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  • Looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs <= $5

    - by CyanPrime
    I'm looking for a royalty free sci-fi sounding song thats 1:00+ long, and costs less then, or is $5 usd. I want to have a nice BGM for my engine demo I'm going to release for a game I'm planing on having go commercial. I don't want to spend too much money on it, so my limit is $5 usd. I want it to be at least a 1:00 in length. Where should I look? Or even better, do you have a link to a song that meets the criteria?

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  • Where to get sounds for game development for kids [closed]

    - by at.
    I'm teaching kids to program using Ruby and the gaming framework Gosu/Chingu. For the sounds for their games I've been showing them http://www.bfxr.net/. It's decent, but the samples are limited and some of them are pretty cheap (check the explosion, it's like an explosion on a commodore 64 game). Is there an easy resource kids can get the sounds they want? I'm happy to pay some kind of educational license for it.

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  • Is there a media player that works on HTTPS sites?

    - by Iain Hallam
    I'm currently using Yahoo! Media Player for a site that needs to play MP3 files that are stored on our server. In total, there's quite a bit more than the free limits at Soundcloud, but each file is only a few minutes long. YMP is pretty good, but causes security warnings on HTTPS pages, because it can only be served via HTTP. Is there an equivalent free player I can embed for the HTTPS pages? EDIT: Just to clarify, I'm initially looking for something that will scan the page and turn media links playable.

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  • Why is this beat detection code failing to register some beats properly?

    - by Quincy
    I made this SoundAnalyzer class to detect beats in songs: class SoundAnalyzer { public SoundBuffer soundData; public Sound sound; public List<double> beatMarkers = new List<double>(); public SoundAnalyzer(string path) { soundData = new SoundBuffer(path); sound = new Sound(soundData); } // C = threshold, N = size of history buffer / 1024 B = bands public void PlaceBeatMarkers(float C, int N, int B) { List<double>[] instantEnergyList = new List<double>[B]; GetEnergyList(B, ref instantEnergyList); for (int i = 0; i < B; i++) { PlaceMarkers(instantEnergyList[i], N, C); } beatMarkers.Sort(); } private short[] getRange(int begin, int end, short[] array) { short[] result = new short[end - begin]; for (int i = 0; i < end - begin; i++) { result[i] = array[begin + i]; } return result; } // get a array of with a list of energy for each band private void GetEnergyList(int B, ref List<double>[] instantEnergyList) { for (int i = 0; i < B; i++) { instantEnergyList[i] = new List<double>(); } short[] samples = soundData.Samples; float timePerSample = 1 / (float)soundData.SampleRate; int sampleIndex = 0; int nextSamples = 1024; int samplesPerBand = nextSamples / B; // for the whole song while (sampleIndex + nextSamples < samples.Length) { complex[] FFT = FastFourier.Calculate(getRange(sampleIndex, nextSamples + sampleIndex, samples)); // foreach band for (int i = 0; i < B; i++) { double energy = 0; for (int j = 0; j < samplesPerBand; j++) energy += FFT[i * samplesPerBand + j].GetMagnitude(); energy /= samplesPerBand; instantEnergyList[i].Add(energy); } if (sampleIndex + nextSamples >= samples.Length) nextSamples = samples.Length - sampleIndex - 1; sampleIndex += nextSamples; samplesPerBand = nextSamples / B; } } // place the actual markers private void PlaceMarkers(List<double> instantEnergyList, int N, float C) { double timePerSample = 1 / (double)soundData.SampleRate; int index = N; int numInBuffer = index; double historyBuffer = 0; //Fill the history buffer with n * instant energy for (int i = 0; i < index; i++) { historyBuffer += instantEnergyList[i]; } // If instantEnergy / samples in buffer < instantEnergy for the next sample then add beatmarker. while (index + 1 < instantEnergyList.Count) { if(instantEnergyList[index + 1] > (historyBuffer / numInBuffer) * C) beatMarkers.Add((index + 1) * 1024 * timePerSample); historyBuffer -= instantEnergyList[index - numInBuffer]; historyBuffer += instantEnergyList[index + 1]; index++; } } } For some reason it's only detecting beats from 637 sec to around 641 sec, and I have no idea why. I know the beats are being inserted from multiple bands since I am finding duplicates, and it seems that it's assigning a beat to each instant energy value in between those values. It's modeled after this: http://www.flipcode.com/misc/BeatDetectionAlgorithms.pdf So why won't the beats register properly?

