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  • Mixing two wav music files of different size

    - by iphoneDev
    Hi, I want to mix audio files of different size into a one single .wav file. There is a sample through which we can mix files of same size [(http://www.modejong.com/iOS/#ex4 )(Example 4)]. I modified the code to get the mixed file as a .wav file. But I am not able to understand that how to modify this code for unequal sized files. If someone can help me out with some code snippet,i'll be really thankful.

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  • how to get the css keys and values for any html tag

    - by artsince
    I would like to dump all css key/value pairs for an html tag. In particular, I would like to learn the css properties for <audio> tag, so I can try to customize the look. document.getElementById('myaudio').style returns a CSSStyleDeclaration object but length returns 0 and I cannot figure out to iterate over the key/value pairs. Thank you

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  • MP3 and OGG tags in PHP

    - by Quamis
    Except http://us3.php.net/manual/en/book.ktaglib.php and http://getid3.sourceforge.net/ does anyone know of any other way to work from PHP with tags on audio files? I need to read and write them, and KTagLib seems a little too much for the job, and also don't really get the documentation, and getID3 seems to only write ID3v1 tags.

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  • How do I tell if the master volume is muted?

    - by John_Sheares
    I am using the following to mute/unmute the master audio on my computer. Now, I am looking for a way to determine the mute state. Is there a just as easy way to do this in C#? private const int APPCOMMAND_VOLUME_MUTE = 0x80000; private const int WM_APPCOMMAND = 0x319; [DllImport("user32.dll")] public static extern IntPtr SendMessageW(IntPtr hWnd, int Msg, IntPtr wParam, IntPtr lParam);

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  • Java writes bad wave files

    - by Cliff
    I'm writing out wave files in Java using AudioInputStream output = new AudioInputStream(new ByteArrayInputStream(rawPCMSamples), new AudioFormat(22000,16,1,true,false), rawPCMSamples.length) AudioSystem.write(output, AudioFileFormat.Type.WAVE, new FileOutputStream('somefile.wav')) And I get what appears to be corrupt wave files on OSX. They won't play from Finder however using the same code behind a servlet writing directly to the response stream and setting the Content-Type to audio/wave seems to play fine in quicktime. What gives?

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  • Play an AudioBufferSourceNode twice?

    - by alltom
    Should I be able to use the same AudioBufferSourceNode to play a sound multiple times? For some reason, calling noteGrainOn a second time doesn't play audio, even with an intervening noteOff. This code only plays the sound once: var node = audioContext.createBufferSource() node.buffer = audioBuffer node.connect(audioContext.destination) var now = audioContext.currentTime node.noteGrainOn(now, 0, 2) node.noteOff(now + 2) node.noteGrainOn(now + 3, 0, 2) node.noteOff(now + 5)

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  • how to save Audiorecorded file to another location? i m trying but i got exception...

    - by rakesh-bhatt99
    NSString recordFile = [NSTemporaryDirectory() stringByAppendingPathComponent: (NSString)inRecordFile]; NSArray *docPaths=NSSearchPathForDirectoriesInDomains(NSDocumentDirectory,NSUserDomainMask,YES); NSString docDir=[[docPaths objectAtIndex:0]stringByAppendingPathComponent: (NSString)inRecordFile]; url = CFURLCreateWithString(kCFAllocatorDefault, (CFStringRef)docPaths, NULL); // create the audio file XThrowIfError(AudioFileCreateWithURL(url, kAudioFileCAFType, &mRecordFormat, kAudioFileFlags_EraseFile, &mRecordFile), "AudioFileCreateWithURL failed"); CFRelease(url);

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  • Access to iTunes Sound Check Results on iPhone

    - by Baldoph
    I would like to propose to the user some songs whose volume doesn't exceed a certain level. Is there any way to access to the results of the 'Sound Check' option, from the iPhone ? If not, do you know if I can calculate that with the audio tools in the iPhone SDK ? Thanks a lot.

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  • Mobile opera have background sound support?

    - by Mark
    I make browser/html/js games. One of my biggest pains in the arse is the lack of background sound support in mobile safari. This lack of support makes high value games pretty much impossible. Does anyone know if opera mini supports html5 audio, or any mobile browser for that matter. If not, what are some alternatives methods.

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  • AVFoundation: Video to OpenGL texture working - How to play and sync audio?

