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  • Online audio stream using ruby on rails

    - by Avdept
    I'm trying to write small website that can stream audio online(radio station) and got few questions: 1. Do i have to index all my music files into database, or i can randomily pick file from file system and play it. 2. When should i use ajax to load new song(right after last finished, or few seconds before to get responce from server with link to file?) 3. Is it worth to use ajax, or better make list, that will play its full time and then start over?

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  • Linux, C++ audio capturing (just microphone) library

    - by TheOm3ga
    I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

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  • How to play audio file ios

    - by Camus
    I am trying to play an audio file but I can get it working. I imported the AVFoundation framework. Here is the code: NSString *fileName = [[NSBundle mainBundle] pathForResource:@"Alarm" ofType:@"caf"]; NSURL *url = [[NSURL alloc] initFileURLWithPath:fileName]; NSLog(@"Test: %@ ", url); AVAudioPlayer *audioFile = [[AVAudioPlayer alloc] initWithContentsOfURL:url error:NULL]; audioFile.delegate = self; audioFile.volume = 1; [audioFile play]; I am receiving an error nil string parameter I copied the file to the supporting files folder so the file is there. Can you guys help me? Thanks

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  • Audio File continues to play even on leaving the view

    - by Swastik
    What I am doing is -(void)viewWillAppear:(BOOL)animated{ [NSTimer scheduledTimerWithTimeInterval:0.3 target:self selector:@selector(clickEvent:) userInfo:nil repeats:YES]; } -(void)clickEvent:(NSTimer *)aTimer{ NSDate* finishDate = [NSDate date]; if([finishDate timeIntervalSinceDate: self.startDate] 11 && touched == NO){ NSString *mp3Path = [[[NSBundle mainBundle] resourcePath] stringByAppendingPathComponent:@"test.mp3"]; [self playMusicFile:mp3Path]; NSLog(@"Timer from First Page"); [aTimer invalidate]; //[touchCheckTimer release]; aTimer = nil; } else{ } -(void)playMusicFile:(NSString *)mp3Path{ NSURL *mp3Url = [NSURL fileURLWithPath:mp3Path]; NSError *err; AVAudioPlayer *audPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:mp3Url error:&err]; [self setAudioPlayer1:audPlayer]; if(audioPlayer1) [audioPlayer1 play]; [audPlayer release]; } Now, on pushing another view this audio file keeps playing in the background. Please help!

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  • Background audio not working in windows 8 store / metro app

    - by roryok
    I've tried setting background audio through both a mediaElement in XAML <MediaElement x:Name="MyAudio" Source="Assets/Sound.mp3" AudioCategory="BackgroundCapableMedia" AutoPlay="False" /> And programmatically async void setUpAudio() { var package = Windows.ApplicationModel.Package.Current; var installedLocation = package.InstalledLocation; var storageFile = await installedLocation.GetFileAsync("Assets\\Sound.mp3"); if (storageFile != null) { var stream = await storageFile.OpenAsync(Windows.Storage.FileAccessMode.Read); _soundEffect = new MediaElement(); _soundEffect.AudioCategory = AudioCategory.BackgroundCapableMedia; _soundEffect.AutoPlay = false; _soundEffect.SetSource(stream, storageFile.ContentType); } } // and later... _soundEffect.Play(); But neither works for me. As soon as I minimise the app the music fades out

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  • iphone - Images (slide show) and audio snychronization

    - by Qaiser
    I have 20 images and some audio. I would like to show a single image at a time and change the images at (unequal) intervals. For example, I want to show image 1 for 1.44 seconds and image 2 for 1.67 seconds and so on. Can someone suggest how to go about doing this please? What I have seen are examples that show how to setup an array of images with one field that denotes total time. This causes the images to show for an equal amount of time (each). ... and that not what I am looking for ...

