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  • How do I convert a System::IO::Stream^ to an LPCSTR for PlaySound?

    - by Jon Cage
    I'm trying to embed and then play back a .wav file in a C++/CLI app but all the examples I've seen which use PlaySound are in VB. I can't see how to get froma Stream^ to the LPCSTR which PlaySound requires: System::IO::Stream^ s = Assembly::GetExecutingAssembly()->GetManifestResourceStream ("Ping.wav"); LPCSTR buf = s->????; PlaySound(buf, NULL, SND_ASYNC|SND_MEMORY|SND_NOWAIT); I guess I need some sort of horrible .net memory conversion magic.

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  • Type Casting variables in PHP: Is there a practical example?

    - by Stephen
    PHP, as most of us know, has weak typing. For those who don't, PHP.net says: PHP does not require (or support) explicit type definition in variable declaration; a variable's type is determined by the context in which the variable is used. Love it or hate it, PHP re-casts variables on-the-fly. So, the following code is valid: $var = "10"; $value = 10 + $var; var_dump($value); // int(20) PHP also alows you to explicitly cast a variable, like so: $var = "10"; $value = 10 + $var; $value = (string)$value; var_dump($value); // string(2) "20" That's all cool... but, for the life of me, I cannot conceive of a practical reason for doing this. I don't have a problem with strong typing in languages that support it, like Java. That's fine, and I completely understand it. Also, I'm aware of—and fully understand the usefulness of—type hinting in function parameters. The problem I have with type casting is explained by the above quote. If PHP can swap types at-will, it can do so even after you force cast a type; and it can do so on-the-fly when you need a certain type in an operation. That makes the following valid: $var = "10"; $value = (int)$var; $value = $value . ' TaDa!'; var_dump($value); // string(8) "10 TaDa!" So what's the point? Can anyone show me a practical application or example of type casting—one that would fail if type casting were not involved? I ask this here instead of SO because I figure practicality is too subjective. Edit in response to Chris' comment Take this theoretical example of a world where user-defined type casting makes sense in PHP: You force cast variable $foo as int -- (int)$foo. You attempt to store a string value in the variable $foo. PHP throws an exception!! <--- That would make sense. Suddenly the reason for user defined type casting exists! The fact that PHP will switch things around as needed makes the point of user defined type casting vague. For example, the following two code samples are equivalent: // example 1 $foo = 0; $foo = (string)$foo; $foo = '# of Reasons for the programmer to type cast $foo as a string: ' . $foo; // example 2 $foo = 0; $foo = (int)$foo; $foo = '# of Reasons for the programmer to type cast $foo as a string: ' . $foo; UPDATE Guess who found himself using typecasting in a practical environment? Yours Truly. The requirement was to display money values on a website for a restaurant menu. The design of the site required that trailing zeros be trimmed, so that the display looked something like the following: Menu Item 1 .............. $ 4 Menu Item 2 .............. $ 7.5 Menu Item 3 .............. $ 3 The best way I found to do that wast to cast the variable as a float: $price = '7.50'; // a string from the database layer. echo 'Menu Item 2 .............. $ ' . (float)$price; PHP trims the float's trailing zeros, and then recasts the float as a string for concatenation.

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  • Decoding ima4 audio format

    - by MrDatabase
    To reduce the download size of an iPhone application I'm compressing some audio files. Specifically I'm using afconvert on the command line to change .wav format to .caf format w/ ima4 compression. I've read this (wooji-juice.com) awesome post about this exact topic. I'm having trouble w/ the "decoding ima4 packets" step. I've looked at their sample code and I'm stuck. Please help w/ some pseudo code or sample code that can guide me in the right direction. Thanks! Additional info: Here is what I've completed and where I'm having trouble... I can play .wav files in both the simulator and on the phone. I can compress .wav files to .caf w/ ima4 compression using afconvert on the command line. I'm using the SoundEngine that came w/ CrashLanding (I fixed one memory leak). I modified the SoundEngine code to look for the mFormatID 'ima4'. I don't understand the blog post linked above starting w/ "Calculating the size of the unpacked data". Why do I need to do this? Also, what does the term "packet" refer to? I'm very new to any sort of audio programming.

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  • Relative Uri works for BitmapImage, but not for MediaPlayer?

