationInSeconds = (frames+0.0) / format.getFrameRate(); and: AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(file); AudioFormat format = audioInputStream.getFormat(); long audioFileLength = file.length(); int frameSize = format.getFrameSize(); float frameRate = format.getFrameRate(); float durationInSeconds = (audioFileLength / (frameSize * frameRate)); They give the same correct result for wav files, but wrong and different results for mp3 files. Any idea what do I have to do to get the mp3 file's duration?

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  • What language/API to use for a standalone live-input audio visualizer app?

    - by knuckfubuck
    I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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  • OpenAL - determine maximum sources

    - by Bill Kotsias
    Is there an API that allows you to define the maximum number of OpenAL "sources" allowed by the underlying sound hardware? Searching the internet, I found 2 recommendations : keep generating OpenAL sources till you get an error. However, there is a note in FreeSL (OpenAL wrapper) stating that this is "very bad and may even crash the library" assume you only have 16; why would anyone ever require more? (!) The second recommendation is even adopted by FreeSL. So, is there a common API to define the number of simultaneous "voices" supported? Thank you for your time, Bill

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