Search Results

Search found 5304 results on 213 pages for 'audio streaming'.

Page 81/213 | < Previous Page | 77 78 79 80 81 82 83 84 85 86 87 88  | Next Page >

  • Can I play any Buffer only once at a given time?

    - by mystify
    From the OpenAL documentation: The basic OpenAL objects are a Listener, a Source, and a Buffer. There can be a large number of Buffers, which contain audio data. Each buffer can be attached to one or more Sources My problem is, that I have one sound file which I need to play multiple times per second, at the same time. The sound is 2 seconds long. So it will overlap. Would I need multiple filled buffers for this (= multiple times that sound in memory)? If I would attach one Buffer to multiple Sources, would I be able to play the sound 10 times, overlapping itself, with just one copy in memory? Or would I still have to deal with 10 copies of that sound in memory?

    Read the article

  • Stuggling with webkit-transition in javascript

    - by Mungbeans
    I've tried a few variations of using webkit-transition that I've found from googling but I've not been able to get any to work. I have some audio controls that I make appear on a click event, they appear suddenly and jerky so I want to fade them in. The target browser is iOS so I am trying webkit extensions. This is what I currently have: <div id = "controls"> <audio id = "audio" controls></audio> </div> #controls { position:absolute; top: 35px; left:73px; height: 20px; width: 180px; display:none; } #audio { opacity:0.0; } audio.src = clip; audio.addEventListener('pause', onPauseOrStop, false); audio.addEventListener('ended', onPauseOrStop, false); audio.play(); audioControls.style.display = 'block'; audio.style.setProperty("-webkit-transition", "opacity 0.4s"); audio.style.opacity = 0.7; The documentation for webkit-transition says it takes effect on a change in the property, so I was assuming changing style.opacity in the last line would kick it off. The controls appear with an opacity of 0.7 but I want it to fade in and that animation isn't happening. I also tried this: #audio { opacity:0.0; -webkit-transition-property: opacity; -webkit-transition-duration: 1s; -webkit-timing-function: ease-in; } Also tried audio.style.webkitTransition = "opacity 1.4s"; from this posting How to set CSS3 transition using javascript? I can't get anything to work, I'm testing on iOS, Safari desktop and Chrome. Same non result on all of them.

    Read the article

  • Playing an InputStream video in Blackberry JDE.

    - by Jenny
    I think I'm using InputStream incorrectly with a Blackberry 9000 simulator: I found some sample code, http://www.blackberry.com/knowledgecenterpublic/livelink.exe/fetch/2000/348583/800332/1089414/How%5FTo%5F-%5FPlay%5Fvideo%5Fwithin%5Fa%5FBlackBerry%5Fsmartphone%5Fapplication.html?nodeid=1383173&vernum=0 that lets you play video from within a Blackberry App. The code claims it can handle HTTP, but it's taken some fandangling to get it to actually approach doing so: http://pastie.org/609491 Specifically, I'm doing: StreamConnection s = null; s = (StreamConnection)Connector.open("http://10.252.9.15/eggs.3gp"); HttpConnection c = (HttpConnection)s; InputStream i = c.openInputStream(); System.out.println("~~~~~I have a connection?~~~~~~" + c); System.out.println("~~~~~I have a URL?~~~~" + c.getURL()); System.out.println("~~~~~I have a type?~~~~" + c.getType()); System.out.println("~~~~~I have a status?~~~~~~" + c.getResponseCode()); System.out.println("~~~~~I have a stream?~~~~~~" + i); player = Manager.createPlayer(i, c.getType()); I've found that this is the only way I can get an InputStream from an HTTPConnection without causing a: "JUM Error 104: Uncaught NullPointer Exception". (That is, the casting as a StreamConnection, and THEN as an HttpConnection stops it from crashing). However, I'm still not streaming video. Before, a stream wasn't able to be created (it would crash with the null pointer exception). Now, a stream is being made, the debugger claims it's begining to stream video from it...and nothing happens. No video plays. The app doesn't freeze, or crash or anything. I can 'pause' and 'play' freely, and get appropriate debug messages for both. But no video shows up. If I'm playing a video stored locally on the blackberry, everything is fine (it actually plays the video), so I know the Player itself is working fine, I"m just wondering if maybe I have something wrong with my stream? The API says the player can take in an InputStream. Is there a specific kind it needs? How can I query my inputstream to know if it's valid? It existing is further than I've gotten before. -Jenny Edit: I'm on a Blackberry Bold simulator (9000). I've heard that some versions of phones do NOT stream video via HTTP, however, the Bold does. I have yet to see examples of this though. When I go to the internet and point at a blackberry playable video, it attempts to stream, and then asks me to physically download the file (and then plays fine once I download). Edit: Also, I have a physical blackberry Bold, as well, but it can't stream either (I've gone to m.youtube.com, only to get a server/content not found error). Is there something special I need to do to stream RTSP content?