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  • How do I get a Line6 UX1 soundcard to work?

    - by the_drow
    I own a Line6 UX1 soundcard and I would like to make it work for Ubuntu. I have followed the instructions here and it worked. But at some point I upgraded my kernel version (not sure what uname -a prints but it's related) and it stopped working. Here's what uname -a prints: Linux ubuntu 2.6.32-29-generic #58-Ubuntu SMP Fri Feb 11 20:52:10 UTC 2011 x86_64 GNU/Linux I figured out that maybe it's installed per version so I used svn update and hit make again. My guess was right as it copied the relevant files to the new version's folder. I restarted and still nothing. Should I revert to an older version? Or is there a solution here?

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  • How to trace a function array argument in DTrace

    - by uejio
    I still use dtrace just about every day in my job and found that I had to print an argument to a function which was an array of strings.  The array was variable length up to about 10 items.  I'm not sure if the is the right way to do it, but it seems to work and is not too painful if the array size is small.Here's an example.  Suppose in your application, you have the following function, where n is number of item in the array s.void arraytest(int n, char **s){    /* Loop thru s[0] to s[n-1] */}How do you use DTrace to print out the values of s[i] or of s[0] to s[n-1]?  DTrace does not have if-then blocks or for loops, so you can't do something like:    for i=0; i<arg0; i++        trace arg1[i]; It turns out that you can use probe ordering as a kind of iterator. Probes with the same name will fire in the order that they appear in the script, so I can save the value of "n" in the first probe and then use it as part of the predicate of the next probe to determine if the other probe should fire or not.  So the first probe for tracing the arraytest function is:pid$target::arraytest:entry{    self->n = arg0;}Then, if I want to print out the first few items of the array, I first check the value of n.  If it's greater than the index that I want to print out, then I can print that index.  For example, if I want to print out the 3rd element of the array, I would do something like:pid$target::arraytest:entry/self->n > 2/{    printf("%s",stringof(arg1 + 2 * sizeof(pointer)));}Actually, that doesn't quite work because arg1 is a pointer to an array of pointers and needs to be copied twice from the user process space to the kernel space (which is where dtrace is). Also, the sizeof(char *) is 8, but for some reason, I have to use 4 which is the sizeof(uint32_t). (I still don't know how that works.)  So, the script that prints the 3rd element of the array should look like:pid$target::arraytest:entry{    /* first, save the size of the array so that we don't get            invalid address errors when indexing arg1+n. */    self->n = arg0;}pid$target::arraytest:entry/self->n > 2/{    /* print the 3rd element (index = 2) of the second arg. */    i = 2;    size = 4;    self->a_t = copyin(arg1+size*i,size);    printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}If your array is large, then it's quite painful since you have to write one probe for every array index.  For example, here's the full script for printing the first 5 elements of the array:#!/usr/sbin/dtrace -spid$target::arraytest:entry{        /* first, save the size of the array so that we don't get           invalid address errors when indexing arg1+n. */        self->n = arg0;}pid$target::arraytest:entry/self->n > 0/{        i = 0;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 1/{        i = 1;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 2/{        i = 2;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 3/{        i = 3;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));}pid$target::arraytest:entry/self->n > 4/{        i = 4;        size = sizeof(uint32_t);        self->a_t = copyin(arg1+size*i,size);        printf("%s: a[%d]=%s",probefunc,i,copyinstr(*(uint32_t *)self->a_t));} If the array is large, then your script will also have to be very long to print out all values of the array.