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  • Record/Playback with AudioQueue on iPhone

    - by Biranchi
    Hi, I am currently using Audio Queues on the iPhone to record and playback audio. What I would like to be able to do is to record some audio, allow the user to pause the record queue, and to seek back and forward through the audio to select a position from where they can start recording from again. I have got over the seeking issue by making the playback AudioQueueBuffer sizes small enough so that the play audio queue callback happens at a rate that allows the user to use a slider control to hear the audio as they adjust the slider back and forth. I think I can achieve the recording at a new position by setting the inStartingPacket parameter of the AudioFileWritePackets function that I call from the Audio Recording Queue callback. The trouble is this only inserts audio over the previously recorded audio. The file length obviously doesn't change so if the user were to go backwards and record less audio than before, the old audio still remains after the end of the newly recorded audio. Is there a way I can get the AudioFile to truncate at the point the user starts to insert the new audio, is there some other way I can remove the old audio starting at the insert position or is there a better way about going about this task? Thanks

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  • audio cd s not burning to mp3 format-burning to wav format in k3b and brasero using ubuntu 12.04.2

    - by robert
    It started in ubuntu 13.04-I was doing what I usually do,I opened brasero to make an audio cd from a few mp3 audio files..When burned I noticed the files on cd were in wav format.I then tried k3b with the same result.At that point and because of several issues with 13.04 I formatted my hdd and dropped back to ubuntu 12.04.On 12.04 I tried brasero and k3b once again with same results.I know that when I used to burn cd s using brasero they were burned to cd in mp3 format not wave.Can anyone tell me a fix for this?I have restricted codecs installed.

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  • Using Audio Queue Services to play PCM data over a socket connection

    - by Rohan
    I'm writing a remote desktop client for the iPhone and I'm trying to implement audio redirection. The client is connected to the server over a socket connection, and the server sends 32K chunks of PCM data at a time. I'm trying to use AQS to play the data and it plays the first two seconds (1 buffer worth). However, since the next chunk of data hasn't come in over the socket yet, the next AudioQueueBuffer is empty. When the data comes in, I fill the next available buffer with the data and enqueue it with AudioQueueEnqueueBuffer. However, it never plays these buffers. Does the queue stop playing if there are no buffers in the queue, even if you later add a buffer? Here's the relevant part of the code: void wave_out_write(STREAM s, uint16 tick, uint8 index) { if(items_in_queue == NUM_BUFFERS){ return; } if(!playState.busy){ OSStatus status; status = AudioQueueNewOutput(&playState.dataFormat, AudioOutputCallback, &playState, CFRunLoopGetCurrent(), NULL, 0, &playState.queue); if(status == 0){ for(int i=0; i<NUM_BUFFERS; i++){ AudioQueueAllocateBuffer(playState.queue, 40000, &playState.buffers[i]); } AudioQueueAddPropertyListener(playState.queue, kAudioQueueProperty_IsRunning, MyAudioQueuePropertyListenerProc, &playState); status = AudioQueueStart(playState.queue, NULL); if(status ==0){ playState.busy = True; } else{ return; } } else{ return; } } playState.buffers[queue_hi]->mAudioDataByteSize = s->size; memcpy(playState.buffers[queue_hi]->mAudioData, s->data, s->size); AudioQueueEnqueueBuffer(playState.queue, playState.buffers[queue_hi], 0, 0); queue_hi++; queue_hi = queue_hi % NUM_BUFFERS; items_in_queue++; } void AudioOutputCallback(void* inUserData, AudioQueueRef outAQ, AudioQueueBufferRef outBuffer) { PlayState *playState = (PlayState *)inUserData; items_in_queue--; } Thanks!

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  • Play and record streaming audio

