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  • Playing Ogg Sound in Android

    - by baba tenor
    In my application, I am trying to play a sound file in ogg format, stored in raw folder in res directory of my application. When I press the button that calls below function, it just freezes with the button pressed and does not respond. In the end, I have to terminate the application from Eclipse. Nothing about an error or exception in Logcat. In debugging mode, it enters create function and never comes back. What am I doing wrong? private void playbeep() { mPlayer = MediaPlayer.create(this, R.raw.beep); mPlayer.start(); mPlayer.release(); }

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  • Experiences teaching or learning map/reduce/etc before recursion?

    - by Jay
    As far as I can see, the usual (and best in my opinion) order for teaching iterting constructs in functional programming with Scheme is to first teach recursion and maybe later get into things like map, reduce and all SRFI-1 procedures. This is probably, I guess, because with recursion the student has everything that's necessary for iterating (and even re-write all of SRFI-1 if he/she wants to do so). Now I was wondering if the opposite approach has ever been tried: use several procedures from SRFI-1 and only when they are not enough (for example, to approximate a function) use recursion. My guess is that the result would not be good, but I'd like to know about any past experiences with this approach.

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  • How can I set different sound for different players by using [[SimpleAudioEngine sharedEngine] playE

    - by srikanth rongali
    I need to set sounds for different players in my game. There are 10 players. And I have 10 sounds. The players are loaded int his way for( int i = 1; i <5; i++ ) { [playerAnimation addFrameWithFilename: [NSString stringWithFormat:@"Player %02d gun draw_%02d.png", playerNumber, i]]; } How can I set the sounds in this way by giving the filenames. And player1 shoots player1sound should play. How can I do it using [[SimpleAudioEngine sharedEngine] playEffect:@"player1.sound.wav"];

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  • how can i stop my sound file from being auto download in php jquery

    - by testkhan
    i have following code in my php jquery call... <object type="application/x-shockwave-flash" data="mysounds/player.swf" id="audioplayer1" height="1" width="1"> <param name="movie" value="mysounds/player.swf" /> <param name="FlashVars" value="playerID=audioplayer1&autostart=yes&soundFile=mysounds/online.mp3" /> <param name="quality" value="high" /> <param name="menu" value="false" /> <param name="wmode" value="transparent" /> </object> and i have internetdownloadmanager installed on my pc when ever i try to load the page it start downloading the sound with internetdownloadmanager how can i stop that....and prevent it from auto downloading from any downloader...

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  • Definition of "lisp form"?

    - by josh
    Hi, What exactly the definition of a "Lisp form"? As far as I know, it's "either an atom or a list that has a symbol as its first element". But then, this (in Scheme) would not be a form: ((lambda () 42)) ;; The answer to Life, the Universe and Everything. Because the first element of the list is itself another list. And after it's evaluated it will be a procedure (not a symbol). I can find several different websites and tutorials talking about Lisp forms, but none which gives a complete and detailed definition. Where can I find one?

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  • sound not playing when i press the button and how to fix overlapping sounds

    - by alfredjunco
    the code is giving me an error"Unused variable'path'" and when i press a button there is no sound playing how do i fix this the aSound is in the h file - (void)playOnce:(NSString *)aSound; - (IBAction) beatButton50 { [self playOnce:@"racecars"]; } - (void)playOnce:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if([theAudio isPlaying]) { [theAudio stop]; } } - (void)playLooped:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if (!theAudio) { theAudio = [[AVAudioPlayer alloc] initWithContentsOfURL: [NSURL fileURLWithPath: path] error: NULL]; } [theAudio setDelegate: self]; // loop indefinitely [theAudio setNumberOfLoops: -1]; [theAudio setVolume: 1.0]; [theAudio play]; } - (void)stopAudio { [theAudio stop]; [theAudio setCurrentTime:0]; }

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  • Writing lambda functions in Scala

    - by user2433237
    I'm aware that you can write anonymous functions in Scala but I'm having trouble trying to convert a piece of code from Scheme. Could anyone help me convert this to Scala? (define apply-env (lambda (env search-sym) (cases environment env (empty-env () (eopl:error 'apply-env "No binding for ~s" search-sym)) (extend-env (var val saved-env) (if (eqv? search-sym var) val (apply-env saved-env search-sym))) (extend-env-rec (p-name b-var p-body saved-env) (if (eqv? search-sym p-name) (proc-val (procedure b-var p-body env)) (apply-env saved-env search-sym)))))) Thanks in advance

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  • Extra line breaks inserted in MrEd text%

    - by Jesse Millikan
    In a DrScheme project, I'm using a MrEd editor-canvas% with text% and inserting a string from a literal in a Scheme file. This results in an extra blank line in the editor for each line of text I'm trying to insert. Is this a Windows vs. Unix linebreak problem? I can't find anything about text% treats line breaks in the documentation. ; Inside a class definition: (define/public (edit-pattern p j b d h) (send input-beat set-value (number->string b)) (send input-dwell set-value (number->string d)) (send hold-beats set-value (number->string h)) (send juggler-t erase) ; Why do these add extra newlines (send juggler-t insert j) (send pattern-t erase) (send pattern-t insert p)) (define juggler-ec (new editor-canvas% [parent this] [line-count 12])) (define juggler-t (new text%)) (send juggler-ec set-editor juggler-t) (define pattern-ec (new editor-canvas% [parent this] [line-count 20])) (define pattern-t (new text%)) (send pattern-ec set-editor pattern-t) ; Lots of other stuff...