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  • Routing audio from GSM module to a Bluetooth HandsFree device

    - by Shaihi
    I have a system with the following setup: I use: Windows CE 6 R3 Microsoft's Bluetooth stack including all profiles Motorola H500 The Audio Gateway service is up and running (checked through services list in cmd) GSM Module is functional - I am able to set outgoing calls and to answer calls. Bluetooth is functional - the A2DP profile plays music to Motorola headphones (can't remember the model right now) I want to hold a conversation using a headset device. I have included all Bluetooth components in the catalog. I pair the device using the Control Panel applet. When I press the button on the Motorla device to answer a call I get a print by the Audio Gateway: BTAGSVC: ConnectionEvent. BTAGSVC: SCOListenThread_Int - Connection Event. BTAGSVC: ConnectionEvent. BTAGSVC: SCOListenThread_Int - Connection Event. BTAGSVC: ConnectionEvent. BTAGSVC: A Bluetooth peer device has connected to the Audio Gateway. BTAGSVC: Could not open registry key for BT Addr: 2. BTAGSVC: The peer device was not accepted since the user has never confirmed it as a device to be used. So my questions are as follows: What do I need to do to pair the device with the Audio Gateway? Once my device is paired, do I need to set anything else up? (except for the GSM module of course)

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  • Method for launching audio player on Android from web page for streaming media

    - by Brad
    To link to SHOUTcast/HTTP internet radio streams, traditionally you would link to a playlist file, such as an M3U or PLS. From there, the browser would launch the audio player registered to handle the playlist. This works great on any PC, Palm, Blackberry, and iPhone. This method does not work in Android without installing extra software. Sure, Just Playlists or StreamFurious can handle it just fine, but I am assuming there has to be a way to invoke the audio or video player commonly installed by default on Android installations. By default, no audio player is capable of handling M3U or PLS. The player seems to open it, but says "Unsupported Media Type". To make this more annoying, the browser is capable of streaming MP3 audio over HTTP, simply by opening a link to an MP3 file. I have tried simply linking directly to the MP3 stream hosted by SHOUTcast, which should end up in the same result, but SHOUTcast detects "Mozilla" in the user-agent string, and instead of sending the stream, it sends the information page for the station. How should I link to a SHOUTcast stream on Android, from a normal mobile site, without using extra applications?

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  • Visualizing volume of PCM samples

    - by genevincent
    I have several chunks of PCM audio (G.711) in my C++ application. I would like to visualize the different audio volume in each of these chunks. My first attempt was to calculate the average of the sample values for each chunk and use that as an a volume indicator, but this doesn't work well. I do get 0 for chunks with silence and differing values for chunks with audio, but the values only differ slighly and don't seem to resemble the actual volume. What would be a better algorithem calculate the volume ? I hear G.711 audio is logarithmic PCM. How should I take that into account ?

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  • General question about DirectShow.NET, DirectShow and Windows Media Format

    - by Paul Andrews
    I searched and googled for an answer but couldn't find one. Basically I'm developing a webcam/audio streaming application which should capture audio and video from a pc (usb webcam/microphone) and send them to a receiving server. What the server will do with that it's another story and phase two (which I'm skipping for now) I wrote some code using DirectShow and Windows Media Format and it worked great for capture audio/video and sending them to another client, but there's a major problem: latency. Everywhere in the internet everyone gave me the same answer: "sorry dude but media format isn't for video conferencing, their codecs have too high latency". I thought I could skip the .wmv problems but seems like it's not possible to do... this road ends here then. So I saw a few examples with DirectShow.NET which were faster for both audio and video.. my question is: how come that DirectShow.NET is faster and better for video/audio conferencing? Shouldn't it be just a .NET porting of C++'s DirectShow? Am I missing something? I'm a bit confused at this point

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  • AudioFileWriteBytes fails with error code -40