    - by Igor
    I'm working on an iPhone app that should be able to play and record audio streaming data simultaneously. Is it actually possible? I'm trying to mix SpeakHere and AudioRecorder samples and getting an empty file with no audio data... Here is my .m code: import "AzRadioViewController.h" @implementation azRadioViewController static const CFOptionFlags kNetworkEvents = kCFStreamEventOpenCompleted | kCFStreamEventHasBytesAvailable | kCFStreamEventEndEncountered | kCFStreamEventErrorOccurred; void MyAudioQueueOutputCallback( void* inClientData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer, const AudioTimeStamp inStartTime, UInt32 inNumberPacketDescriptions, const AudioStreamPacketDescription inPacketDesc ) { NSLog(@"start MyAudioQueueOutputCallback"); MyData* myData = (MyData*)inClientData; NSLog(@"--- %i", inNumberPacketDescriptions); if(inNumberPacketDescriptions == 0 && myData-dataFormat.mBytesPerPacket != 0) { inNumberPacketDescriptions = inBuffer-mAudioDataByteSize / myData-dataFormat.mBytesPerPacket; } OSStatus status = AudioFileWritePackets(myData-audioFile, FALSE, inBuffer-mAudioDataByteSize, inPacketDesc, myData-currentPacket, &inNumberPacketDescriptions, inBuffer-mAudioData); if(status == 0) { myData-currentPacket += inNumberPacketDescriptions; } NSLog(@"status:%i curpac:%i pcdesct: %i", status, myData-currentPacket, inNumberPacketDescriptions); unsigned int bufIndex = MyFindQueueBuffer(myData, inBuffer); pthread_mutex_lock(&myData-mutex); myData-inuse[bufIndex] = false; pthread_cond_signal(&myData-cond); pthread_mutex_unlock(&myData-mutex); } OSStatus StartQueueIfNeeded(MyData* myData) { NSLog(@"start StartQueueIfNeeded"); OSStatus err = noErr; if (!myData-started) { err = AudioQueueStart(myData-queue, NULL); if (err) { PRINTERROR("AudioQueueStart"); myData-failed = true; return err; } myData-started = true; printf("started\n"); } return err; } OSStatus MyEnqueueBuffer(MyData* myData) { NSLog(@"start MyEnqueueBuffer"); OSStatus err = noErr; myData-inuse[myData-fillBufferIndex] = true; AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; fillBuf-mAudioDataByteSize = myData-bytesFilled; err = AudioQueueEnqueueBuffer(myData-queue, fillBuf, myData-packetsFilled, myData-packetDescs); if (err) { PRINTERROR("AudioQueueEnqueueBuffer"); myData-failed = true; return err; } StartQueueIfNeeded(myData); return err; } void WaitForFreeBuffer(MyData* myData) { NSLog(@"start WaitForFreeBuffer"); if (++myData-fillBufferIndex = kNumAQBufs) myData-fillBufferIndex = 0; myData-bytesFilled = 0; myData-packetsFilled = 0; printf("-lock\n"); pthread_mutex_lock(&myData-mutex); while (myData-inuse[myData-fillBufferIndex]) { printf("... WAITING ...\n"); pthread_cond_wait(&myData-cond, &myData-mutex); } pthread_mutex_unlock(&myData-mutex); printf("<-unlock\n"); } int MyFindQueueBuffer(MyData* myData, AudioQueueBufferRef inBuffer) { NSLog(@"start MyFindQueueBuffer"); for (unsigned int i = 0; i < kNumAQBufs; ++i) { if (inBuffer == myData-audioQueueBuffer[i]) return i; } return -1; } void MyAudioQueueIsRunningCallback( void* inClientData, AudioQueueRef inAQ, AudioQueuePropertyID inID) { NSLog(@"start MyAudioQueueIsRunningCallback"); MyData* myData = (MyData*)inClientData; UInt32 running; UInt32 size; OSStatus err = AudioQueueGetProperty(inAQ, kAudioQueueProperty_IsRunning, &running, &size); if (err) { PRINTERROR("get kAudioQueueProperty_IsRunning"); return; } if (!running) { pthread_mutex_lock(&myData-mutex); pthread_cond_signal(&myData-done); pthread_mutex_unlock(&myData-mutex); } } void MyPropertyListenerProc( void * inClientData, AudioFileStreamID