    - by Thomas Stock
    This will be simple for you guys: var uri = new Uri("pack://application:,,,/LiftExperiment;component/pics/outside/elevator.jpg"); imageBitmap = new BitmapImage(); imageBitmap.BeginInit(); imageBitmap.UriSource = uri; imageBitmap.EndInit(); image.Source = imageBitmap; = Works perfectly on a .jpg with Build Action: Content Copy to Output Directory: Copy always MediaPlayer mp = new MediaPlayer(); var uri = new Uri("pack://application:,,,/LiftExperiment;component/sounds/DialingTone.wav"); mp.Open(uri); mp.Play(); = Does not work on a .wav with the same build action and copy to output. I see the file in my /debug/ folder.. MediaPlayer mp = new MediaPlayer(); var uri = new Uri(@"E:\projects\LiftExp\_solution\LiftExperiment\bin\Debug\sounds\DialingTone.wav"); mp.Open(uri); mp.Play(); = Works perfectly.. So, how do I get the sound to work with a relative path? Why is it not working this way? Let me know if you want more code or screenshots. Thanks.

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  • PlaySound linker error in C++

    - by logic-unit
    Hello, I'm getting this error: [Linker error] undefined reference to 'PlaySoundA@12' Id returned 1 exit status From this code: // c++ program to generate a random sequence of numbers then play corresponding audio files #include <windows.h> #include <mmsystem.h> #include <iostream> #pragma comment(lib, "winmm.lib") using namespace std; int main() { int i; i = 0; // set the value of i while (i <= 11) // set the loop to run 11 times { int number; number = rand() % 10 + 1; // generate a random number sequence // cycling through the numbers to find the right wav and play it if (number == 0) { PlaySound("0.wav", NULL, SND_FILENAME); // play the random number } else if (number == 1) { PlaySound("1.wav", NULL, SND_FILENAME); // play the random number } //else ifs repeat to 11... i++; // increment i } return 0; } I've tried absolute and relative paths for the wavs, the file size of them is under 1Mb each too. I've read another thread here on the subject: http://stackoverflow.com/questions/1565439/how-to-playsound-in-c As you may well have guessed this is my first C++ program, so my knowledge is limited with where to go next. I've tried pretty much every page Google has on the subject including MSDN usage page. Any ideas? Thanks in advance...

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  • iPhone AVAudioPlayer failed to find codec

    - by Anthony
    Hello, I am writing an app that downloads a wav file from a server and needs to play that file. The files use the mulaw codec with 2:1 compression. These wav files are dynamically created by a seperate process so there is no way for me to preconvert the files to a different format or codec, I need to be able to play them as is. I am using an AVAudioPlayer instance initialized as follows: NSURL *audioURL = [[NSURL alloc] initWithString:@"http://xxx.../file.wav"]; NSData *audioData = [[NSData alloc] initWithContentsOfURL:audioURL]; AVAudioPlayer *audio = [[AVAudioPlayer alloc] initWithData:audioData error:nil]; [audio play]; However, when the play method executes, I get the following Console Output when executing on the Simulator: AudioQueue codec policy 1: failed to find a codec of the requested type I also tried saving the downloaded data to a local file and using a file URL, however that yeilds the same results. The downloaded file does play fine on both Mac and Windows based desktop media players. The SDK docs state that the mulaw codec is supported on the iPhone, so I am unsure why it is failing to find it. Any assistance would be greatly appreciated. Thanks.

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  • Swt file dialog too much files selected?

    - by InsertNickHere
    Hi there, the swt file dialog will give me an empty result array if I select too much files (approx. 2500files). The listing shows you how I use this dialog. If i select too many sound files, the syso will show 0. Debugging tells me, that the files array is empty in this case. Is there any way to get this work? FileDialog fileDialog = new FileDialog(mainView.getShell(), SWT.MULTI); fileDialog.setText("Choose sound files"); fileDialog.setFilterExtensions(new String[] { new String("*.wav") }); Vector<String> result = new Vector<String>(); fileDialog.open(); String[] files = fileDialog.getFileNames(); for (int i = 0, n = files.length; i < n; i++) { if( !files[i].contains(".wav")) { System.out.println(files[i]); } StringBuffer stringBuffer = new StringBuffer(); stringBuffer.append(fileDialog.getFilterPath()); if (stringBuffer.charAt(stringBuffer.length() - 1) != File.separatorChar) { stringBuffer.append(File.separatorChar); } stringBuffer.append(files[i]); stringBuffer.append(""); String finalName = stringBuffer.toString(); if( !finalName.contains(".wav")) { System.out.println(finalName); } result.add(finalName); } System.out.println(result.size()) ;

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  • waveInProc / Windows audio question...