    - by j00hi
    I've managed to load a video-track of a movie frame by frame into a OpenGL texture with AVFoundation. I followed the steps described in the answer here: iOS4: how do I use video file as an OpenGL texture? and took some code from the GLVideoFrame sample from WWDC2010 which can be downloaded here: http://bit.ly/cEf0rM How do I play the audio-track of the movie synchronously to the video. I think it would not be a good idea to play it in a separate player, but to use the audio-track of the same AVAsset. AVAssetTrack* audioTrack = [[asset tracksWithMediaType:AVMediaTypeAudio] objectAtIndex:0]; I retrieve a videoframe and it's timestamp in the CADisplayLink-callback via CMSampleBufferRef sampleBuffer = [self.readerOutput copyNextSampleBuffer]; CMTime timestamp = CMSampleBufferGetPresentationTimeStamp( sampleBuffer ); where readerOutput is of type AVAssetReaderTrackOutput* How to get the corresponding audio-samples? And how to play them? Edit: I've looked around a bit and I think, best would be to use AudioQueue from the AudioToolbox.framework using the approach described here: AVAssetReader and Audio Queue streaming problem There is also an audio-player in the AVFoundation: AVAudioPlayer. But I don't know exactly how I should pass data to it's initWithData-initializer which expects NSData. Furthermore I don't think it's the best choice for my case because a new AVAudioPlayer-instance would have to be created for every new chunk of audio samples, as I understand it. Any other suggestions? What's the best way to play the raw audio samples which i get from the AVAssetReaderTrackOutput?

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  • issue getting dynamic Config parameter in Grails taglib

    - by Mick Knutson
    I have a dynamic config parameter I want to get like: String srcProperty = "${attrs ['src']}.audio" + ((attrs['locale'])? "_${attrs['locale']}" : '') assert srcProperty == "prompt.welcomeMessageOverrideGreeting.audio" where my config has: prompt{ welcomeMessageOverrideGreeting { audio = "/en/someFileName.wav" txt = "Text alternative for /en/someFileName.wav" audio_es = "/es/promptFileName.wav" txt_es = "Texto alternativo para /es/someFileName.wav" } } While this works fine: String audio = "${config.prompt.welcomeMessageOverrideGreeting.audio}" and: assert "${config.prompt.welcomeMessageOverrideGreeting.audio}" == "/en/someFileName.wav" I can not get this to work: String audio = config.getProperty("prompt.welcomeMessageOverrideGreeting.audio")

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  • How to stop XBox Music commercials while playing music in collection?

    - by bill weaver
    So i decided to give Xbox Music a try, and played a song in my collection. Then moved to another song and a commercial started playing. Huh? Searching revealed others with this problem, but i didn't see any answers. Yes, i know Xbox Music plays commercials when streaming free music that you don't own, but this is mp3 music i own, on my hard drive, in my collection. MS claims "You’ll never get ads when you’re playing MP3s that are on your PC or when you’re playing music you bought from Xbox Music." (FWIW, this is running on Windows 8.1 Pro, though the problem seems to have been reported last year too, so it's probably not a new issue.)

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  • What is making iTunes stop playing when my computer idles?

    - by OwenP
    I've got a rare weekend with nothing to do, so I'm getting some housework done. I have iTunes playing for some background noise. Every 20 minutes or so, it just stops playing; if I move the mouse it starts again. I'm on Windows 7 64-bit. My power settings have my monitor turning off at 10 minutes and hard drives at 20. Both sleep and hibernate are disabled. "Aha!", you say, "Clearly when the hard drive is turned off iTunes is stopping!" Not so. I fiddled with the settings and changed them to make the hard drive sleep in 5 minutes, and iTunes kept playing for the 7 minutes I watched it. I'm currently trying to see what happens if I set the hard drive to never turn off, but I'd prefer to leave it at 20 minutes to save minor amounts of energy. What other settings could be the culprit?