    - by alexbw
    I'm trying to write raw audio bytes to a file using AudioFileWriteBytes(). Here's what I'm doing: void writeSingleChannelRingBufferDataToFileAsSInt16(AudioFileID audioFileID, AudioConverterRef audioConverter, ringBuffer *rb, SInt16 *holdingBuffer) { // First, figure out which bits of audio we'll be // writing to file from the ring buffer UInt32 lastFreshSample = rb->lastWrittenIndex; OSStatus status; int numSamplesToWrite; UInt32 numBytesToWrite; if (lastFreshSample < rb->lastReadIndex) { numSamplesToWrite = kNumPointsInWave + lastFreshSample - rb->lastReadIndex - 1; } else { numSamplesToWrite = lastFreshSample - rb->lastReadIndex; } numBytesToWrite = numSamplesToWrite*sizeof(SInt16); Then we copy the audio data (stored as floats) to a holding buffer (SInt16) that will be written directly to the file. The copying looks funky because it's from a ring buffer. UInt32 buffLen = rb->sizeOfBuffer - 1; for (int i=0; i < numSamplesToWrite; ++i) { holdingBuffer[i] = rb->data[(i + rb->lastReadIndex) & buffLen]; } Okay, now we actually try to write the audio from the SInt16 buffer "holdingBuffer" to the audio file. The NSLog will spit out an error -40, but also claims that it's writing bytes. No data is written to file. status = AudioFileWriteBytes(audioFileID, NO, 0, &numBytesToWrite, &holdingBuffer); rb->lastReadIndex = lastFreshSample; NSLog(@"Error = %d, wrote %d bytes", status, numBytesToWrite); return;

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  • Extract wav file from video file

    - by Nikos Steiakakis
    I am developing an application in which I need to extract the audio from a video. The audio needs to be extracted in .wav format but I do not have a problem with the video format. Any format will do, as long as I can extract the audio in a wav file. Currently I am using Windows Media Player COM control in a windows form to play the videos, but any other embedded player will do as well. Any suggestions on how to do this? Thanks

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  • 2 AudioQueue questions

    - by iter
    I am learning to use AudioQueue. I wish to generate an audio stream programmatically. I have 2 issues that I cannot account for. I am getting audio when I run in the simulator, but not on an iPhone. (Other apps do produce sound on the phone). I get about 20ms-long gaps of silence between buffers. In my testing, I generate an audio buffer on startup and repeatedly enqueue it without modification. I don't spend any processing on filling audio buffers at runtime, not even copying them. Ari.

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  • Dynamically calculate frequency value.

    - by MS Nathan
    Hi, In my app, I want to find/calculate the audio frequency as dynamically when i am recording an audio and no need to save, play and all. Now i am trying to do that with help of an aurioToch sample code. In that sample, inside FFTBufferManager class methods such as GrabAudioData and ComputeFFT,Here I am not able to find where they are calculating frequency value as dynamically depends on the audio sound and I spent more than 5 days.please help me.

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  • How do you control the playback levels (decibles?) using the iPhone AVAudioPlayer? Or do I need to u

    - by Joshua
    My audio clips sound perfect when I upload them to the iPhone via iTunes. And I am pretty sure it is because the iPod has a maximum playback level, so the audio doesn't sound overdriven. In my app, I include the same audio files, and when I play them [myAudio play]; the levels are so high that the audio becomes indiscernible. I found in the library http://developer.apple.com/iphone/library/documentation/AVFoundation/Reference/AVAudioPlayerClassReference/Reference/Reference.html#//apple_ref/doc/uid/TP40008067-CH1-SW2 that it says that you can "Control relative playback level for each sound you are playing" but I've been searching this issue out for hours and I haven't gotten anywhere. Any help would be wonderful!