    - by BTR
    I'm using the Windows API to get audio input. I've followed all the steps on MSDN and managed to record audio to a WAV file. No problem. I'm using multiple buffers and all that. I'd like to do more with the buffers than simply write to a file, so now I've got a callback set up. It works great and I'm getting the data, but I'm not sure what to do with it once I have it. Here's my callback... everything here works: // Media API callback void CALLBACK AudioRecorder::waveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD dwInstance, DWORD dwParam1, DWORD dwParam2) { // Data received if (uMsg == WIM_DATA) { // Get wav header LPWAVEHDR mBuffer = (WAVEHDR *)dwParam1; // Now what? for (unsigned i = 0; i != mBuffer->dwBytesRecorded; ++i) { // I can see the char, how do get them into my file and audio buffers? cout << mBuffer->lpData[i] << "\n"; } // Re-use buffer mResultHnd = waveInAddBuffer(hWaveIn, mBuffer, sizeof(mInputBuffer[0])); // mInputBuffer is a const WAVEHDR * } } // waveInOpen cannot use an instance method as its callback, // so we create a static method which calls the instance version void CALLBACK AudioRecorder::staticWaveInProc(HWAVEIN hWaveIn, UINT uMsg, DWORD_PTR dwInstance, DWORD_PTR dwParam1, DWORD_PTR dwParam2) { // Call instance version of method reinterpret_cast<AudioRecorder *>(dwParam1)->waveInProc(hWaveIn, uMsg, dwInstance, dwParam1, dwParam2); } Like I said, it works great, but I'm trying to do the following: Convert the data to short and copy into an array Convert the data to float and copy into an array Copy the data to a larger char array which I'll write into a WAV Relay the data to an arbitrary output device I've worked with FMOD a lot and I'm familiar with interleaving and all that. But FMOD dishes everything out as floats. In this case, I'm going the other way. I guess I'm basically just looking for resources on how to go from LPSTR to short, float, and unsigned char. Thanks much in advance!

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  • Showing multiple elements onClick

    - by Nimbuz
    HTML: <ul id="mode"> <li><a href="#tab1">tab1</a> <div class="extra">tada</div> </li> <li><a href="#tab2">tab2</a> <div class="extra">tada</div> </li> </ul> <div id="tab1" class="tab-content" style="display: none">content 1</div> <div id="tab2" class="tab-content" style="display: none">content 2</div> ? jQuery: $(function(){ var mode = $('#mode'); var arrow = $('<span/>', {'class': 'arrow'}); $('li a', mode).bind('click.mytabs', function() { $('li', mode).removeClass('active'); $(this).parent().addClass('active').append(arrow); var a = $(this).attr('href'); $('.tab-content').hide(); $(a).show(); return false; }).filter(':first').triggerHandler('click.mytabs'); // eq(0) works as well }); JSFiddle here: http://jsfiddle.net/wwMJL/ I'd like to also show each li's 'extra' div on click and hide when the tab is inactive, what do I need to change in the code? Thanks!

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  • Filter design for audio signal.

    - by beanyblue
    What I am trying to do is simple. I have a few .wav files. I want to remove noise and filter out specific frequencies. I don't have matlab and I intend to write my own code for all the filters. Right now, I have a way to read the .wav file and dump out the structure into a text file. My questions are the following: Can I directly apply the digital filters on this sampled data?{ ie, can I directly do a convolution between my input samples and h(n) for the filter function that i choose?). How do I choose the number of coefficients for the Window function? I have octave, so if someone can point me to anything that gives me some idea on how to process the .wav file using octave, that would be great too. I want to be able to filter out the frequency and then listen to the sound again. Is this possible with octave? I'm just a beginner with these kinds of things, so please bear with me if my questions are too naive. Any help will be great.

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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  • Change Audio title from English to Sinhalese using ffmpeg

    - by user330461
    I insert an extra Sound track in my video file and it works well. ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 1 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 -an -y /dev/null && ffmpeg -i news.mov -i news.wav -map 0:0 -map 0:1 -map 1:0 -pass 2 -acodec libfaac -ab 128k -ac 2 -vcodec libx264 -preset fast -b 512k -minrate 512k -maxrate 512k -bufsize 512k -threads 0 -f mp4 news.mp4 The default audio track come with the label "English" and I would like to give it a label "Sinhalese" The Second Audio track come up without a label as "track#1" and I would like to give that a label of "Tamil". How do I do that ?