inAudioFileStream, AudioFileStreamPropertyID inPropertyID, UInt32 * ioFlags) { NSLog(@"start MyPropertyListenerProc"); MyData* myData = (MyData*)inClientData; OSStatus err = noErr; printf("found property '%c%c%c%c'\n", (inPropertyID24)&255, (inPropertyID16)&255, (inPropertyID8)&255, inPropertyID&255); switch (inPropertyID) { case kAudioFileStreamProperty_ReadyToProducePackets : { AudioStreamBasicDescription asbd; UInt32 asbdSize = sizeof(asbd); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataFormat, &asbdSize, &asbd); if (err) { PRINTERROR("get kAudioFileStreamProperty_DataFormat"); myData-failed = true; break; } err = AudioQueueNewOutput(&asbd, MyAudioQueueOutputCallback, myData, NULL, NULL, 0, &myData-queue); if (err) { PRINTERROR("AudioQueueNewOutput"); myData-failed = true; break; } for (unsigned int i = 0; i < kNumAQBufs; ++i) { err = AudioQueueAllocateBuffer(myData-queue, kAQBufSize, &myData-audioQueueBuffer[i]); if (err) { PRINTERROR("AudioQueueAllocateBuffer"); myData-failed = true; break; } } UInt32 cookieSize; Boolean writable; err = AudioFileStreamGetPropertyInfo(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, &writable); if (err) { PRINTERROR("info kAudioFileStreamProperty_MagicCookieData"); break; } printf("cookieSize %d\n", cookieSize); void* cookieData = calloc(1, cookieSize); err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_MagicCookieData, &cookieSize, cookieData); if (err) { PRINTERROR("get kAudioFileStreamProperty_MagicCookieData"); free(cookieData); break; } err = AudioQueueSetProperty(myData-queue, kAudioQueueProperty_MagicCookie, cookieData, cookieSize); free(cookieData); if (err) { PRINTERROR("set kAudioQueueProperty_MagicCookie"); break; } err = AudioQueueAddPropertyListener(myData-queue, kAudioQueueProperty_IsRunning, MyAudioQueueIsRunningCallback, myData); if (err) { PRINTERROR("AudioQueueAddPropertyListener"); myData-failed = true; break; } break; } } } static void ReadStreamClientCallBack(CFReadStreamRef stream, CFStreamEventType type, void *clientCallBackInfo) { NSLog(@"start ReadStreamClientCallBack"); if(type == kCFStreamEventHasBytesAvailable) { UInt8 buffer[2048]; CFIndex bytesRead = CFReadStreamRead(stream, buffer, sizeof(buffer)); if (bytesRead < 0) { } else if (bytesRead) { OSStatus err = AudioFileStreamParseBytes(globalMyData-audioFileStream, bytesRead, buffer, 0); if (err) { PRINTERROR("AudioFileStreamParseBytes"); } } } } void MyPacketsProc(void * inClientData, UInt32 inNumberBytes, UInt32 inNumberPackets, const void * inInputData, AudioStreamPacketDescription inPacketDescriptions) { NSLog(@"start MyPacketsProc"); MyData myData = (MyData*)inClientData; printf("got data. bytes: %d packets: %d\n", inNumberBytes, inNumberPackets); for (int i = 0; i < inNumberPackets; ++i) { SInt64 packetOffset = inPacketDescriptions[i].mStartOffset; SInt64 packetSize = inPacketDescriptions[i].mDataByteSize; size_t bufSpaceRemaining = kAQBufSize - myData-bytesFilled; if (bufSpaceRemaining < packetSize) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } AudioQueueBufferRef fillBuf = myData-audioQueueBuffer[myData-fillBufferIndex]; memcpy((char*)fillBuf-mAudioData + myData-bytesFilled, (const char*)inInputData + packetOffset, packetSize); myData-packetDescs[myData-packetsFilled] = inPacketDescriptions[i]; myData-packetDescs[myData-packetsFilled].mStartOffset = myData-bytesFilled; myData-bytesFilled += packetSize; myData-packetsFilled += 1; size_t packetsDescsRemaining = kAQMaxPacketDescs - myData-packetsFilled; if (packetsDescsRemaining == 0) { MyEnqueueBuffer(myData); WaitForFreeBuffer(myData); } } } (IBAction)buttonPlayPressedid)sender { label.text = @"Buffering"; [self connectionStart]; } (IBAction)buttonSavePressedid)sender { NSLog(@"save"); AudioFileClose(myData.audioFile); AudioQueueDispose(myData.queue, TRUE); } bool getFilename(char* buffer,int maxBufferLength) { NSArray paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString docDir = [paths objectAtIndex:0]; NSString* file = [docDir stringByAppendingString:@"/rec.caf"]; return [file getCString:buffer maxLength:maxBufferLength encoding:NSUTF8StringEncoding]; } -(void)connectionStart { @try { MyData* myData = (MyData*)calloc(1, sizeof(MyData)); globalMyData = myData; pthread_mutex_init(&myData-mutex, NULL); pthread_cond_init(&myData-cond, NULL); pthread_cond_init(&myData-done, NULL); NSLog(@"Start"); myData-dataFormat.mSampleRate = 16000.0f; myData-dataFormat.mFormatID = kAudioFormatLinearPCM; myData-dataFormat.mFramesPerPacket = 1; myData-dataFormat.mChannelsPerFrame = 1; myData-dataFormat.mBytesPerFrame = 2; myData-dataFormat.mBytesPerPacket = 2; myData-dataFormat.mBitsPerChannel = 16; myData-dataFormat.mReserved = 0; myData-dataFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked; int i, bufferByteSize; UInt32 size; AudioQueueNewInput( &myData-dataFormat, MyAudioQueueOutputCallback, &myData, NULL /* run loop /, kCFRunLoopCommonModes / run loop mode /, 0 / flags */, &myData-queue); size = sizeof(&myData-dataFormat); AudioQueueGetProperty(&myData-queue, kAudioQueueProperty_StreamDescription, &myData-dataFormat, &size); CFURLRef fileURL; char path[256]; memset(path,0,sizeof(path)); getFilename(path,256); fileURL = CFURLCreateFromFileSystemRepresentation(NULL, (UInt8*)path, strlen(path), FALSE); AudioFileCreateWithURL(fileURL, kAudioFileCAFType, &myData-dataFormat, kAudioFileFlags_EraseFile, &myData-audioFile); OSStatus err = AudioFileStreamOpen(myData, MyPropertyListenerProc, MyPacketsProc, kAudioFileMP3Type, &myData-audioFileStream); if (err) { PRINTERROR("AudioFileStreamOpen"); return 1; } CFStreamClientContext ctxt = {0, self, NULL, NULL, NULL}; CFStringRef bodyData = CFSTR(""); // Usually used for POST data CFStringRef headerFieldName = CFSTR("X-My-Favorite-Field"); CFStringRef headerFieldValue = CFSTR("Dreams"); CFStringRef url = CFSTR(RADIO_LOCATION); CFURLRef myURL = CFURLCreateWithString(kCFAllocatorDefault, url, NULL); CFStringRef requestMethod = CFSTR("GET"); CFHTTPMessageRef myRequest = CFHTTPMessageCreateRequest(kCFAllocatorDefault, requestMethod, myURL, kCFHTTPVersion1_1); CFHTTPMessageSetBody(myRequest, bodyData); CFHTTPMessageSetHeaderFieldValue(myRequest, headerFieldName, headerFieldValue); CFReadStreamRef stream = CFReadStreamCreateForHTTPRequest(kCFAllocatorDefault, myRequest); if (!stream) { NSLog(@"Creating the stream failed"); return; } if (!CFReadStreamSetClient(stream, kNetworkEvents, ReadStreamClientCallBack, &ctxt)) { CFRelease(stream); NSLog(@"Setting the stream's client failed."); return; } CFReadStreamScheduleWithRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); if (!CFReadStreamOpen(stream)) { CFReadStreamSetClient(stream, 0, NULL, NULL); CFReadStreamUnscheduleFromRunLoop(stream, CFRunLoopGetCurrent(), kCFRunLoopCommonModes); CFRelease(stream); NSLog(@"Opening the stream failed."); return; } } @catch (NSException *exception) { NSLog(@"main: Caught %@: %@", [exception name], [exception reason]); } } (void)viewDidLoad { [[UIApplication sharedApplication] setIdleTimerDisabled:YES]; [super viewDidLoad]; } (void)didReceiveMemoryWarning { [super didReceiveMemoryWarning]; } (void)viewDidUnload { } (void)dealloc { [super dealloc]; } @end

