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  • FMOD.net streaming, callback and exinfo parameters

    - by Tesserex
    I posted a question on gamedev about how to play nsf files (NES console music) in FMOD. It didn't get any results, but since then I made some progress. I decided that the easiest method was just to compile an existing player into a dll and then call it from C# to populate my buffer. The problem now is getting it to sound right, and making sure all my paremeters are correct. Here are the facts so far: The nsf dll is dealing with shorts, so the data is PCM16. The sample nsf I'm using has a playback rate of 60 Hz. Just for playing around now, I'm using a frequency of 48000. Based on 2 and 3, the dll calculates a necessary buffer size of 48000 / 60hz = 800. This means it will render 800 shorts worth of buffer for every simulated NES frame. I've so far got my C# code to play the nsf, at the correct pitch and tempo, but it's very grainy / fuzzy, which I'm attributing to the fact that the FMOD read callback is giving a data length of 1600, whereas I should be expecting 800. I've tried playing around with all the numbers and it either crashes, or the music changes pitch, tempo, or both. Here's some of my C# code: uint channels = 1, frequency = 48000; FMOD.MODE mode = (FMOD.MODE.DEFAULT | FMOD.MODE.OPENUSER | FMOD.MODE.LOOP_NORMAL); FMOD.Sound sound = new FMOD.Sound(); FMOD.CREATESOUNDEXINFO ex = new FMOD.CREATESOUNDEXINFO(); ex.cbsize = Marshal.SizeOf(ex); ex.fileoffset = 0; ex.format = FMOD.SOUND_FORMAT.PCM16; // does this even matter? It doesn't change my results as long as it's long enough for one update ex.length = frequency; ex.numchannels = (int)channels; ex.defaultfrequency = (int)frequency; ex.pcmreadcallback = pcmreadcallback; ex.dlsname = null; // eventually I will calculate this with frequency / nsf hz, but I'm just testing for now ex.decodebuffersize = 800; // from the dll load_nsf_file("file.nsf", 8, (int)frequency); // 8 is the track number to play var result = system.createSound( (string)null, (mode | FMOD.MODE.CREATESTREAM), ref ex, ref sound); channel = new FMOD.Channel(); result = system.playSound(FMOD.CHANNELINDEX.FREE, sound, false, ref channel); private FMOD.RESULT PCMREADCALLBACK(IntPtr soundraw, IntPtr data, uint datalen) { // from the dll process_buffer(data, (int)800); // if I use datalen, it usually crashes (I can't get datalen to = 800 safely) return FMOD.RESULT.OK; } So here are some of my questions: What is the relationship between exinfo.decodebuffersize, frequency, and the datalen parameter of the read callback? With this code sample, it's coming in as 3200. I don't know where that factor of 4 between it and the decodebuffersize comes from. Is datalen in the callback referring to number of bytes, or shorts? The process_buffer function takes a short array and its length. I would expect fmod is talking about shorts as well because I told it PCM16. Maybe my playback quality is bad for some totally different reason. If so I have no idea where to begin solving that. Any ideas there?

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  • I am having trouble using FileReader to write a txt file to an array (Java), what am I doing wrong?

    - by deliriumtremens
    Scanner s = null; try { s = new Scanner(new BufferedReader(new FileReader("rates.txt"))); for (int i=0; i<9; i++){ while(s.hasNext()){rates[i] = s.next();} System.out.println(rates[i]); } }catch (IOException e){ System.out.println(e); } finally { if (s != null) { s.close(); } } When I run this code, it reads the last chunk of characters in my txt file, places them in rates[0], sticks null in 1-8, then puts that same last chunk in rates[9]. I'm not sure why it's reading the end of my file first. The contents of the txt are below.. USD 1.34 EUR 1.00 JPY 126.28 GBP 0.88 INR 60.20 It reads the 60.20, which is all it is recording in the array. Any help would be appreciated.

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  • SQL group and order

    - by John Lambert
    I have multiple users with multiple entries recording times they arrive at destinations Somehow, with my select query I would like to only show the most recent entries for each unique user name. Here is the code that doesn't work: SELECT * FROM $dbTable GROUP BY xNAME ORDER BY xDATETIME DESC This does the name grouping fine, but as far as showing ONLY their most recent entry, is just shows the first entry it sees in the SQL table. I guess my question is, is this possible? Here is my data sample: john 7:00 chris 7:30 greg 8:00 john 8:15 greg 8:30 chris 9:00 and my desired result should only be john 8:15 chris 9:00 greg 8:30

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  • Inserting snippets from macros

    - by Matt
    In Visual Studio 2008 I had a macro which would insert a snippet and input today's date in one of the replacements. When I try to run this macro in VS 2010 it doesn't work. No matter how I try it will not insert a snippet. When I try the following command: DTE.ExecuteCommand("Edit.InvokeSnippetFromShortcut", "snippetName") This fails with error "Error HRESULT E_FAIL has been returned from a call to a COM component. I tried recording a new macro and when I pressed the keyboard shortcut for inserting a snippet (Ctrl+K, Ctrl+X) VS gave the error "The command Insert Snippet is not currently available." Did MS remove the ability to insert snippets from macros? If so this really is a shame because macros have functionality not found in snippets and vice versa.