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  • merging video and audio with custom panning

    - by cherouvim
    I have: a video which has mono audio inside a audio (mono) I'd like to merge those two to a single video file containing: video from #1 audio from #1 full left pan + audio #2 full right pan Is this possible in ffmpeg using 1 command? I've tried the following which almost does this but the video/audio gets out of sync: ffmpeg -i video.mp4 -filter_complex "amovie=audio.wav [r] ; [r] amerge" output.mp4 -y I've managed to do it with multiple commands: #1 create right panned audio ffmpeg -i audio.wav -ac 2 -vbr 5 audio-stereo.mp3 -y ffmpeg -i audio-stereo.mp3 -af pan=stereo:c1=c1 audio-right.mp3 -y #2 create left panned video ffmpeg -i video.mp4 -af pan=stereo:c0=c0 video-left.mp4 -y #3 merge the two ffmpeg -i video-left.mp4 -i audio-right.mp3 -filter_complex "amix=inputs=2" video-mixed.mp4 -y It does the job, but is it possible with 1 command?

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  • Notification area balloon tip pop sound in Windows 7

    - by Worm Regards
    When I was using Windows XP, there was a distinct sound when an application showed a balloon tip in the notification area (aka system tray). Unfortunately, I didn't look any deeper into it. Now Windows 7 has this behavior disabled by default and I do not know how to configure it. Discovered the name of sound file used to accompany tray balloon tips in Windows XP Windows XP Balloon.wav More clues: interesting registry key is HKEY_USERS\XP Registry Hive\AppEvents\Schemes\Apps\.Default\SystemNotification\.Default Default value is %SystemRoot%\media\Windows XP Balloon.wav So, the System Notification event label appears to be correct, but tray balloons are silenced elsewhere.

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  • No audio in Google Chrome

    - by Z9iT
    I started with Ubuntu 12.04 Minimal. Then installed only 3 utils sudo apt-get install xorg xinit google-chrome-stable alsa-base alsa-utils alsa-oss I have added google-chrome to .xinitrc file. Used sudo alsamixer to unmute everything using M. Also I am able to hear sound when I run this independently in a terminal sudo aplay /usr/share/sounds/alsa/Front_Center.wav However Google Chrome is not giving any sound output be it on youtube or the same file (/usr/share/sounds/alsa/Front_Center.wav) opened by browsing in chrome. UPDATE : the moment i install some Desktop (display) Manager like gnome or lxde and launch chrome then, the audio is perfect success. However if i kill the xsession and the desktop manager (lxde) AND then start with loading only the chrome (without DM) then again i loose the sound. This makes me wonder that there is something which is not allowing the sound to be loaded into chrome directly, but once the session like lxde loads, then it works flawless. I am thinking that i should rather ask, how to authorize google-chrome to use sound software? Miscellaneous : I am surprised to know that I cannot start google-chrome by sudo command (it asks to be a normal user) && that i cannot start alsamixer as a normal user (i must use sudo alsamixer ) May someone please help what i need to do so that google chrome speaks????

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  • Prevent Windows Exporer to extract metadata

    - by olafure
    Windows Explorer (windows 7 x64) crashes when it sees allegedly corrupted .wav files. I'm dealing with this problem and the hotfix doesn't work for me: http://support.microsoft.com/kb/976417/en-us The hotfix says that this happens if the .wav file is corrupt (which btw I don't think it is). What makes this even worse is that I can't access the file in any program! As soon as the open dialog sees the file, windows tries it's metadata extraction trick and exporer.exe halts. So my question: Can I by any means tell windows to stop this "metadata extraction" action ? (I have seen multiple problems associated with it in the past).

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  • How can I use the Band Pass Filter in GarageBand?

    - by Another Registered User
    What I want to do: I have a music WAV file and want to put a Band Pass Filter over it, to filter out anoying high frequencies. I was reading on the net that there is a "AuBandPass" plugin in Mac OS X. I just can't figure out how I could use that in GarageBand. I don't even find the effects at all. I created a new GarageBand file and dropped the WAV file in there. Now I can play that song in GarageBand. What must I do next?