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  • Amarok on Ubuntu 9.10 rapidly skipping tracks

    - by danwoods
    Hello all. Subject really says it all. When I try to click on any track, Amarok rapidly moves through the playlist, pulling external information (wikipedia etc) for each song, but not playing it. I feel like it might be a problem with my sound card or driver or something, but all other media players work fine. I can play sounds from within Amarok when I test different sound devices (they all play). I removed the program and re-installed it and still no luck. Any ideas?

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  • How to get the contents of the wav file into array so as to cut the required segment and convert it

    - by kaushik
    How to get the contents of the wav file into array so as to cut the required segment and convert it back to wav format using python?? My prob is similar to "ROMANs" prob,i hav seen earlier in the post at this site.. Basically,i want to combine parts of different wav file into one wav file?? if there is ne other apporach thn takin the contents into an array and cuting part and combining and again converting bac? please suggest... edited: I prefer unpacking the contents of the wave file into an array and editing by cutting the required segment of sound from the wav file,as i am working on speech processing,and guess this way would be easy to enchance the quality of sound later... can ne one suggest a way for this?? Plz help.. Thanks in advance.

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  • Installing Skype on Amazon EC2 instance

    - by Adrian
    For my application, I need to have Skype working on my Amazon EC2 Windows instance. I got the application installed and am able to log in, however, I can't make a phone call, since I am getting an 'Can't detect your sound card' error. Since I'm trying to inject audio from an audio file into the phone call, I don't need the sound card on the server. Thus, I need a way to bypass this error message. I have tried installing Virtual Audio Cable, which unfortunately didn't work (even though it worked on my desktop machine).

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  • What is the best API in any language for Audio and MIDI music application development?

    - by noneme
    What, in your opinion, is the best API to utilize in developing an application that handles both realtime MIDI and audio input and output? This would be for an application that is used in the process of making music as opposed to playing audio or MIDI files. I'm aware that this may be a subjective question, but if you know of an API that is dominantly used for these purposes, please share it. I'm agnostic about which language the API is for, and I also don't care about portability. The real concern is for an API that is well documented, well designed (e.g. thought out and intuitive to developers using it), and actively maintained. OS portability would be nice, but it is second to having an API/Language that meets the previous requirements. Please note that the emphasis is not on API's for sound synthesis or for composing music with code. It is intended for the handling of sound file and MIDI data in a real-time context.

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  • What should I use to replace the WinAPI Beep() function?

    - by Jon Cage
    I've got a Visual C++/CLI app which uses beeps to signify good and bad results (used when the user can't see the screen). Currently I use low pitched beeps for bad results and high pitched beeps for good results: if( goodResult == true ) { Beep(1000, 40); } else { Beep(2000, 20); } This works okay on my Vista laptop, but I've tried it on other laptops and some seem to play the sounds for less time (they sound more like clicks than beeps) or the sound doesn't play at all. So I have two questions here: Is there a more reliable beep function? Is there a (simple) way I can play a short .wav file or something similar instead (preferred solution).

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  • How to redirect sound to USB headset when plugged in?

    - by LM
    I often have to switch between audio output from my speakers and my headset (P5Q mobo with integrated sound and Microsoft headset). I've already got it so that when my headset is plugged in, sound will be played through it, and if it isn't, sound will play through my speakers. The problem is that if I have a game or similar program started while my headset is plugged in, if I unplug it, I will get no sound. Also, if I start the program with no headset, and plug it in, I get sound still through speakers. Is there any way to do this?

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  • How do I turn off the click sound when closing a tab in chrome?

    - by nos
    Every time I close a tab in Chrome, it makes a click sound. How do I turn off that sound? I reported that issue back in Oct 2010. The problem doesn't appear on all clients and the reason is still unclear. Common attempts at solving the issue include simply turning off the sound in Windows. But I would prefer to solve the problem at the source. Why is Chrome even triggering that sound to be played? And why is it delayed? The problem would be far less annoying if the sound could easily be related to the action taken. Installing the Chrome Toolbox and muting all tabs has no effect on this issue. When switching to a different Chrome user profile, the new user profile does exhibit the same issue.