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  • Sound sample recognition library/code

    - by Daniel Mošmondor
    I don't want sound-to-text software. What I need is the following: I'll record multiple (say 50+) audio streams (recordings of radio stations) from that recordings, I'll mark interesting audio clips - their length ranges from 2 to 60 seconds - there will be few thousands of such audio clips library should be able to find other instances of same audio clips from recorded sound streams confidence factor should be reported to used and additional input provided so the recognition could perform better next time Do you know of such software library? LGPL would be most valuable to me, but I can go for commercial license as well.

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  • iPhone - openAL stops playing if I record with AVAudioRecorder

    - by Oscar Peli
    Hi there, this is an iPhone-related question: I use openAL to play some sound (I have to manage gain, pitch, etc.). I want to record what I'm playing and I use AVAudioRecorder but when I "prepareToRecord" openAL stops to play audio. What's the problem? Here is the record IBAction I use: - (IBAction) record: (id) sender { NSError *error; NSMutableDictionary *settings = [NSMutableDictionary dictionary]; [settings setValue: [NSNumber numberWithInt:kAudioFormatLinearPCM] forKey:AVFormatIDKey]; [settings setValue: [NSNumber numberWithFloat:8000.0] forKey:AVSampleRateKey]; [settings setValue: [NSNumber numberWithInt: 1] forKey:AVNumberOfChannelsKey]; [settings setValue: [NSNumber numberWithInt:16] forKey:AVLinearPCMBitDepthKey]; [settings setValue: [NSNumber numberWithBool:NO] forKey:AVLinearPCMIsBigEndianKey]; [settings setValue: [NSNumber numberWithBool:NO] forKey:AVLinearPCMIsFloatKey]; NSURL *url = [NSURL fileURLWithPath:FILEPATH]; self.recorder = [[AVAudioRecorder alloc] initWithURL:url settings:settings error:&error]; self.recorder.delegate = self; self.recorder.meteringEnabled = YES; [self.recorder prepareToRecord]; [self.recorder record]; } Thanks

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  • Playing a .TS file on iOS

    - by Jonathan Grynspan
    We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

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  • Design considerations for temporarily transforming a player into an animal in a role playing game

    - by mikedev
    I am working on a role playing game for fun and to practice design patterns. I would like players to be able to transform themselves into different animals. For example, a Druid might be able to shape shift into a cheetah. Right now I'm planning on using the decorator pattern to do this but my question is - how do I make it so that when a druid is in the cheetah form, they can only access skills for the cheetah? In other words, they should not be able to access their normal Druid skills. Using the decorator pattern it appears that even in the cheetah form my druid will be able to access their normal druid skills. class Druid : Character { // many cool druid skills and spells void LightHeal(Character target) { } } abstract class CharacterDecorator : Character { Character DecoratedCharacter; } class CheetahForm : CharacterDecorator { Character DecoratedCharacter; public CheetahForm(Character decoratedCharacter) { DecoratedCharacter= decoratedCharacter; } // many cool cheetah related skills void CheetahRun() { // let player move very fast } } now using the classes Druid myDruid = new Druid(); myDruid.LightHeal(myDruid); // casting light heal here is fine myDruid = new CheetahForm(myDruid); myDruid.LightHeal(myDruid); // casting here should not be allowed Hmmmm...now that I think about it, will myDruid be unable to us the Druid class spells/skills unless the class is down-casted? But even if that's the case, is there a better way to ensure that myDruid at this point is locked out from all Druid related spells/skills until it is cast back to a Druid (since currently it's in CheetahForm)

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  • Playing wave file ends immediatly (C++, Windows)

    - by TyBoer
    I've got a following situation. On a machine there is a Fritz ISDN card. There is a process that is responsible for playing a certain wave file on this device's wave out (ISDN connection is made at startup and made persistent). The scenario is easy, whenever needed the process calls waveOutWrite() on the previously opened wave device (everything initialized without any problems of course) and a callback function waits for MMWOMDONE msg to know that the playback has been finished. Since a few days however (nothing changed neither in the process nor the machine) the MMWOMDONE message comes immediately after calling waveOutWrite() even though the wave lasts a couple of seconds. Again no error is reported, it looks like the file was played but had zero length (which is not the case). I am also sure that waveOutReset() was not called by my process (it would also trigger sending the mentioned message). I already used to have some strange problems in the past that where solved simply by reinstalling TAPI drivers. This time for some reason it is problematic form me to perform that once again and am trying more analytical approach :). Any suggestions what might cause such a behavior? Maybe sth on the other end of the ISDN line?