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  • 8 New Features in Ubuntu 12.10, Quantal Quetzal

    - by Chris Hoffman
    Ubuntu 12.10 has been released and you can download it now. From better integration with web apps and online services to improvements in Unity, there are quite a few changes – although none of them are huge or groundbreaking. The list of new features may be more exciting next time around, with Mark Shuttleworth promising secret development of new “tada!” features that will be unveiled closer to Ubuntu 13.04’s release. Can Dust Actually Damage My Computer? What To Do If You Get a Virus on Your Computer Why Enabling “Do Not Track” Doesn’t Stop You From Being Tracked

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  • [Silverlight] DataGrid

    - by nCdy
    I'm making tutorials. Silverlight + MSSQL And I'm on the last step when it says "Copy-paste my code and tada it will works"... :-/ But after I added System.Windows.Controls.Data reference it still can't find Error 3 The type or namespace name 'Data' does not exist in the namespace 'System.Windows.Controls' (are you missing an assembly reference?) I really don't miss the reference ... Maybe I need to add it somehow else or ... I really have no idea. (VSWDEE2010)

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  • How do I get a mp3 file's total time in Java?

    - by Tom Brito
    The answers provided in How do I get a sound file’s total time in Java? work well for wav files, but not for mp3 files. They are (given a file): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(file); AudioFormat format = audioInputStream.getFormat(); long frames = audioInputStream.getFrameLength(); double durationInSeconds = (frames+0.0) / format.getFrameRate(); and: AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(file); AudioFormat format = audioInputStream.getFormat(); long audioFileLength = file.length(); int frameSize = format.getFrameSize(); float frameRate = format.getFrameRate(); float durationInSeconds = (audioFileLength / (frameSize * frameRate)); They give the same correct result for wav files, but wrong and different results for mp3 files. Any idea what do I have to do to get the mp3 file's duration?

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  • SoundPlayer causing Memory Leaks?

    - by Nick Udell
    I'm writing a basic writing app in C# and I wanted to have the program make typewriter sounds as you typed. I've hooked the KeyPress event on my RichTextBox to a function that uses a SoundPlayer to play a short wav file every time a key is pressed, however I've noticed after a while my computer slows to a crawl and checking my processes, audiodlg.exe was using 5 GIGABYTES of RAM. The code I'm using is as follows: I initialise the SoundPlayer as a global variable on program start with SoundPlayer sp = new SoundPlayer("typewriter.wav") Then on the KeyPress event I simply call sp.Play(); Does anybody know what's causing the heavy memory usage? The file is less than a second long, so it shouldn't be clogging the thing up too much.

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  • [[NSURL alloc] initFileURLWithPath:(NSString)] returns null

    - by Ajay Pandey
    NSString *soundFilePath = [[NSBundle mainBundle] pathForResource:@"Opening" ofType:@"wav"]; NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; NSLog(@"@ajay"); AVAudioPlayer *audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:fileURL error:nil]; [fileURL release]; [audioPlayer play]; i have inserted a wav file in my project.But NSURL *fileURL = [[NSURL alloc] initFileURLWithPath: soundFilePath]; returns NULL and console prints following: and application KILLS... Can someone help me?

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  • Pipe ffmpeg to oggenc(2) with .NET

    - by acidzombie24
    I am trying to encode ogg files at -q 6/192k. ffmpeg doesnt seem to be listenting to my command ffmpeg -i blah -acodec vorbis -ab 192k -y out.ogg So i would like to use the recommended ogg encoder. It says the input must be wav or similar. I would like to pipe a wav from ffmpeg to ogg. However not only am i unsure if oggenc2 will accept input from stdin, i have no idea how to pipe one process to another inside of .net using the Process class.

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  • How does one capture H.323 voice traffic on a VOIP network?

    - by Chris Holmes
    What I am trying to do is capture the WAV data of a phone conversation on a VOIP network using SharpPCap/PCap.Net. We are using the H.323 recommendation and my understanding is that voice data is located in the RTP packets. However, there is no way to heuristically determine if a UDP packet is a RTP packet, so we have to do more work before we can capture the data. The H.323 recommendation apparently uses a lot of traffic on specific TCP ports to negotiate the call before the WAV data is sent via RTP. However, I am having very little luck determining what data is actually sent on those TCP ports, when it is sent, what the packets look like, how to handle it, etc. If anyone has any information on how to go about this I'd really appreciate it. My Google-Fu seems to be failing me on this one.

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  • feature extraction from acoustic signals

    - by Dolphin
    Hi everyone, It's been a while. I found APIs in Java for extracting features from acoustic audio files and symbolic files separately. But now I have a problem in mapping from low level wav audio features to high level midi features. i.e. I need to write the extracted wav audio features on to midi format. But I cannot think of anything even close to it. Can someone pls provide me some insight as in how I can approach this. Greatly appreciate your responses. Advance thanks

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