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  • FMOD.net streaming, callback and exinfo parameters

    - by Tesserex
    I posted a question on gamedev about how to play nsf files (NES console music) in FMOD. It didn't get any results, but since then I made some progress. I decided that the easiest method was just to compile an existing player into a dll and then call it from C# to populate my buffer. The problem now is getting it to sound right, and making sure all my paremeters are correct. Here are the facts so far: The nsf dll is dealing with shorts, so the data is PCM16. The sample nsf I'm using has a playback rate of 60 Hz. Just for playing around now, I'm using a frequency of 48000. Based on 2 and 3, the dll calculates a necessary buffer size of 48000 / 60hz = 800. This means it will render 800 shorts worth of buffer for every simulated NES frame. I've so far got my C# code to play the nsf, at the correct pitch and tempo, but it's very grainy / fuzzy, which I'm attributing to the fact that the FMOD read callback is giving a data length of 1600, whereas I should be expecting 800. I've tried playing around with all the numbers and it either crashes, or the music changes pitch, tempo, or both. Here's some of my C# code: uint channels = 1, frequency = 48000; FMOD.MODE mode = (FMOD.MODE.DEFAULT | FMOD.MODE.OPENUSER | FMOD.MODE.LOOP_NORMAL); FMOD.Sound sound = new FMOD.Sound(); FMOD.CREATESOUNDEXINFO ex = new FMOD.CREATESOUNDEXINFO(); ex.cbsize = Marshal.SizeOf(ex); ex.fileoffset = 0; ex.format = FMOD.SOUND_FORMAT.PCM16; // does this even matter? It doesn't change my results as long as it's long enough for one update ex.length = frequency; ex.numchannels = (int)channels; ex.defaultfrequency = (int)frequency; ex.pcmreadcallback = pcmreadcallback; ex.dlsname = null; // eventually I will calculate this with frequency / nsf hz, but I'm just testing for now ex.decodebuffersize = 800; // from the dll load_nsf_file("file.nsf", 8, (int)frequency); // 8 is the track number to play var result = system.createSound( (string)null, (mode | FMOD.MODE.CREATESTREAM), ref ex, ref sound); channel = new FMOD.Channel(); result = system.playSound(FMOD.CHANNELINDEX.FREE, sound, false, ref channel); private FMOD.RESULT PCMREADCALLBACK(IntPtr soundraw, IntPtr data, uint datalen) { // from the dll process_buffer(data, (int)800); // if I use datalen, it usually crashes (I can't get datalen to = 800 safely) return FMOD.RESULT.OK; } So here are some of my questions: What is the relationship between exinfo.decodebuffersize, frequency, and the datalen parameter of the read callback? With this code sample, it's coming in as 3200. I don't know where that factor of 4 between it and the decodebuffersize comes from. Is datalen in the callback referring to number of bytes, or shorts? The process_buffer function takes a short array and its length. I would expect fmod is talking about shorts as well because I told it PCM16. Maybe my playback quality is bad for some totally different reason. If so I have no idea where to begin solving that. Any ideas there?

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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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  • The fastest way to encode image+audio for Youtube from command line?

    - by Pavel Vlasov
    I have an mp3 and image and I want to make a simple clip to upload onto Youtube. Is there a fast solution? If video formats are so bad designed, then maybe it is possible to use a prerendered video-only clip? This works good except it takes as much time as the audio lasts: ffmpeg -loop_input -r ntsc -i "%IMAGE%" -i "%AUDIO%" -r 1 -acodec copy -shortest -re -force_fps "%VIDEO%" This takes a second but results in a black screen video that is successfully played by a desktop video player but not acceptable by Youtube: ffmpeg -i "%IMAGE%" -i "%AUDIO%" -acodec copy "%VIDEO%" Windows 7. Preserving audio quality is preferred over video quality.

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  • How do I merge MP4 files without audio going out of sync?

    - by djangofan
    Is there a tool I can use that can merge MP4 files without throwing the audio out of sync? I generated some MP4 files from a DVD using AVIDemux but whatever tool I try to use always ends up throwing the audio out of sync with the video. The further you get into the video the further off-sync the audio is. By themselves the MP4/AAC videos have perfect audio-video sync. later tonight i might try http://www.headbands.com/gspot/ to examine the file before and after to see if anything changed in the media format.

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  • How do I add another audio stream to an MP4 file?

    - by RandomEngy
    I've got an MP4 video file and I want to add another AAC audio track to it. I've tried YAMB and MeGUI (frontends for MP4Box) and it plays correctly in Zoom Player, but it picks the wrong track in WMP and plays both at once in Quicktime. I think this might have to do with designating the default audio track somehow. Does anyone know how to specify the default audio track with YAMB/MeGUI or know of another way of adding a track to an MP4 file?