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  • Can I run alsa and pulse side by side ? I think there is some problem with the alsa ! My ubunu login sound and alert sound are not working?

    - by Curious Apprentice
    I think I have Alsa driver installed. Pulse not working may be I dont have it installed. Not sure If I can run Pulse and Alsa. I had to configure each application prior to work which use pulse.(SMplayer by default select pulse. I had to change that) I know a little about these. So if the question is stupid then please help me. Smplayer always showing a cross(x) icon in front of speaker icon as it is disabled, though Im playing sound.

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  • record output sound in python

    - by aaronstacy
    i want to programatically record sound coming out of my laptop in python. i found PyAudio and came up with the following program that accomplishes the task: import pyaudio, wave, sys chunk = 1024 FORMAT = pyaudio.paInt16 CHANNELS = 1 RATE = 44100 RECORD_SECONDS = 5 WAVE_OUTPUT_FILENAME = sys.argv[1] p = pyaudio.PyAudio() channel_map = (0, 1) stream_info = pyaudio.PaMacCoreStreamInfo( flags = pyaudio.PaMacCoreStreamInfo.paMacCorePlayNice, channel_map = channel_map) stream = p.open(format = FORMAT, rate = RATE, input = True, input_host_api_specific_stream_info = stream_info, channels = CHANNELS) all = [] for i in range(0, RATE / chunk * RECORD_SECONDS): data = stream.read(chunk) all.append(data) stream.close() p.terminate() data = ''.join(all) wf = wave.open(WAVE_OUTPUT_FILENAME, 'wb') wf.setnchannels(CHANNELS) wf.setsampwidth(p.get_sample_size(FORMAT)) wf.setframerate(RATE) wf.writeframes(data) wf.close() the problem is i have to connect the headphone jack to the microphone jack. i tried replacing these lines: input = True, input_host_api_specific_stream_info = stream_info, with these: output = True, output_host_api_specific_stream_info = stream_info, but then i get this error: Traceback (most recent call last): File "./test.py", line 25, in data = stream.read(chunk) File "/Library/Python/2.5/site-packages/pyaudio.py", line 562, in read paCanNotReadFromAnOutputOnlyStream) IOError: [Errno Not input stream] -9975 is there a way to instantiate the PyAudio stream so that it inputs from the computer's output and i don't have to connect the headphone jack to the microphone? is there a better way to go about this? i'd prefer to stick w/ a python app and avoid cocoa.

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  • Graphing the pitch (frequency) of a sound

    - by Coronatus
    I want to plot the pitch of a sound into a graph. Currently I can plot the amplitude. The graph below is created by the data returned by getUnscaledAmplitude(): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(new FileInputStream(file))); byte[] bytes = new byte[(int) (audioInputStream.getFrameLength()) * (audioInputStream.getFormat().getFrameSize())]; audioInputStream.read(bytes); // Get amplitude values for each audio channel in an array. graphData = type.getUnscaledAmplitude(bytes, this); public int[][] getUnscaledAmplitude(byte[] eightBitByteArray, AudioInfo audioInfo) { int[][] toReturn = new int[audioInfo.getNumberOfChannels()][eightBitByteArray.length / (2 * audioInfo. getNumberOfChannels())]; int index = 0; for (int audioByte = 0; audioByte < eightBitByteArray.length;) { for (int channel = 0; channel < audioInfo.getNumberOfChannels(); channel++) { // Do the byte to sample conversion. int low = (int) eightBitByteArray[audioByte]; audioByte++; int high = (int) eightBitByteArray[audioByte]; audioByte++; int sample = (high << 8) + (low & 0x00ff); if (sample < audioInfo.sampleMin) { audioInfo.sampleMin = sample; } else if (sample > audioInfo.sampleMax) { audioInfo.sampleMax = sample; } toReturn[channel][index] = sample; } index++; } return toReturn; } But I need to show the audio's pitch, not amplitude. Fast Fourier transform appears to get the pitch, but it needs to know more variables than the raw bytes I have, and is very complex and mathematical. Is there a way I can do this?

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  • Help with dynamic-wind and call/cc

    - by josh
    I am having some trouble understanding the behavior of the following Scheme program: (define c (dynamic-wind (lambda () (display 'IN)(newline)) (lambda () (call/cc (lambda (k) (display 'X)(newline) k))) (lambda () (display 'OUT)(newline)))) As I understand, c will be bound to the continution created right before "(display 'X)". But using c seems to modify itself! The define above prints (as I expected) IN, X and OUT: IN X OUT And it is a procedure: #;2> c #<procedure (a9869 . results1678)> Now, I would expect that when it is called again, X would be printed, and it is not! #;3> (c) IN OUT And now c is not a procedure anymore, and a second invokation of c won't work! #;4> c ;; the REPL doesn't answer this, so there are no values returned #;5> (c) Error: call of non-procedure: #<unspecified> Call history: <syntax> (c) <eval> (c) <-- I was expecting that each invokation to (c) would do the same thing -- print IN, X, and OUT. What am I missing?

