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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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  • Any way to surround code block with Curly Braces {} in VS2008?

    - by Jim McKeeth
    I always find myself needing to enclose a block of code in curly braces { }, but unfortunately that isn't included in the C# surround code snippets, which seems to be an oversight. I couldn't find anything on building your own surround snippets either (just other kinds of snippets). I am actually running Resharper too, but it doesn't seem to have this functionality either (or I haven't figured how to activate it). We have a coding standard of including even a single line of code after an if or else in curly braces, so if I could just make Resharper do that refactor automatically that would be even better!

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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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  • Anyway to surround code block with curly braces {} in VS2008?

    - by Jim McKeeth
    I always find myself needing to enclose a block of code in curly braces { }, but unfortunately that isn't included in the C# surround code snippets, which seems to be an oversight. I couldn't find anything on building your own surround snippets either (just other kinds of snippets). I am actually running Resharper too, but it doesn't seem to have this functionality either (or I haven't figured how to activate it). We have a coding standard of including even a single line of code after an if or else in curly braces, so if I could just make Resharper do that refactor automatically that would be even better!

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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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  • Speech recognition with Flash or Silverlight

    - by Sebastián Grignoli
    I'm developing a web user interface to enter some information that is not very complex but needs to be loaded in real time. I think that the application could make use of speech recognition to facilitate the task. Te core of the interface is being built with Javascript and jQuery, but can easily include a flash or silverlight component. I believe that´s probably the way to go... I don't need to recognize everything that the user says, but only a few prerecorded commands. Also, I don't want the user to click on a button to specify the begining and the end of the spoken command. It should be detected live. Is there anything that does this? I would be grateful if anyone tells me about a complete solution, free or commercial, as well as any advice on capturing a sound stream from the mic and process it with flash or sliverlight. Sebastian.-

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  • Check if song is buffering in AS3

    - by SXMC
    I have the following piece of code: var song:Sound; var sndChannel:SoundChannel; var context:SoundLoaderContext = new SoundLoaderContext(2000); function songLoad():void { song.load(new URLRequest(songs[selected]),context); sndChannel = song.play(); } Now I want to be able to check if the song is buffering or not. Is there a way to do this? Or should I approach it differently? Thanks in advance!

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  • SoundPool.load() and FileDescriptor from file

    - by Hans
    I tried using the load function of the SoundPool that takes a FileDescriptor, because I wanted to be able to set the offset and length. The File is not stored in the Ressources but a file on the storage card. Even though neither the load nor the play function of the SoundPool throw any Exception or print anything to the console, the sound is not played. Using the same code, but use the file path string in the SoundPool constructor works perfectly. This is how I have tried the loading (start equals 0 and length is the length of the file in miliseconds): FileInputStream fileIS = new FileInputStream(new File(mFile)); mStreamID = mSoundPool.load(fileIS.getFD(), start, length, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); If I would use this, it works: mStreamID = mSoundPool.load(mFile, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); Any ideas anyone? Thanks

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  • iPhone: How many instances of AVAudioPlayer should I have for multiple sounds?

    - by foreyez
    So I'm using AvAudioPlayer to play multiple wav files. About 20 different sounds (each about 1 sec long), and you can think of each being played on a button press. Also I don't need them all to play simultaneously, i.e., one plays and you press another button to play another one (which stops the currently played one). What I'm wondering, should I have multiple instances of AVAudioPlayer (20 of them) and then preload the audio files, or should I just use one instance of AvAudioPlayer and each time a button is pressed, initialize the AvAudioPlayer with the sound url (or would this be too slow?) Thanks in advance!

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  • Linux, C++ audio capturing (just microphone) library

    - by TheOm3ga
    I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

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  • Comparing two speech sounds

    - by JessicaB
    I need to be able to determine if two sounds are very similar. The goal is to have a very limited vocabulary (10 or 15) of short one or two syllable words, then compare a captured sound to determine if it is one of those items with all the usual variability in environmental and capture conditions. The idea is that the user can issue a few simple commands by voice instead of keyboard or mouse. Does anyone know the best approach to this? I don't want to do full blown speech recognition, just something much more limited.

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  • Using Pidgin 2.5.2 in Linux no sound is made for incoming messages (as it should)

    - by Kent
    Hi, I have a problem with Pidgin 2.5.2 in Linux (Ubuntu 8.10). When someone sends me a message no sound is played (the tray indicates a new message and blinks, that's it). Sounds play fine when I send someone a message. If I preview the Message received and Message sent sound events both of them do make a sound. Automatic and ALSA is what's working from the alternatives in Sound method selection. I include a screenshot containing a lot of relevant information: Screenshot (It's to big to fit nicely inline.)

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  • Slideshow from excel file listing the caption, sound file and image file?

    - by Slabo
    Hello, I have excel files with the following header: Caption Sound: Location of sound file Image: Location of image file How can I make a slideshow from this? Each slide should show image, caption, and play sound automatically according to the excel list. I don't care what software I use, if I can get the job done. Total slides ~10,000. In case interested,this is review material for English second language students. Any help appreciated, Thanks

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  • How to play the same Sound multiple times with overlap, using OpenAL or Finch?

    - by mystify
    Finch uses OpenAL. However, when I have an instance of Sound, and say -play, the sound plays. When I call -play multiple times one after another in a fast paced way, every -play makes the current sound playback of that sound stop and restart it. That's not what I want. Would I have to create multiple sources or buffers to get that working? Or would I just instantiate multiple Sounds with the same file?

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