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  • Is it possible to have AVFramework and AudioToolbox framework in one app?

    - by Satyam
    I'm trying to write develop audio related application. In that, I'm using AudioToolBox framework for recording the sound. And I'm using AVFramework to play soudns. When app is stared, it will play some mp3 file using AVFramework. And also initializes Audiotoolbox. In simulator, I'm able to record and play. But when I'm testing it on iPhone, I'm getting following error for initializing AudioToolBox. 2009-12-11 22:25:51.599 StoryBook[807:207] AudioRecorder init AudioSessionInitialize failed with error: 1768843636 Can some one tell me whether we can use both AV as well as Audio Toolbox frame works in one application? Why I'm getting that error?

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  • C# Sequence Diagramm Reverse Engineering Tool?

    - by Wayne
    It's essential for me to find a tool that will reverse engineer sequence diagrams by integrating with the debugger. I suppose using the profiler could work also but less desirable. It's a key requirement that the tool in question will record all threads of execution since the app, TickZoom, is heavily parallelized. We just evaluated a most awesome tool from Sparx called Enterprise Architect which integrates with the debugger. It allows you to set a break point, start recording method traces from that break point. It's a lovely design and GUI. Hope it works for you but it only records a single thread of execution so that makes it unusable for us. I will put in a feature request to Sparx. But hope to find a similar tool that does this now since that's the only features we need--not all the other stuff that Sparx also does rather well. Sincerely, Wayne

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  • Python proxy an application

    - by sharvey
    Does anyone know of a library that enables you to run an application inside some kind of sandbox, with virtual mouse and keyboard support. The use case would be to create some kind of visual test runner, that would replay all actions taken during recording and play them back. So far I found autopy, but the fact that it controls the real mouse position is problematic, because it prevents user interaction with other tools (debugger or anything) while running. Cross platform would be nice, but either windows or os x is fine. Python would be ideal but anything that you could create python bindings for would be ok too.

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  • Unselected UIPickerView value

    - by morticae
    According to the documentation, if a UIPickerView has no selected value, the expected return from selectedRowInComponent: should be: "A zero-indexed number identifying the selected row, or -1 if no row is selected." However, if I check the value the very line after initializing one, its value is 0. Even if I then manually set it to -1, it still returns 0. I would like to be able to detect whether the user has chosen a value yet or not, without recording it in a local variable. Is this possible? example: UIPickerView *picker = [[UIPickerView alloc] initWithFrame:CGRectMake(0.0, 46.0, 320.0, 216.0)]; [picker selectRow:-1 inComponent:0 animated:NO]; NSLog(@"SELECTED %d", [picker selectedRowInComponent:0]); expected output: SELECTED -1 actual output: SELECTED 0

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  • How to associate newly created SurfaceHolder to MediaPlayer