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  • Stopping and Play button for Audio (Android)

    - by James Rattray
    I have this problem, I have some audio I wish to play... And I have two buttons for it, 'Play' and 'Stop'... Problem is, after I press the stop button, and then press the Play button, nothing happens. -The stop button stops the song, but I want the Play button to play the song again (from the start) Here is my code: final MediaPlayer mp = MediaPlayer.create(this, R.raw.megadeth); And then the two public onclicks: (For playing...) button.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { // Perform action on click button.setText("Playing!"); try { mp.prepare(); } catch (IllegalStateException e) { // TODO Auto-generated catch block e.printStackTrace(); } catch (IOException e) { // TODO Auto-generated catch block e.printStackTrace(); } mp.start(); // } }); And for stopping the track... final Button button2 = (Button) findViewById(R.id.cancel); button2.setOnClickListener(new View.OnClickListener() { public void onClick(View v) { mp.stop(); mp.reset(); } }); Can anyone see the problem with this? If so could you please fix it... (For suggest) Thanks alot... James

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  • how to solve error processing /usr/lib/python2.7/dist-packages/pygst.pth:?

    - by ChitKo
    Error processing line 1 of /usr/lib/python2.7/dist-packages/pygst.pth: Traceback (most recent call last): File "/usr/lib/python2.7/site.py", line 161, in addpackage if not dircase in known_paths and os.path.exists(dir): File "/usr/lib/python2.7/genericpath.py", line 18, in exists os.stat(path) TypeError: must be encoded string without NULL bytes, not str Remainder of file ignored Error processing line 1 of /usr/lib/python2.7/dist-packages/pygtk.pth: Traceback (most recent call last): File "/usr/lib/python2.7/site.py", line 161, in addpackage if not dircase in known_paths and os.path.exists(dir): File "/usr/lib/python2.7/genericpath.py", line 18, in exists os.stat(path) TypeError: must be encoded string without NULL bytes, not str Remainder of file ignored Traceback (most recent call last): File "/usr/share/apport/apport-gtk", line 16, in <module> from gi.repository import GObject File "/usr/lib/python2.7/dist-packages/gi/importer.py", line 76, in load_module dynamic_module._load() File "/usr/lib/python2.7/dist-packages/gi/module.py", line 222, in _load version) File "/usr/lib/python2.7/dist-packages/gi/module.py", line 90, in __init__ repository.require(namespace, version) gi.RepositoryError: Failed to load typelib file '/usr/lib/girepository-1.0/GLib-2.0.typelib' for namespace 'GLib': Invalid magic header

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  • Can I use Ubuntu as a wireless media server which performs all decoding/processing server-side?

    - by AthloX
    I want to setup UBUNTU 12.04 desktop as Home media server. I have window 7 netbook and UBUNTU 12.04lts laptop even a samsun galaxy note tablet (android). Two desktop in other room with dualboot win7 and ubuntu. SHARP AQUOS Plasma Tv with Wi-Fi connected. I want to install ubuntu as media server to stream audio/video files over wi-fi. Not only this i want this media server to use its own processing power to decode ans stream so that on remote end only file can play without using their own resource. Is it possible to use ubuntu as media server to stream files without making the remote end to use there own resource. I want only bandwidth of Wi-Fi to be use in this and media center hardware resource.Remote end gadget should use only speaker and screen and not processing power of their own. Please any suggestion is it possible to do so ?

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  • image filters for iphone sdk development

    - by plsp
    Hi All, I am planning to develop an iphone app which makes use of image filters like blurring, sharpening,etc. I noticed that there are few approaches for this one, Use openGL ES. I even found an example code on apple iphone dev site. How easy is openGL for somebody who has never used it? Can the image filters be implemented using the openGL framework? There is a Quartz demo as well posted on apple iphone dev site. Has anybody used this framework for doing image processing? How is this approach compared to openGL framework? Don't use openGL and Quartz framework. Basically access the raw pixels from the image and do the manipulation myself. Make use of any custom built image processing libraries like this one. Do you know of any other libraries like this one? Can anybody provide insights/suggestions on which option is the best? Your opinions are highly appreciated. Thanks!

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  • GPGPU programming with OpenGL ES 2.0

    - by Albus Dumbledore
    I am trying to do some image processing on the GPU, e.g. median, blur, brightness, etc. The general idea is to do something like this framework from GPU Gems 1. I am able to write the GLSL fragment shader for processing the pixels as I've been trying out different things in an effect designer app. I am not sure however how I should do the other part of the task. That is, I'd like to be working on the image in image coords and then outputting the result to a texture. I am aware of the gl_FragCoords variable. As far as I understand it it goes like that: I need to set up a view (an orthographic one maybe?) and a quad in such a way so that the pixel shader would be applied once to each pixel in the image and so that it would be rendering to a texture or something. But how can I achieve that considering there's depth that may make things somewhat awkward to me... I'd be very grateful if anyone could help me with this rather simple task as I am really frustrated with myself. UPDATE It seems I'll have to use an FBO, getting one like this: glBindFramebuffer(...)

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  • WPF Storyboard delay in playing wma files

    - by Rita
    I'm a complete beginner in WPF and have an app that uses StoryBoard to play a sound. public void PlaySound() { MediaElement m = (MediaElement)audio.FindName("MySound.wma"); m.IsMuted = false; FrameworkElement audioKey = (FrameworkElement)keys.FindName("MySound"); Storyboard s = (Storyboard)audioKey.FindResource("MySound.wma"); s.Begin(audioKey); } <Storyboard x:Key="MySound.wma"> <MediaTimeline d:DesignTimeNaturalDuration="1.615" BeginTime="00:00:00" Storyboard.TargetName="MySound.wma" Source="Audio\MySound.wma"/> </Storyboard> I have a horrible lag and sometimes it takes good 10 seconds for the sound to be played. I suspect this has something to do with the fact that no matter how long I wait - The sound doesn't get played until after I leave the function. I don't understand it. I call Begin, and nothing happens. Is there a way to replace this method, or StoryBoard object with something that plays instantly and without a lag?