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  • Looping music with intro in XNA using SoundEffect

    - by Jordan Roher
    I have two sound files: Sound A is an 18 second intro designed to be played once Sound B is a 1 minute looping track I'd like to play Sound A once, then once Sound A is done, immediately play Sound B and keep looping Sound B until I tell it to stop. This is supposed to be looping town music in an RPG. I've tried doing this in code using just SoundEffect, but there's a tiny yet noticeable gap between the end of Sound A and the beginning of Sound B. Even if I put monitoring code watching Sound A's SoundEffectInstance.State in the Update() function, I haven't been able to start Sound B exactly when Sound A finishes so that it's seamless. I'd prefer to use SoundEffect because I can load WMA files rather than being stuck with WAVs in XACT.

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  • SoundManager2 has irregular latency

    - by Stefan Monov
    I'm playing some notes at regular intervals. Each one is delayed by a random number of milliseconds, creating a jarring irregular effect. How do I fix it? Note: I'm OK with some latency, just as long as it's consistent. Answers of the type "implement your own small SoundManager2 replacement, optimized for timing-sensitive playback" are OK, if you know how to do that :) but I'm trying to avoid rewriting my whole app in flash for now. For an example of app with zero audible latency see the flash-based ToneMatrix. Testcase (see it here live or get it in an zip): <head> <title></title> <script type="text/javascript" src="http://www.schillmania.com/projects/soundmanager2/script/soundmanager2.js"> </script> <script type="text/javascript"> soundManager.url = '.' soundManager.flashVersion = 9 soundManager.useHighPerformance = true soundManager.useFastPolling = true soundManager.autoLoad = true function recur(func, delay) { window.setTimeout(function() { recur(func, delay); func(); }, delay) } soundManager.onload = function() { var sound = soundManager.createSound("test", "test.mp3") recur(function() { sound.play() }, 300) } </script> </head> <body> </body> </html>

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  • Flash As3 Mute Button problems

    - by Lee
    Hey guys, I am trying to create a UI movie clip that can be used across different scenes. It uses variables from the root scope to determine states. When i press the mute button is works fine, however when i try to un-mute things go weird. Sometimes it takes 2 clicks to unmute, sometimes more. It seems random. Muting however seems to work first time.. Any ideas? Main Timeline: var mute:Boolean = false; var playerName = "Fred"; function setMute(vol) { var sTransform:SoundTransform = new SoundTransform(1,0); sTransform.volume = vol; SoundMixer.soundTransform = sTransform; } function toggleMuteBtn(event:Event) { if (mute) { // Sound On, Mute Off mute = false; setMute(1); ui_mc.muteCross_mc.visible = false; } else { // Sound Off, Mute On mute = true; setMute(0); ui_mc.muteCross_mc.visible = true; } } ui_mc Action Script: if (MovieClip(parent).mute == false) { muteCross_mc.visible = false; } mute_btn.addEventListener(MouseEvent.CLICK, MovieClip(parent).toggleMuteBtn);

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  • Are there any 5.1 surround audio switches on the market?

    - by thepurplepixel
    (Somewhat related to this question) I have a set of Logitech 5.1 surround speakers, which use 3 stereo 3.5mm TRS connectors (minijacks) to transfer the audio (the typical green/black/orange audio outputs). I have a Griffin Firewave hooked up to my MacBook Pro, and my desktop has a Realtek ALC889 audio chipset. I have looked for a way to, essentially, switch the speaker inputs between my Firewave and my desktop without having to disconnect the cables from one, route them around my desk, and plug them into the other. I'd love to have something like an old Belkin DB-25/LPT switch, but for these audio cables. Of course, purchasing one and soldering my own cables on the connection terminals is always an option, but, is there a reasonably priced 5.1 audio switch (or 3x stereo) on the market that will accomplish the simple task of switching audio outputs between two computers into a set of 5.1 speakers? Thanks in advance!

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  • Windows XP-64 loses audio sounds, drive letters... why?

    - by Ira Baxter
    Until sometime in early December, I had a wonderfully functioning XP-64 system. It was configured to auto download/install MS patches. I occassionally update the software on it, e.g. Open Office, Adobe Reader, Skype, but I don't fetch hundreds of tools or anything much beyond what I just mentioned. In December, suddenly my audio stopped, and drive letters assigned to various mount points on other machines quit being available. Apparantly, the services that support these (and some others) are now not starting up when I boot/login. There isn't anything obvious in the event log. If I manually restart the associated services, these facilities come back on line and work for awhile (a day) but pretty soon the problem reappears. I don't reboot very often, nor do I log out out much. Hints?

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