    - by fayerth
    In my code, I want to be able to temporarily hide (and subsequently show) a video. I am using a SurfaceView + MediaPlayer combination instead of the VideoView due to requirements. However, I am facing difficulties in getting video playback to occur as expected after I show the SurfaceView. My code excerpts includes the following: public void show() { if (mSurface != null) mSurface.setVisibility(View.VISIBLE); } public void hide() { if (mSurface != null) { if (isInPlaybackState()) pause(); mSurface.setVisibility(View.INVISIBLE); } } @Override public void surfaceCreated(final SurfaceHolder holder) { mHolder = holder; openVideo(); } @Override public void surfaceDestroyed(final SurfaceHolder holder) { // After this, the surface can't be used again mHolder = null; } private void openVideo() { if (mAssetPath == null || !mAssetPath.isEmpty() || mHolder == null) { // Not ready yet; try again later return; } // Pause music playback service Intent i = new Intent("com.android.music.musicservicecommand"); i.putExtra("command", "pause"); getActivity().sendBroadcast(i); if (mPlayer == null) { initializePlayer(); } else { mPlayer.setDisplay(mHolder); } } Based on the above, when I call hide(), surfaceDestroyed(SurfaceHolder) gets triggered. When I later call show(), surfaceCreated(SurfaceHolder) gets triggered, which will call openVideo() and associate the player with the newly provided SurfaceHolder. The above works as expected, and I believe this should be the correct process; however, when I call mPlayer.start(), I would hear the video's audio playing without any video and see the following error messages (which eventually causes the media playback to stop and complete, as noted by the disconnect logs): 10-23 11:29:42.775: E/MediaPlayer(4204): Error (1,-2147483648) 10-23 11:29:42.795: E/MediaPlayer(4204): Error (-38,0) 10-23 11:29:42.805: E/MediaPlayer(4204): Error (1,-2147483648) 10-23 11:29:42.810: V/MediaPlayer(4204): message received msg=100, ext1=1, ext2=-2147483648 10-23 11:29:42.810: E/MediaPlayer(4204): error (1, -2147483648) 10-23 11:29:42.810: V/MediaPlayer(4204): callback application 10-23 11:29:42.810: V/MediaPlayer(4204): back from callback 10-23 11:29:42.825: E/MediaPlayer(4204): Error (1,-2147483648) 10-23 11:29:42.850: V/MediaPlayer-JNI(4204): getCurrentPosition: 671668 (msec) 10-23 11:29:42.850: V/MediaPlayer-JNI(4204): getCurrentPosition: 671668 (msec) 10-23 11:29:42.850: V/MediaPlayer(4204): message received msg=100, ext1=1, ext2=-2147483648 10-23 11:29:42.850: E/MediaPlayer(4204): error (1, -2147483648) 10-23 11:29:42.850: V/MediaPlayer(4204): callback application 10-23 11:29:42.850: V/MediaPlayer(4204): back from callback 10-23 11:29:42.875: V/MediaPlayer-JNI(4204): stop 10-23 11:29:42.875: V/MediaPlayer(4204): stop 10-23 11:29:42.875: E/MediaPlayer(4204): stop called in state 0 10-23 11:29:42.875: V/MediaPlayer(4204): message received msg=100, ext1=-38, ext2=0 10-23 11:29:42.875: E/MediaPlayer(4204): error (-38, 0) 10-23 11:29:42.875: V/MediaPlayer(4204): callback application 10-23 11:29:42.875: V/MediaPlayer(4204): back from callback 10-23 11:29:42.875: V/MediaPlayer-JNI(4204): reset 10-23 11:29:42.875: V/MediaPlayer(4204): reset 10-23 11:29:42.900: V/MediaPlayer-JNI(4204): release 10-23 11:29:42.900: V/MediaPlayer(4204): setListener 10-23 11:29:42.900: V/MediaPlayer(4204): disconnect 10-23 11:29:42.910: V/MediaPlayer(4204): destructor 10-23 11:29:42.910: V/MediaPlayer(4204): disconnect Has anyone encountered this issue before and found a workaround? Or would the only option be to create a new MediaPlayer as well?

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  • Sequence Diagram Reverse Engineering Tool?

    - by Wayne
    It's essential for me to find a tool that will reverse engineer sequence diagrams by integrating with the debugger. I suppose using the profiler could work also but less desirable. It's a key requirement that the tool in question will record all threads of execution since the app, TickZoom, is heavily parallelized. We just evaluated a most awesome tool from Sparx called Enterprise Architect which integrates with the debugger. It allows you to set a break point, start recording method traces from that break point. It's a lovely design and GUI. Hope it works for you but it only records a single thread of execution so that makes it unusable for us. I will put in a feature request to Sparx. But hope to find a similar tool that already does this since that's the only feature we need--not all the other amazing features that Sparx Enterprise Architect appears to do rather well.

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  • Looking for some thoughts on an image printing app

    - by Alex
    Hey All, Im looking for thoughts/advice. I have an upcoming project (all .net) that will require the following: pulls data once a day from an online service provider based on certain criteria. saves data locally for reference and reporting the data thats pulled will be used to create gift cards. So after the data is loaded, a process will run to generate "virtual cards" and send them to a network printer. Once printed, the system will updated the local data recording a successful or failed print. My initial thought was to create a windows service to pull the data...but then I couldnt decide how I was going to put a "virtual card" together and get it to print. Then I considered doing it as a WPF app. I figure that will give me access to the graphics and printing ability. Maybe neither of these are the right direction....Any ideas or thoughts would be greatly appreciated. Alex

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  • Novocaine - How to loop file playback? (iOS)

