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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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  • Music player with a few specific requirements

    - by Jordan Uggla
    I am looking for a music player with a few specific requirements: Must have a search function that whittles down results as you type, searching the entire library. Must start playing a song when double clicked, and not continue to another song when that song finishes. Must be approachable and immediately usable by people completely unfamiliar with the program. I think this is mostly covered by the first two requirements being met. I've tried many players but unfortunately every one has failed to meet at least one of the requirements. Rhythmbox meets 1 and 3, but continues to the next search result after the song which was double clicked ends. Banshee is basically the same as Rhythmbox. While it has an option to "Stop when finished" this cannot (as far as I can tell) be made the default when double clicking a song. Audacious (as far as I can tell) fails at 1. Muine meets requirements 1 and 2, but unfortunately I couldn't make the search dialog always shown like it is with Rhythmbox / Banshee which, despite its very simple interface, made Muine incomprehensible to people trying to use it for the first time. Amarok I could not configure to meet requirement 1, but I think it's likely I was just missing something, and with its configurability I'm confident that I can set it up to meet requirements 2 and 3.

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  • Difference between Sound and Music

    - by Southpaw Hare
    What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

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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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  • What's the difference between Pygame's Sound and Music classes?

    - by Southpaw Hare
    What are the key differences between the Sound and Music classes in Pygame? What are the limitations of each? In what situation would one use one or the other? Is there a benefit to using them in an unintuitive way such as using Sound objects to play music files or visa-versa? Are there specifically issues with channel limitations, and do one or both have the potential to be dropped from their channel unreliably? What are the risks of playing music as a Sound?

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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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  • Playing video and audio in iPhone not working...

    - by Scott
    So we have buttons linked up to display images/videos/audio on click depending on a check we do earlier. That part works fine. It knows which one to play, however, when we click the buttons for video and audio, nothing happens. The image one works fine. The video and audio are being taken for a URL online, they are not local, but everywhere said this was still possible. Here is a little snippet of the code where we play the two files: if ( [fName hasSuffix:@".png"]) { NSLog(@"PICTURE"); NSURL *url = [NSURL URLWithString: fName]; UIImage *image = [UIImage imageWithData: [NSData dataWithContentsOfURL:url]]; self.view = [[UIView alloc] initWithFrame:[[UIScreen mainScreen] applicationFrame]]; // self.view.backgroundColor = [[UIColor alloc] initWithPatternImage:[UIImage imageNamed:@"MainBG.jpg"]]; [self.view addSubview:[[UIImageView alloc] initWithImage:image]]; } if ( [fName hasSuffix:@".mp4"]) { NSLog(@"VIDEO"); //NSString *path = [[NSBundle mainBundle] pathForResource:fName ofType:@"mp4"]; //NSLog(path); NSURL *url = [NSURL fileURLWithPath:fName]; MPMoviePlayerController *player = [[MPMoviePlayerController alloc] initWithContentURL:url]; [player play]; } if ( [fName hasSuffix:@".mp3"]) { NSLog(@"AUDIO"); NSURL *url = [NSURL fileURLWithPath:fName]; NSData *soundData = [NSData dataWithContentsOfURL:url]; AVAudioPlayer *avPlayer = [[AVAudioPlayer alloc] initWithData:soundData error: nil]; [avPlayer play]; } See anything wrong? By the way it compiles and runs, but nothing happens when we hit the button that executes that code.

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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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  • 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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  • 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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  • 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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  • Why does my Intel HDA onboard sound card not have a "Mix" device / channel?

    - by Hanno Fietz
    I want to be able to record what my sound card outputs on the speakers / headphones. This question is all over the interwebs again and again, and there seem to be two outcomes: in your selection of audio input devices, there's a device called "Stereo Mix", or similar, which is the "loopback" device for audio. Choose that in your recording tool and you're done. there's no such device and only speculative posts about why that may be. Now, I'm using ALSA and an Intel HDA chipset on my mainboard under Kubuntu Karmic. I have some 5-10 output channels and "Mic", "Front Mic" and "Line" for input. All of those are available in KMix, Audacity and other software. No "loopback" / "Mix" / whatever. Do I have to get some driver / kernel module set up ALSA in some way set up my system configuration in some way use a software solution (such as JACK) I had a look at JACK, and found it rather hard to understand, it's either an expert tool or just clumsy, I couldn't say. At least, I wasn't able to figure out how to achieve what I wanted. One of my problems seems to be that I don't understand where and how the mixing happens. Are there sound cards which just aren't able to do it? Why does the sound card matter at all, since I could in theory grab the data stream at some point before it goes to the hardware, right?

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  • How can I control which sound card Ubuntu uses for playback?

    - by GorillaSandwich
    I am dual-booting Ubuntu 9.04 and Windows XP but am new to Ubuntu. In Windows, I use an M-Audio Audiophile 2496 sound card for recording (because it has RCA input jacks for my mixer), but I don't use it for playback (because my speakers use a 1/8 inch jack); instead, I use the motherboard's built-in sound card. I tried to recreate this arrangement in Ubuntu, but despite selecting the built-in card for all playback under System > Preferences > Sound, I still have inconsistent results. Rhythmbox plays back through the integrated card, but Flash content in the browser and games in the OS send their audio to the Audiophile card. I have seen recommendations to use a program called "Jack" to control this, but I installed it and found it baffling. How can I control which card is used for playback, other than disabling one card (as I discovered how to do and explain below)? Also, is there a GUI for disabling hardware, or is it necessary to edit a configuration file?

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  • Video/audio streaming does not stop even if UIWebView is closed - iPad

    - by lostInTransit
    Hi I see this issue only on the iPad. The same things works as expected on the iPhone. I am opening the URL from my application in a UIWebView. If the URL is a normal web page, it works fine as expected. But if the URL is that of a remote video/audio file, the UIWebView opens the default player which is again good. Now when I dismiss the UIWebView (by clicking on the Done button on the player), the streaming doesn't stop and the audio/video keeps playing in the background (I cannot see it but it does keep playing in the background, can hear it). The UIViewController in which the webview was created is also dealloced (I put in a log statement in the dealloc method) but the streaming doesn't stop. Can someone please help me out on why this could be happening? And how can I stop the audio/video streaming when the UIWebView is closed? Thanks.

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  • How to create an Audio CD using C# or Java

    - by Elie
    I'm looking for an API that would allow me to create an audio CD from within a C# application. The CDs are to be created and closed in the same session (no rewrite required). Basically, my application locates files on behalf of a user, and, if a blank CD is present in the drive, creates an audio CD for the user. If no CD is present, it checks to see if there's a USB drive attached and copies the files there (this part I already know how to do). I would prefer to write this application in either C# or Java, as I'm most comfortable with those, but I don't know how hard it would be to create CDs using either language. There are several other questions here that deal with regular CDs, but I didn't see any discussing audio CDs.

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  • Recording Audio through RTMP/Rails

    - by Lowgain
    I am in the process of building a rails/flex application which requires audio to be recorded and then stored in our amazon s3 account. I have found no alternative to using some form of RTMP server for recording audio through flash, but our hosting environment will not allow us to install anything like FMS, Red5, etc. Is there any existing Ruby/Rails RTMP solution that will allow audio recording? If not, is it possible for Rails to at least intercept the RTMP stream and then I can hope to reference red5 or something for parsing the data (long shot, I know)? The other alternative I can think of is hosting a red5 server on another host and communicating with our rails app once the saving/uploading is done, which is not preferred. Am I going to have any luck here?

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