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  • Flash AS3: Loading bar not finished when movie starts playing

    - by flashey
    Hi, I'm using the same code I always use for preloading another swf but it's not working this time. The problem this time is that when the loading bar gets to 16% every time you can hear the movie I'm loading playing in the background. I can just add a stop to the first frame of the movie I'm loading ("trial_1.swf") but how do I tell it to go to the second frame once it has loaded? Any help is much appreciated! Here's my code: var myrequest:URLRequest=new URLRequest ("trial_1.swf"); var myloader:Loader = new Loader(); myloader.load(myrequest); myloader.contentLoaderInfo.addEventListener(ProgressEvent.PROGRESS, progresshandler); function progresshandler(myevent:ProgressEvent):void { var myprogress:Number=myevent.target.bytesLoaded/myevent.target.bytesTotal; bar_mc.scaleX=myprogress*1.5; myTextField_txt.text = Math.round(myprogress*100)+"%" } myloader.contentLoaderInfo.addEventListener(Event.COMPLETE, finished); function finished(myevent:Event):void { addChild(myloader); removeChild(myTextField_txt) removeChild(bar_mc); removeChild(logo_mc); }

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  • Detect TCP connection close when playing Flash video

    - by JoJo
    On the Flash client side, how do I detect when the server purposely closes the TCP connection to its video stream? I'll need to take action when this occurs - maybe attempt to restart the video or display an error message. Currently, the connection closing and the connection being slow look the same to me. The NetStream object ushers a NetStream.Play.Stop event in both cases. When the connection is slow, it usually recovers by itself within seconds. I wish to only take action when the connection is closed, not when it is slow. Here's how my general setup looks like. It's the basic NetConnection-NetStream-Video setup. this.vidConnection = new NetConnection(); this.vidConnection.addEventListener(AsyncErrorEvent.ASYNC_ERROR, this.connectionAsyncError); this.vidConnection.addEventListener(IOErrorEvent.IO_ERROR, this.connectionIoError); this.vidConnection.addEventListener(NetStatusEvent.NET_STATUS, this.connectionNetStatus); this.vidConnection.connect(null); this.vidStream = new NetStream(this.vidConnection); this.vidStream.addEventListener(AsyncErrorEvent.ASYNC_ERROR, this.streamAsyncError); this.vidStream.addEventListener(IOErrorEvent.IO_ERROR, this.streamIoError); this.vidStream.addEventListener(NetStatusEvent.NET_STATUS, this.streamNetStatus); this.vid.attachNetStream(this.vidStream); None of the error events fire when the server closes the TCP or when the connection freezes up. Only the NetStream.Play.Stop event fires. Here's a trace of what happens from initially playing the video to the TCP connection closing. connection net status = NetConnection.Connect.Success playStream(http://192.168.0.44/flv/4d29104a9aefa) NetStream.Play.Start NetStream.Buffer.Flush NetStream.Buffer.Full NetStream.Buffer.Empty checkDimensions 0 0 onMetaData NetStream.Buffer.Full NetStream.Buffer.Flush checkDimensions 960 544 NetStream.Buffer.Empty NetStream.Buffer.Flush NetStream.Play.Stop

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  • Toggling between instances of NiftyPlayer on a page - won't stop playing when hidden on IE