    - by lppier
    I'm using Novocaine by alexbw Novocaine for my audio project. I'm playing around with the example code here for file reading. The file plays back with no problem. I would like to loop this recording with the gap between the loops - any suggestion as to how I can do so? Thanks. Pier. // AUDIO FILE READING OHHH YEAHHHH // ======================================== NSArray *pathComponents = [NSArray arrayWithObjects: [NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES) lastObject], @"testrecording.wav", nil]; NSURL *inputFileURL = [NSURL fileURLWithPathComponents:pathComponents]; NSLog(@"URL: %@", inputFileURL); fileReader = [[AudioFileReader alloc] initWithAudioFileURL:inputFileURL samplingRate:audioManager.samplingRate numChannels:audioManager.numOutputChannels]; [fileReader play]; [fileReader setCurrentTime:0.0]; //float duration = fileReader.getDuration; [audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels) { [fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels]; NSLog(@"Time: %f", [fileReader getCurrentTime]); }];

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  • HTML - Word Doc Images

    - by Michael
    Okay I have roughly 150 + pages of procedures all written in MS word. The person who wrote the procedures did an excellent job of recording the process how to perform specific tasks. This individual went though and created screen shots a MS word doc. There are approx 2 screen shots per page. So it is roughly 300 images and I do not want to recreate the wheel. Does anyone know a quick way of handling the images? The written portion is pretty straightforward but the images is what I am struggling with. Regards, Mike

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  • Web Performance testing using VS2010 "Testing a file download"

    - by cheedep
    Hi All, I am trying out the VS 2010 testing tools for the first time. And I tried recording a web performance test and my actions had a file download implemented as in the KB article here http://support.microsoft.com/kb/812406 by streaming chunks of 10000 bytes. However my test is failing at the download saying "The response stream has been closed". Please help me understand why it is happening this way also any suggestions how you would test such a file download. My main aim was to see how the download was performing for a load test with Intercontinental 350kbps connection on files of about 30-50 MB. Thanks.

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  • Looking for "tech call" tracking software.

    - by jacook11
    The company I work for is looking for the best way to track "tech calls". We would most likely develop in house using vb.net, but possibly could look at using some open source vb.net software already out there. We will probably want to track just the basic info like client, datetime, length of call & a notes section about the call. One idea that has floated around is recording everyone's calls, watching a directory for new files and popping up a form so the user can enter the info when the call is over. We really don't want to spend a lot of time tracking/logging these calls, something quick & simple. Anybody have a good idea or solution that they have used before?

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  • Prevent Visual Studio Web Test from changing request details

    - by keithwarren7
    I have a service that accepts Xmla queries for Analysis services, often times those queries themselves will have a string that contains a fragment that looks something like {{[Time].[Year].[All]}} Recording these requests works fine but when I try to re-run the test I get an error from the test runner... Request failed: Exception occurred: There is no context parameter with the name ' [Time].[Year].[All]' in the WebTestContext This was confusing for some time but when I asked VS to generate a coded version of the test I was able to see the problem a bit better. VS searches for the '{{' and '}}' tokens and makes changes, considering those areas to refer to Context parameters, the code looks like this.Context["\n\t[Time].[Year].[All]"].ToString() Anyone know how to instruct Visual Studio to not perform this replacement operation? Or another way around this issue?

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  • Determining the magnitude of a certain frequency on the iPhone

    - by eagle
    I'm wondering what's the easiest/best way to determine the magnitude of a given frequency in a sound. It's my understanding that a FFT function will return the magnitudes of all frequencies in a signal. I'm wondering if there is any shortcut I could use if I'm only concerned about a specific frequency. I'll be using the iPhone mic to record the audio. My guess is that I'll be using the Audio Queue Services for recording since I don't need to record the audio to a file. I'm using SDK 4.0, so I can use any of the functions defined in the Accelerate framework (e.g. FFT functions) if needed. Update: I updated the question to be more clear as per Conrad's suggestion.

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  • Using Accelerometer in Wiimote for Physics Practicals

    - by Omar
    I have to develop some software in my school to utilize the accelerometer in the Wiimote for recording data from experiments, for example finding the acceleration and velocity of a moving object. I understand how the accelerometer values will be used but I am sort of stuck on the programming front. There is a set of things that I would like to do: Live streaming of data from the Wiimote via bluetooth Use the accelerometer values to find velocity and displacment via integration Plot a set of results Avoid the use of the infrared sensor on the Wiimote Please can anyone give me their thoughts on how to go about this. Also it would be great if people could direct me to existing projects that utizlise the wiimote. Also can someone suggest what would be the best programming language to use for this. My current bet is on using Visual basic. Any sort of help is greatly appretiated.

