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  • FMOD surround sound openframeworks

    - by user1449425
    Ok, I hope I don't mess this up, I have had a look for some answers but can't find anything. I am trying to make a simple sampler in openframeworks using the FMOD sound player in 3D mode. I can make a single instance work fine (recording a new file using libsndfilerecorder and then playing it back and moving it in surround. However I want to have 8 layers of looping audio that I can record and replace one layer at a time in a live show. I get a lot of problems as soon as I have more than 1 layer. The first part of my question relates to the FMOD 3D modes, it is listener relative, so I have to define the position of my listener for every sound (I would prefer to have head relative mode but I cannot make this work at all. Again this works fine when I am using a single player but with multiple players only the last listener I update actually works. The main problem I have is that when I use multiple players I get distortion, and often a mix of other currently playing sounds (even when the microphone cannot hear them) in my new recordings. Is there an incompatability with libsndfilerecorder and FMOD? Here I initialise the players for (int i=0; i<CHANNEL_COUNT; i++) { lvelocity[i].set(1, 1, 1); lup[i].set(0, 1, 0); lforward[i].set(0, 0, 1); lposition[i].set(0, 0, 0); sposition[i].set(3, 3, 2); svelocity[i].set(1, 1, 1); //player[1].initializeFmod(); //player[i].loadSound( "1.wav" ); player[i].setVolume(0.75); player[i].setMultiPlay(true); player[i].play(); setupHold[i]==false; recording[i]=false; channelHasFile[i]=false; settingOsc[i]=false; } When I am recording I unload the file and make sure the positions of the player that is not loaded are not updating. void fmodApp::recordingStart( int recordingId ){ if (recording[recordingId]==false) { setupHold[recordingId]=true; //this stops the position updating cout<<"Start recording Channel " + ofToString(recordingId+1)+" setup hold is true \n"; pt=getDateName() +".wav"; player[recordingId].stop(); player[recordingId].unloadSound(); audioRecorder.setup(pt); audioRecorder.setFormat(SF_FORMAT_WAV | SF_FORMAT_PCM_16); recording[recordingId]=true; //this starts the libSndFIleRecorder } else { cout<<"Channel" + ofToString(recordingId+1)+" is already recording \n"; } } And I stop the recording like this. void fmodApp::recordingEnd( int recordingId ){ if (recording[recordingId]=true) { recording[recordingId]=false; cout<<"Stop recording" + ofToString(recordingId+1)+" \n"; audioRecorder.finalize(); audioRecorder.close(); player[recordingId].loadSound(pt); setupHold[recordingId]=false; channelHasFile[recordingId]=true; cout<< "File recorded channel " + ofToString(recordingId+1) + " file is called " + pt + "\n"; } else { cout << "Sorry track" + ofToString(recordingId+1) + "is not recording"; } } I am careful not to interrupt the updating process but I cannot see where I am going wrong. Many Thanks

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  • Can I disable UIPickerView scroll sound?

    - by cocoaholic
    Hi, I want to disable the annoying clicks that the UIPickerView generates upon scrolling up and down. Is there a way to do this? I want to play short sounds for each item that the picker view lands upon. It gets ruined by the built in sound. I understand that the picker sounds can be turned off globally by switching off the keyboard sounds in iPhone/iPod settings. But is there a way to programatically do this? Any help will be much appreciated! Thanks

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  • loading mp3 from file using random access to flash.media.Sound

    - by Irfan Mulic
    We are migrating application from Delphi to Flex (Air) that plays mp3 files from random access big file. it has positions and sizes to extract mp3 data to FileStream-MemoryStream and then we use bass.dll to play it from memory stream. Now I have to play those same mp3's in flex but I am not sure how... I was reading something similar for reading/writing data using ByteArray from here but how to apply it to flash.media.Sound ? http://livedocs.adobe.com/flex/3/html/help.html?content=ByteArrays_2.html Any help?

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  • How to set notification sound volume programatically?

    - by Vitalii Korsakov
    I have this method in my main activity private void beep() { AudioManager manager = (AudioManager) getSystemService(Context.AUDIO_SERVICE); manager.setStreamVolume(AudioManager.STREAM_NOTIFICATION, 0, AudioManager.FLAG_SHOW_UI + AudioManager.FLAG_PLAY_SOUND); Uri notification = RingtoneManager .getDefaultUri(RingtoneManager.TYPE_NOTIFICATION); Ringtone r = RingtoneManager.getRingtone(getApplicationContext(), notification); r.play(); } As I understand, notification sound volume should be regulated by STREAM_NOTIFICATION. But notification always plays with the same volume despite that volume number in setStreamVolume method. Why is that?

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  • What trick will give most reliable/compatible sound alarm in a browser window for most browsers

    - by Dirk Paessler
    I want to be able to play an alarm sound using Javascript in a browser window, preferably with the requirement for any browser plugins (Quicktime/Flash). I have been experimenting with the tag and the new Audio object in Javascript, but results are mixed: As you can see, there is no variant that works on all browsers. Do I miss a trick that is more cross-browser compatible? This is my code: // mp3 with Audio object var snd = new Audio("/sounds/beep.mp3");snd.play(); // wav with Audio object var snd = new Audio("/sounds/beep.wav");snd.play(); // mp3 with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.mp3" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); // wav with EMBED tag $("#alarmsound").empty().append ('<embed src="/sounds/beep.wav" autostart="true" loop="false" '+ 'volume="100" hidden="true" width="1" height="1" />'); }

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  • Cannot play a recorded sound on device.

    - by B_
    I'm using the exact code from the iPhone Application Programming Guide Multimedia Support to use AVAudioRecorder to record a file to the disk and then AVAudioPlayer to load and play that file. This is working fine in the simulator but is not working on the device. The file gets loaded (we can see the NSTimeInterval) but does not play (play returns false). After it didn't work with the sample code from the website, we tried changing to a bunch of different codecs with no success. And of course, the sound is on. Thanks a bunch.

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  • Android: Playing sound when button clicked?

    - by gazeebo
    Hi all, I'm trying to play a sound file when a button is clicked but keeps getting an error. The error is "The method create(Context, int) in the type MediaPlayer is not applicable for the arguments (new View.OnClickListener(){}, int)" Here's my code: @Override public void onClick(View v) { // TODO Auto-generated method stub Button zero = (Button)this.findViewById(R.id.btnZero); zero.setOnClickListener(new View.OnClickListener() { @Override public void onClick(View v) { // TODO Auto-generated method stub mp = MediaPlayer.create(this, R.raw.mamacita_zero); } }); } Any help or tips would be appreciated. Thnx!

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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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  • 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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  • 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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  • 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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  • 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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