    - by Ashley
    Hi, i've got a page with links to MP3s, when the link is clicked I use javascript to show a small Flash player (NiftyPlayer) under the link. When a different link is clicked, the old player is hidden and the new player is revealed. The player auto-starts when the element is shown, and auto-stops when hidden - in Firefox. In IE it will only auto-start and NOT auto-stop. This is what I would like to solve. This is an example HTML with link and player <a href="Beat The Radar - Misunderstood What You Said.mp3" onclick="toggle_visibility('player662431');return false;" class="mp3caption">Misunderstood What You Said</a> <div id="player662431" class="playerhide"><embed src="http://www.xxx.com/shop/flash/player.swf?file=/mp3/Beat The Radar - Misunderstood What You Said.mp3&as=1" quality="high" bgcolor="#000000" width="161" height="13" name="niftyPlayer662431" align="" type="application/x-shockwave-flash" swLiveConnect="true" pluginspage="http://www.macromedia.com/go/getflashplayer"></embed> Here is the javascript (i've got jquery installed to let me hide all the open players on this page apart from the new one) function toggle_visibility(id) { $('.playerhide').hide(); var e = document.getElementById(id); e.style.display = 'block'; } I think what I need to do is start the player manually with javascript (rather than using the autostart as=1 function in the URL string) There is some javascript that comes with NiftyPlayer to allow this EG niftyplayer('niftyPlayer1').play() there is also a stop method. I need some help with javascript - how do I add this call to play into my toggle_visibility function (it has the same unique ID number added to the name of the player as the ID of the div that's being shown, but I don't know how to pull this ID number out of one thing and put it in another) I also would like to be able to do niftyplayer('niftyPlayer1').stop() to stop the audio of the previously running player. Is it possible to store the current ID number somewhere and call it back when needed? Thanks for the help, i'm a PHP programmer who needs some support with Javascript - I know what I want to achieve, just don't know the commands to do it! Thanks

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  • Flash CS4 [AS3]: Playing Card Deck Array

    - by Ben
    I am looking to make a card game in Flash CS4 using AS3 and I am stuck on the very first step. I have created graphics for a standard 52 card deck of playing cards and imported them into the library in Flash and then proceeded to convert them all to Movie Clips. I have also used the linkage to make them available in the code. The movie clips and the linkage are named in sequence, as in the Ace of Clubs would be C1, two of Diamonds is called D2, Jack of Spades is S11. (C = Clubs, D = Diamonds, S = Spades, H = Hearts and numbers 1 through 13 are the card values. 1 being Ace, 11 being Jack, 12 being Queen, 13 being King). As far as I know my next step would be to arrange the cards into an array. This is the part that I am having problems with. Can someone please point me in the right direction, what would be the best way to do this. Could you provide me with a bit of sample code as well? I have had a look at few tutorials online but they are all telling me different things, some are incomplete and the rest...well...they're just cr*ppy. Thanks in advance! Ben

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  • Video not playing on android webview

    - by rand
    I am working with an Android and PhoneGap application and am using the HTML5 video tag to play videos on my web page. When I play the video is not visible and video is not playing itself. How can I play a HTML5 video on Android? Code for the same given below <!DOCTYPE HTML> <html> <head> <script type="text/javascript" charset="utf-8" src="cordova-1.8.1.js"></script> <meta http-equiv="content-type" content="text/html; charset="> <title></title> </head> <body > <video id="video" autobuffer height="240" width="360" onclick="this.play();> <source src="test.mp4"> <source src="test.mp4" type="video/webm"> <source src="test.mp4" type="video/ogg"> </video> <div id="msg"></div> <script type="text/javascript"> </script> </body> </html> and the activity class onCreate method-- public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.activity_main); final WebView webView = (WebView) findViewById(R.id.webview); WebSettings webSettings = webView.getSettings(); webSettings.setLayoutAlgorithm(LayoutAlgorithm.NARROW_COLUMNS); webView.getSettings().setJavaScriptEnabled(true); webSettings.setBuiltInZoomControls(true); webSettings.setPluginState(PluginState.ON); webView.getSettings().setPluginsEnabled(true); webSettings.setAllowFileAccess(true); webView.loadUrl("file:///android_asset/www/html5videoEvents.html"); }

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