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  • SSAS Cube reprocessing fails - then succeeds if I try again

    - by EdgarVerona
    So I'm basically brand new to the concept of BI, and I've inherited an existing ETL process that is a two step process: 1) Loads the data into a database that is only used by the cube processing 2) Starts off the SSAS cube processing against said database It seems pretty well isolated, but occasionally (once a week, sometimes twice) it will fail with the following exception: "Errors in the OLAP storage engine: The attribute key cannot be found" Now the interesting thing is that: 1) The dimension having the issue is not usually the same one (i.e. there's no single dimension that consistently has this failure) 2) The source table, when I inspect it, does actually contain the attribute key that it says could not be found And, most interestingly... 3) If I then immediately reprocess the dimensions and cubes manually through SSMS, they reprocess successfully and without incident. In both the aforementioned job and when I reprocess them through SSMS, I am using "ProcessFull", so it should be reprocessing them completely. Has anyone run into such an issue? I'm scratching my head about it... because if it was a genuine data integrity issue, reprocessing the cube again wouldn't fix it. What on earth could be happening? I've been tasked with finding out why this happens, but I can neither reproduce it consistently nor can I point to a data integrity problem as the root cause. Thanks for any input you can provide!

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  • What do you use to play sound in iPhone games?

    - by zoul
    Hello! I have a performance-intensive iPhone game I would like to add sounds to. There seem to be about three main choices: (1) AVAudioPlayer, (2) Audio Queues and (3) OpenAL. I’d hate to write pages of low-level code just to play a sample, so that I would like to use AVAudioPlayer. The problem is that it seems to kill the performace – I’ve done a simple measuring using CFAbsoluteTimeGetCurrent and the play message seems to take somewhere from 9 to 30 ms to finish. That’s quite miserable, considering that 25 ms == 40 fps. Of course there is the prepareToPlay method that should speed things up. That’s why I wrote a simple class that keeps several AVAudioPlayers at its disposal, prepares them beforehand and then plays the sample using the prepared player. No cigar, still it takes the ~20 ms I mentioned above. Such performance is unusable for games, so what do you use to play sounds with a decent performance on iPhone? Am I doing something wrong with the AVAudioPlayer? Do you play sounds with Audio Queues? (I’ve written something akin to AVAudioPlayer before 2.2 came out and I would love to spare that experience.) Do you use OpenAL? If yes, is there a simple way to play sounds with OpenAL, or do you have to write pages of code? Update: Yes, playing sounds with OpenAL is fairly simple.

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  • How to pass dynamic parameters to .pde file

    - by Kalpana
    class Shape contains two methods drawCircle() and drawTriangle(). Each function takes different set of arguments. At present, I invoke this by calling the pde file directly. How to pass these arguments from a HTML file directly if I have to control the arguments being passed to the draw function? 1) Example.html has (current version) <script src="processing-1.0.0.min.js"></script> <canvas data-processing-sources="example.pde"></canvas> 2) Example.pde has class Shape { void drawCircle(intx, int y, int radius) { ellipse(x, y, radius, radius); } void drawTriangle(int x1, int y1, int x2, int y2, int x3, int y3) { rect(x1, y1, x2, y2, x3, y3); } } Shape shape = new Shape(); shape.drawCircle(10, 40, 70); I am looking to do something like this in my HTML file, so that I can move all the functions into a separate file and call them with different arguments to draw different shapes (much similar to how you would do it in Java) A.html <script> Shape shape = new Shape(); shape.drawCircle(10, 10, 3); </script> B.html <script> Shape shape = new Shape(); shape.drawTriangle(30, 75, 58, 20, 86, 75); </script> 2) Iam using Example2.pde has void setup() { size(200,200); background(125); fill(255); } void rectangle(int x1, int y1, int x2, int y2) { rect(x1, y1, x2, y2); } My Example2.html has var processingInstance; processingInstance.rectangle(30, 20, 55, 55); but this is not working. How to pass these parameters dynamically from html.