    Read the article

  • Core Audio - CARIngBuffer

    - by tech74
    Hi, Im looking at using the CARingBuffer in iPhone SDK 3.1 Developer\Extras\CoreAudio\PublicUtility, however was a little puzzled about some of its methods. Firstly this will only make sense really to anyone who's used this class For example the GetTimebounds,SetTimeBounds, ClipTimeBounds functions what are these actually doing? Also when using it, i get crashes caused by example this method in the main Fetch method - ZeroABL(abl, 0, destStartOffset * mBytesPerFrame); CARingBufferError CARingBuffer::Fetch(AudioBufferList *abl, UInt32 nFrames, SampleTime startRead) { SampleTime endRead = startRead + nFrames; SampleTime startRead0 = startRead; SampleTime endRead0 = endRead; SampleTime size; CARingBufferError err = ClipTimeBounds(startRead, endRead); if (err) return err; size = endRead - startRead; SInt32 destStartOffset = startRead - startRead0; if (destStartOffset 0) { ZeroABL(abl, 0, destStartOffset * mBytesPerFrame); } Here the destStartOffset has become larger than the size of the abl Bufferlist so when a memset is done it exceeds the boundaries of the abl Bufferlist causing the crash. Why hasn't this class got checks in to prevent this.

    Read the article

  • Audio Recording and Playback

    - by Siva
    Hi, I am new to iphone development. In my app, I want to record a voice and play the recorded voice. Now I am trying to do via speak here sample code, but i feel it is too hard to understand with AudioToolbox framework. Somebody saying AudioToolbox framework is too difficult to implement it. is there any other sample with other than AudioToolbox framework or which way is best to do that? Please help me!

    Read the article

  • Audio looping in Objective-C/iPhone

    - by Neurofluxation
    So, I'm finishing up an iPhone App. I have the following code in place to play the file: while(![player isPlaying]) { totalSoundDuration = soundDuration + 0.5; //Gives a half second break between sounds sleep(totalSoundDuration); //Don't play next sound until the previous sound has finished [player play]; //Play sound NSLog(@" \n Sound Finished Playing \n"); //Output to console } For some reason, the sound plays once then the code loops and it outputs the following: Sound Finished Playing Sound Finished Playing Sound Finished Playing etc... This just repeats forever, I don't suppose any of you lovely people can fathom what could be the boggle? Cheers!

    Read the article

  • detecting pauses in a spoken word audio file using pymad, pcm, vad, etc

    - by james
    First I am going to broadly state what I'm trying to do and ask for advice. Then I will explain my current approach and ask for answers to my current problems. Problem I have an MP3 file of a person speaking. I'd like to split it up into segments roughly corresponding to a sentence or phrase. (I'd do it manually, but we are talking hours of data.) If you have advice on how to do this programatically or for some existing utilities, I'd love to hear it. (I'm aware of voice activity detection and I've looked into it a bit, but I didn't see any freely available utilities.) Current Approach I thought the simplest thing would be to scan the MP3 at certain intervals and identify places where the average volume was below some threshold. Then I would use some existing utility to cut up the mp3 at those locations. I've been playing around with pymad and I believe that I've successfully extracted the PCM (pulse code modulation) data for each frame of the mp3. Now I am stuck because I can't really seem to wrap my head around how the PCM data translates to relative volume. I'm also aware of other complicating factors like multiple channels, big endian vs little, etc. Advice on how to map a group of pcm samples to relative volume would be key. Thanks!

    Read the article

  • Audio Reminders

    - by abhishek mishra
    Hi , I am developing a reminder application. A part of it is to have voive notes as reminders. On click of voice notes button i want to start the inbuilt voice recorder. How do i go ahead for it ? Also once it starts i want to retrieve the path where it gets stored so that it can be played automatically on the day the timeline is reached. Is it possible ?