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  • Which is better for quickly developing small utilities? AutoIt or AutoHotKey?

    - by Abhijeet Pathak
    Which is better for quickly developing small utilities? AutoIt or AutoHotKey or something else? I need to develop some small software for which I think using some professional suite like Visual Studio will be overkill. Most of the macro recording tools like AutoIt or AutoHotKey provide enough power to write decent application. Plus they are small and free. Which option will be good? Using one of these tools or using some other small/free compiler?

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  • what wrong in the program using tie and array

    - by SCNCN2010
    File : #comment1 #comment2 #comment3 #START HERE a: [email protected] b: [email protected] my perl program : use Data::Dumper; use Tie::File; tie my @array, 'Tie::File', 'ala.txt' or die $!; my $rec = 'p: [email protected]'; my $flag =1 ; my $add_flag = 0; for my $i (0..$#array) { next if ($array[$i] =~ /^\s*$/); if ( $flag == 1 ) { if ($array[$i] =~ /#START HERE/ ) { $flag = 0; } else { next ; } } if (($array[$i] cmp $rec) == 1) { splice @array, $i, 1, $rec; $add_flag = 1; last ; } } if ( $add_flag == 0 ) { my $index = $#array+1; $array[$index] = $rec ; } the recording adding end of file always . I am trying to add to middle or begin or end like aplphbetical order

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  • What exactly does raw microphone data represent?

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

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  • Properly trimming PCM data from a ByteArray

    - by Lowgain
    I have a situation where I need to trim a small amount of audio from the beginning of a recorded clip (generally somewhere between 110-150ms, it is an inconsistent amount). I'm recording in 44100 frequency and 16 bitrate. This is the code I'm using: public function get trimmedData():ByteArray { var ba:ByteArray = new ByteArray(); var bitPosition:uint = 44100 * 16 * (recordGap / 1000); bitPosition -= int(bitPosition % 16); //should keep snapped to nearest sample, I hope ba.writeBytes(_rawData, (bitPosition / 8)); return ba; } This seems to work time-wise, but all the recorded audio gets staticy and gross. Is something off about my rounding? This is the first time I've needed to alter raw PCM data so I'm not sure about the finer details of it. Thanks!

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  • Is there an easy way of obtaining the total page response time in ASP.Net?

    - by Earlz
    Hello, commonly on say PHP or other web frameworks getting the total response time is easy, just start the timer at the top of the file and stop it at the end. In ASP.Net there is the whole Page Lifecycle bit though so I'm not sure how to do this. I would like for this response time recording to take place in a master page and the response time show up in the footer of pages. What would be the best way of doing this? Is there something built in to ASP.Net for it? Is it even possible to include the OnRender time?

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  • Why does Silverlight provides webcam and microphone support without any encoding API?

    - by Shurup
    In the list of new features in Silverlight 4 you will find following: Webcam and microphone to allow sharing of video and audio for instance for chat or customer service applications. Silverlight captures an audio stream as raw pcm. So how would you realize for example audio/video chat or client/server audio recording application without any encoding on the client side, where there is no APIs in Silverlight available? Much less in a Silverlight you cannot use an unmanaged dll. You can use a com automation (a new feature of the Silverlight 4, I think only for Windows) but only if it was already installed on the client side (do you know any encoding COM servers that are installed with the windows). Otherwise, how would you deploy a custom COM server within you Silverlight application? The only way I found is either to deploy a command-line encoding and use it with COM AutomationFactory.CreateObject("WScript.Shell") or to implement an encoding to use it in your own AudioSink.

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  • [Android SDK] Text-To-Speech addSpeech not working properly

    - by arcoraven
    Hi, I'm trying to get my Android app to play a .wav file recording of the word "Spinach Salad" whenever it sees that phrase being spoken by TTS. Here's the relevant code: spinach_salad.wav is located in /res/raw prodName = "Spinach Salad" mTts.addSpeech(prodName, "com.example.textextractor", R.raw.spinach_salad); ...and later in the code: mTts.speak("blah blah blah " + prodName, TextToSpeech.QUEUE_ADD, null); I've also tried: mTts.speak("blah blah blah Spinach Salad", TextToSpeech.QUEUE_ADD, null); and mTts.speak("blah blah blah", TextToSpeech.QUEUE_ADD, null); mTts.speak(productName_str, TextToSpeech.QUEUE_ADD, null); In both cases, I'm just hearing the TTS synthesized audio, rather than my custom .wav file. (On a related note, the last chunk of code sometimes speaks out of order, saying the second line before the first).

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