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  • JMF microphone volume controller

    - by TacB0sS
    How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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  • Graphing the pitch (frequency) of a sound

    - by Coronatus
    I want to plot the pitch of a sound into a graph. Currently I can plot the amplitude. The graph below is created by the data returned by getUnscaledAmplitude(): AudioInputStream audioInputStream = AudioSystem.getAudioInputStream(new BufferedInputStream(new FileInputStream(file))); byte[] bytes = new byte[(int) (audioInputStream.getFrameLength()) * (audioInputStream.getFormat().getFrameSize())]; audioInputStream.read(bytes); // Get amplitude values for each audio channel in an array. graphData = type.getUnscaledAmplitude(bytes, this); public int[][] getUnscaledAmplitude(byte[] eightBitByteArray, AudioInfo audioInfo) { int[][] toReturn = new int[audioInfo.getNumberOfChannels()][eightBitByteArray.length / (2 * audioInfo. getNumberOfChannels())]; int index = 0; for (int audioByte = 0; audioByte < eightBitByteArray.length;) { for (int channel = 0; channel < audioInfo.getNumberOfChannels(); channel++) { // Do the byte to sample conversion. int low = (int) eightBitByteArray[audioByte]; audioByte++; int high = (int) eightBitByteArray[audioByte]; audioByte++; int sample = (high << 8) + (low & 0x00ff); if (sample < audioInfo.sampleMin) { audioInfo.sampleMin = sample; } else if (sample > audioInfo.sampleMax) { audioInfo.sampleMax = sample; } toReturn[channel][index] = sample; } index++; } return toReturn; } But I need to show the audio's pitch, not amplitude. Fast Fourier transform appears to get the pitch, but it needs to know more variables than the raw bytes I have, and is very complex and mathematical. Is there a way I can do this?

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  • Projective transformation

    - by mcwehner
    Given two image buffers (assume it's an array of ints of size width * height, with each element a color value), how can I map an area defined by a quadrilateral from one image buffer into the other (always square) image buffer? I'm led to understand this is called "projective transformation". I'm also looking for a general (not language- or library-specific) way of doing this, such that it could be reasonably applied in any language without relying on "magic function X that does all the work for me". An example: I've written a short program in Java using the Processing library (processing.org) that captures video from a camera. During an initial "calibrating" step, the captured video is output directly into a window. The user then clicks on four points to define an area of the video that will be transformed, then mapped into the square window during subsequent operation of the program. If the user were to click on the four points defining the corners of a door visible at an angle in the camera's output, then this transformation would cause the subsequent video to map the transformed image of the door to the entire area of the window, albeit somewhat distorted.

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  • Python on Mac: Fink? MacPorts? Builtin? Homebrew? Binary installer?

    - by BastiBechtold
    For the last few days, I have been trying to use Python for some audio development. The thing is, Mac OSX does not handle uninstalling stuff well. Actually, there is no way to uninstall anything. Once it is on your system, you better pray that it didn't do any funny stuff. Hence, I don't really want to rely on installer packages for Python. So I turn to Homebrew and install Python using Homebrew. Works fabulously. Using pip, Numpy, SciPy, Matplotlib were no (big) problem, either. Now I want to play audio. There is a host of different packages out there, but pip does not seem willing to install any. But, there is a binary distribution for PyGame, which I guess should work with the built-in Python. Hence my question: What would you do? Would you just install the binary distributions and hope that they interoperate well and never need uninstalling? Would you hack your way through whichever package control management system you prefer and deal with its problems? Something else?

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  • Syncing two AS3 NetStreams

    - by Lowgain
    I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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  • Which is the best way to encode batch videos on server side?

    - by albanx
    Hello I am making a general question since I am a developer and I have no advance experience on video elaboration. I have to preparare a web application with the purpose to allow video files upload on our company server and then video elaboration by server, on user command. The purpose of the web application is to allow to the user to make some elaboration on video depending on user action launch from the web app: (server has to ) convert video in different format(mp4, flv...) extact keyframes from video and saves them in jpeg format possibility to extract audio from video automatic control of quality audio & video (black frames,silences detection) change scene detection and keyframe extraction ..... This what's my bosses wanted from the web based application (with the server support obviously), and I understand only the first 3 points of this list, the rest for me was arabic.... My question is: Which is the best and fastest server side application for this works, that can support multiple batch video conversions, from command line (comand line for php-soap-socket interaction or something else..)? Is suitable Adobe Media Server for batch video conversion? Which are adobe products that can be used for this purpose? Note: I have experience with Indesign Server scripting programing (sending xml with php and soap call...), and I am looking to something similiar for video elaboration. I will appreciate any answers. THANKS ALL

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