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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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  • What exactly does raw microphone data represent?

    - by esperantist
    I'm using PyAudio, a PortAudio wrapper for Python. I'm getting data from a microphone. Data which is represented by a continuous stream of bytes divided into chunks (of a size determined by me). I've tried to plot the signal, assuming the bytes represent the current signal amplitude, but I get an interesting image that I can't easily describe. ^^ It seems to be composed of two waves, one shifted from the other. What exactly do the particular bytes represent, and how does this change when I'm recording only one channel, instead of two? Any explanations, suggestions, code snippets, anything, very welcome! (I'm new at this.) Thanks!

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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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  • 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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  • Write wave files to memory in Java

    - by Cliff
    I'm trying to figure out why my servlet code creates wave files with improper headers. I use: AudioSystem.write( new AudioInputStream( new ByteArrayInputStream(memoryBytes), new AudioFormat(22000, 16, 1, true,false), memoryBytes.length ), AudioFileFormat.Type.WAVE, servletOutputStream ); taking a byte array from memory containing raw PCM samples and a servlet output stream that gets returned to the client. In the result I get a normal wave file but with zeros in the chunk size fields. Is the API broken? I would think that the size could be filled in using the size passed in the audio input stream. But now, after typing this out I'm thinking its not making this info available to the outer write() method on AudioSystem. It seems like the AudioSystem.write call needs a size parameter unless it is able to pull the size from the stream... which wouldn't work with an arbitrary sized stream. Does anyone know how to make this example work?

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  • Why doesnt R.raw.'songname' not work on android devices?

    - by James Rattray
    I have some media (Audio tracks) on an app... With file path 'R.raw.test' I use some code to get it into a mediaplayer... MediaPlayer.create(Textbox.this, R.raw.fly); And it works PERFECTLY on the Android Emulator... (Plays track on click of button) Can someone explain why, when I put it on my Archos (5 IT) it doesnt work at all? -As soon as the button is clicked, it crashes... Do you have to do something to file paths or what? Please help... Thanks alot... James

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  • Create mp3 previews from wav and aiff files

    - by August Lilleaas
    I would like to create a program that makes mp3s of the first 30 seconds of an aiff or wav file. I would also like to be able to choose location and length, such as the audio between 2:12 and 2:42. Are there any tools that lets me do this? Shelling out is OK. The application will run on a linux server, so it would have to be a tool that works on linux. I don't mind doing it in two steps - i.e. a tool that first creates the cutout of the aiff/wav, then pass it to a mp3 encoder.

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  • Architecture of chatroulette

    - by user317163
    Could somebody explain to me the architecture behind chatroulette? I was thinking about a similar project that would only implement Audio support (for starters). Is the best way to set this up a flash server? If so, how should I go about getting into flash, will I need flex 4? I have some beginner experience with c++, c# and java but I have never developed anything for the web. I was also wondering how the randomizer matches up the participants. How would you code something like this. Im obviously pretty clueless here and I'd greatly appreciate some advice regarding this problem -- I don't expect copy and paste solutions. It would just be nice to hear how you guys would tackle this problem. Thank you very much

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  • Java and gstreamer-java initialisation error

    - by Mark
    I am building a small app which will play streaming audio from the internet in java (mainly internet radio stations). I have decided to use the gstreamer-java library for the sound, which uses JNA. I would like to include a check in the code, to see whether the gstreamer library has been initialised. When I have left the "Gst.init()" code out (to mimic when the library has not been initialised correctly), the application throws out the following messages: (process:21888): GLib-GObject-CRITICAL **: /build/buildd/glib2.0-2.22.3/gobject/gtype.c:2458: initialization assertion failed, use IA__g_type_init() prior to this function (process:21888): GLib-CRITICAL **: g_once_init_leave: assertion `initialization_value != 0' failed The app calls the gstreamer-java library. The error messages appear but the thread continues to run, hogging the CPU. Is there any way to catch the error or to add a check to prevent it from happening? An alternative would be to put the "Gst.init()" in the main class, but I am not sure if this would always guarantee the gstreamer library is initialised.