    Read the article

  • HTML5 iPhone Safari Mobile visualize something rather than quicktime symbol when creating an audio t

    - by Antonio Murgia
    I'm writing a very simple webpage in html5 for iPhone. the page is this one Not Working Everything works but in the page from the iPhone i see the quicktime logo with a slash on it and if i tap on it the player shows up the play button and in the background there is the quicktime logo. is it possible to replace the logos with a personal image? thank you in advance.

    Read the article

  • do I need to close an audio Clip?

    - by Michael
    have an application that processes real-time data and is supposed to beep when a certain event occurs. The triggering event can occur multiple times per second, and if the beep is already playing when another event triggers the code is just supposed to ignore it (as opposed to interrupting the current beep and starting a new one). Here is the basic code: Clip clickClip public void prepareProcess() { super.prepareProcess(); clickClip = null; try { clipFile = new File("C:/WINDOWS/Media/CHIMES.wav"); ais = AudioSystem.getAudioInputStream(clipFile); clickClip = AudioSystem.getClip(); clickClip.open(ais); fileIsLoaded = true; } catch (Exception ex) { clickClip = null; fileIsLoaded = false; } } public void playSound() { if (fileIsLoaded) { if ((clickClip==null) || (!clickClip.isRunning())) { try { clickClip.setFramePosition(0); clickClip.start(); } catch (Exception ex) { System.out.println("Cannot play click noise"); ex.printStackTrace(); } } } The prepareProcess method gets run once in the beginning, and the playSound method is called every time a triggering event occurs. My question is: do I need to close the clickClip object? I know I could add an actionListener to monitor for a Stop event, but since the event occurs so frequently I'm worried the extra processing is going to slow down the real-time data collection. The code seems to run fine, but my worry is memory leaks. The code above is based on an example I found while searching the net, but the example used an actionListener to close the Clip specifically "to eliminate memory leaks that would occur when the stop method wasn't implemented". My program is intended to run for hours so any memory leaks I have will cause problems. I'll be honest: I have no idea how to verify whether or not I've got a problem. I'm using Netbeans, and running the memory profiler just gave me a huge list of things that I don't know how to read. This is supposed to be the simple part of the program, and I'm spending hours on it. Any help would be greatly appreciated! Michael

    Read the article

  • Learn mp3 format and audio signal processing

    - by Shankhoneer Chakrovarty
    I am trying to learn the following things: How mp3 file looks like internally? I found this: http://mpgedit.org/mpgedit/mpeg_format/MP3Format.html but it seems old. Is there any recent changes to the format? I couldnt find any. How to open a mp3 file in java and look for bytes? I tried using audiostream but I am getting a lot of zeros and signed short integers which nowhere resemble the header/body format as mentioned in the above link. Am I wrong in interpreting the bytes? How to get amplitude, frequency and pitch of a mp3 file? No idea. Can you please suggest some book or tutorial? Can you please help me in getting the solution for the above questions? I am sorry if some questions appear to be naive, I am a just begun to learn mp3. Thanks

    Read the article

  • How to change default audio input device programatically

    - by f34r
    I am looking for a way to set/change default input device inside my application. I have several different recording devices and it is very anoying to go into the control panel and change default recording device. I was looking around and I did not find anything that could help me with the problem. Application is written in c# and it is targeted for Windows Vista / Windows 7.

    Read the article

  • Synchronizing Java Visualizer Audio and Visual

    - by Matt
    I've run into a problem creating a visualizer for .mp3 files in Java. My goal is to create a visualization that runs in time with the .mp3 file being played. I can currently visualize an .mp3 OR play it, but not both at the same time. I am using libraries which may make this trickier than necessary. I currently: Read in the .mp3 as a FileInputStream. a) Convert the FileInputStream into a Bitstream and run the Visualizer OR b) Pass the FileInputStream to a library Play method where it converts it into a Bitstream, decodes it, and plays it. I am using the JLayer library to play and decode the .mp3. My question is: how do I synchronize the two actions so that I can run both at the same time AND they line up (so my visualizations correspond to the changing frequencies). This implies that they finish at the same time as well.

    Read the article

  • AxWindowsMediaPlayer does not play audio/video from url ?