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  • How to export or view audio file references in a PDF?

    - by redshift
    I have a an interactive PDF file that is over 90+ pages long. Each page is a map with city names that contains a Spanish pronunciation of that city in a .wav file. I'd say there are about 10-15 audio files for each map which comes out to 1000+ audio files. Is there a way to extract/export a list of the sound file names associated with each map? I tried to save the PDF to an HTML file, but it only exported images and text, and because the audio files were embedded in the PDF, the file names did not carry over to the HTML file. Any other ideas? I need to see what audio file goes with what map/page.

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  • how to prevent MPMoviePlayer controls from hiding

    - by huevos de oro
    I am trying to implement a custom MPMoviePlayer to play mp3 audio. I have got it working in portrait mode along with an overlay window over the native controls - thanks to other stackoverflow posts. The current issue is the song progress control shows up when the media window opens (blue bar taking up the first 40 odd pixels), but seems to disappear when the song starts leaving a white bar. It will then re-appear when touching the area, so functionally works fine. I would like to find a way to ensure the controls always stay visible but have not found an appropriate property in the reference. Ideally I would like to have my custom control to replace the default, more because I would like to change the position that the look and feel. This being said, I understand it is not possible as the current position in the song from a MPMoviePlayer cannot be accessed.

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  • Why is AudioOutputUnitStart freezing my app in iOS 4?

    - by Luke
    Hi guys, I have an audio app which uses the RemoteIO AudioUnit. It works fine on iPhone, iPad, and any flavor of the simulator on 3.2, but when it hits AudioOutputUnitStart (), it freezes. I get the message "AddRunningClient starting device on non-zero client count" in the console, which I'm not sure how to resolve. I stop the unit and dispose of the AudioComponent every time the app closes. The app works fine the first time I run after restarting everything, but freezes every time after that. What's strange is there are no error messages - just an unresponsive interface and a frozen line of code. Thanks for your help. Luke

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  • Alternate play / pause button for WordPress wpaudio soundmanager plugin

    - by j-man86
    Hello! I am using the wpaudio plugin to convert mp3 links into a javascript/flash audio player. My problem is that I use this plugin in two areas on my site: one on a black background, and one on a white background. I need to use an alternate set of play/pause buttons for each page (white buttons for the black background and vice versa). I am at a total loss on how to do this. I need to some how incorporate a "if page is..." statement into the wpaudio.js but I don't know how to do this with jQuery. Can anyone help? Thanks so much!

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  • Manipulating multi-track ogg files programatically

    - by Chad Birch
    I'm planning to create a program for manipulating multi-track OGG files, but I don't have any experience with the relevant libraries, so I'm looking for recommendations about which language/library to use for this. I don't really have any preference for the language, I'll happily code it in C, C#, Python, whatever makes things the easiest (or even possible). Perhaps it's even a possibility to automate Audacity somehow? In terms of requirements, I'm not looking for anything particularly fancy. It will probably be a command-line program, I don't need to be able to play the audio, draw image representations of the waveforms, etc. The program will basically be used as a converter, but I need to do some processing before outputting. That is, I need the ability to programatically remove some tracks, set panning per-track, change track volumes, etc. Nothing too complex, just some basic processing, and then output the result in either MP3 or a format easily converted to MP3, such as WAV. Any suggestions or general information would be appreciated, thanks.

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  • AudioQueueOfflineRender returning empty data

    - by hyn
    I'm having problems using AudioQueueOfflineRender to decode AAC data. When I examine the buffer after the call, it is always filled with empty data. I made sure the input buffer is valid and packet descriptions are provided. I searched and found that a few others have had the same problem: http://lists.apple.com/archives/Coreaudio-api/2008/Jul/msg00119.html Also, the inTimestamp argument doesn't make sense to me. Why should the renderer care where in the audio the beginning of the buffer corresponds to? The function throws an error if I pass in NULL, so I pass in the timestamp anyway.

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  • How to extract semi-precise frequencies from a WAV file using Fourier Transforms

    - by Seisatsu
    Let us say that I have a WAV file. In this file, is a series of sine tones at precise 1 second intervals. I want to use the FFTW library to extract these tones in sequence. Is this particularly hard to do? How would I go about this? Also, what is the bast way to write tones of this kind into a WAV file? I assume I would only need a simple audio library for the output. My language of choice is C

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