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  • Detecting when Bluetooth is disabled on iOS5

    - by Non Umemoto
    I'm developing blog speaker app. I wanna pause the audio when bluetooth is disabled like iPod app. I thought it's not possible without using private api after reading this. Check if Bluetooth is Enabled? But, my customer told me that Rhapsody and DI Radio apps both support it. Then I found iOS5 has Core Bluetooth framework. https://developer.apple.com/library/ios/documentation/CoreBluetooth/Reference/CoreBluetooth_Framework/CoreBluetooth_Framework.pdf CBCentralManagerStatePoweredOff status seems like the one. But, the description says this api only supports Bluetooth 4.0 low energy devices. Did anyone try doing the same thing? I want to support current popular bluetooth headsets, or bluetooth enabled steering wheel on the car. I don't know if it's worth trying when it only supports some brand new bluetooth.

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  • Making of a "Babbelbox" where you can speak to for partys

    - by Spidfire
    Ive got a project to make for a party, its called in holland a "Babbelbox". its a computer with a webcam and microphone that can be used to make a kind of video log of everyone who wants to say something about the party. But the problem is that i dont know where to start. ive made a kind of video show system in c but i cant save any data to a good format so it wont jam my harddisk in one hour full. Requirements: Record video + audio Recoding has to start after pressing a button Good compression over the recorded videos (would be even better if it can to be read by final cut pro or premiere pro) Light wight programm would be nice but i could scale up the computer power

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  • For the iPad or iPhone, how do you control the system Volume? For example, have a button that mutes

    - by SolidSnake4444
    I would like to make a button in my iPad app (probably will be similar to iPhone apps) that when I push this button, all audio is muted, even when you exit the app. I don't see anyway that you can control the volume, although I'm sure other apps have that I have seen in the app store for the iPhone. I also read some places that doing this would reject you from the app store. How could I go about lowering, or highering the volume of the iPad from an app that works even when the app closes? Thank you!

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  • "Winamp style" spectrum analyzer

    - by cvb
    I have a program that plots the spectrum analysis (Amp/Freq) of a signal, which is preety much the DFT converted to polar. However, this is not exactly the sort of graph that, say, winamp (right at the top-left corner), or effectively any other audio software plots. I am not really sure what is this sort of graph called (if it has a distinct name at all), so I am not sure what to look for. I am preety positive about the frequency axis being base two exponential, the amplitude axis puzzles me though. Any pointers?

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  • Java stop MIDI playback

    - by user456268
    Hi I have java application which plays midi messages from sequence. I'm doing this using jfugue library. the problem is when I'm tryingto stop playback with stop button (which call sequencer.stop() and sequencer.close()) the last played note is sound all of rest time, and I can't stop it. So I'm asking about solution about stopping all audio and MIDI too! sound playback from java application. Notice: If you want propose just mute volume, you need to know that I want end-use will be able to press play button again and hear the sound again, so muting volumr will be not a solution, or explain please. Thank you!

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  • SoundManager + FFMPEG causing loud popping sound when streaming MP3s?

    - by David
    Hi there, I built an application that plays both uploaded original mp3 files, and copies that have been converted with FFMPEG. I am finding that in some cases the FFMPEG files have a horrible popping/clicking/screeching sound for a split second at startup (hear below). But when I analyze the file in an audio editor there is nothing there, so it seems to be either the browser or soundManager reacting badly to something in that file. Wondering if there is any way I can fix this either by adjusting FFMPEG settings, soundManager settings, or..... Any suggestions? I've uploaded the offending sound in the link below (before the music starts playing). Thanks for your help! Hear sound

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  • Python frequency detection

    - by Tsuki
    Ok what im trying to do is a kind of audio processing software that can detect a prevalent frequency an if the frequency is played for long enough (few ms) i know i got a positive match. i know i would need to use FFT or something simiral but in this field of math i suck, i did search the internet but didn not find a code that could do only this. the goal im trying to accieve is to make myself a custom protocol to send data trough sound, need very low bitrate per sec but im also very limited on the transmiting end so the recieving software will need to be able custom (cant use an actual hardware/software modem) also i want this to be software only (no additional hardware except soundcard) thanks alot for the help.

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