    - by Madhup
    HI, I am using activeXMediaPlayer to play files from url but each time I pass a url to it shows the message , "either the file is corrupted or player does not support the file format u are playing." But when i run the same url on browser the file is downloaded and this downloaded file can be played on the media player. I am not able to find out what the problem is . Because the same cod plays the local file and the downloaded files but not file from url Although the same code worked few months ago for the urls So is this my fault or some server related issues can affect this thing. Please help me I am in big trouble. Regards, Madhup

    Read the article

  • Audio/Voice Visualization

    - by Neurofluxation
    Hey you Objective-C bods. Does anyone know how I would go about changing (transforming) an image based on the input from the Microphone on the iPhone? i.e. When a user speaks into the Mic, the image will pulse or skew. Thanking you!!

    Read the article

  • How to release audio properly? (AVAudioPlayer)

    - by Aluminum
    Hello everyone! I need help with my iOS application ^^,. I want to know if I'm releasing AVAudioPlayer correctly. MyViewController.h #import <UIKit/UIKit.h> @interface MyViewController : UIViewController { NSString *Path; } - (IBAction)Playsound; @end MyViewController.m #import <AVFoundation/AVAudioPlayer.h> #import "MyViewController.h" @implementation MyViewController AVAudioPlayer *Media; - (IBAction)Playsound { Path = [[NSBundle mainBundle] pathForResource:@"Sound" ofType:@"wav"]; Media = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:Path] error:NULL]; [Media play]; } - (void)dealloc { [Media release]; [super viewDidUnload]; } @end

    Read the article

  • Non intrusive notification without audio?

    - by acidzombie24
    i have a C# app that registers a protocol. When you click BLAH://djfhgjfdghjkd in a browser it launches my app. However you can click multiple links and each link is a note added into the app. How can i inform the user that he did fully click the link? Right now i have a console app showing up for 1sec (basically pops up and goes away as fast as possible) which felt better then a hidden console since you are unsure if it went through. The 1 second takes a lot of time when you are trying to rapidly click many notes/links and the console gets in the way. What can i do that is noticeable? I'm thinking have a box that comes up (and is semi transparent) but the click passes through it. Maybe there is a better way? Also i wouldnt know where to start with transparent windows or pass through clicks

    Read the article

  • absolute audio synchronization

    - by user1780526
    I would like to synchronize my computer with an external camcorder recording so that I can know exactly (to the millisecond) when certain recored events happen with respect to other sensors logged by the computer. One idea is to playback short sound pulses or chirps every second from the computer that get picked up by the microphone on the camcorder. But the accuracy of a simple cron job playing a sound clip is not precise enough. I was thinking of using something like gstreamer, but how does one get it to playback a clip at precisely a certain time according to the system clock?

    Read the article

  • aplay -l says no soundcards found; alsaconf says no supported cords; yet /proc/asound contains cards

    - by nimasmi
    I am trying to get HDMI output using a Gainward Nvidia 210 512 MB on Ubuntu 10.04 Lucid Lynx. I have upgraded alsa-driver, alsa-lib and alsa-utils to 1.0.24 by building from source, thanks to this blog post. Some relevant output... user@box:~$ lspci | grep Audio 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 01:09.0 Multimedia video controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder (rev 05) 01:09.2 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [MPEG Port] (rev 05) 01:09.4 Multimedia controller: Conexant Systems, Inc. CX23880/1/2/3 PCI Video and Audio Decoder [IR Port] (rev 05) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) user@box:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.24. Compiled on Sep 15 2012 for kernel 2.6.32-42-generic (SMP). user@box:~$ ls /proc/asound` card0 cards hwdep NVidia oss seq version card1 devices modules NVidia_1 pcm timers user@box:~$ aplay -l aplay: device_list:240: no soundcards found... user@box:~$ sudo /sbin/alsa-utils start * Setting up ALSA... * warning: 'alsactl restore' failed with error message 'alsactl: set_control:1403: Cannot write control '2:0:0:IEC958 Playback Default:0' : Operation not permitted'... amixer: Invalid command! ...done. Any help appreciated. PS my video card is connected only through the PCI-E slot. I assume there is no extra audio connection required.

    Read the article

< Previous Page | 77 78 79 80 81 82 83 84 85 86 87 88  | Next Page >