Search Results

Search found 6479 results on 260 pages for 'audio analysis'.

Page 65/260 | < Previous Page | 61 62 63 64 65 66 67 68 69 70 71 72  | Next Page >

  • SDK 3.2 - Trigger video from UIScrollView Subview (Audio but no video)

    - by stalure
    I am having a hard time getting video to play in an application that I am working on and I think the answer has to do with the views and view controllers. I have the flow depicted below. When the button is clicked, the audio from the video is playing, but nothing is displayed on screen. Anyone have any ideas? -(IBAction) playMovie { Play Movie Code } ViewController UIScrollView UIView UIButton - (playMovie)

    Read the article

  • Writing module to get video/audio from HDMI

    - by Martin
    I would like to write a small module that can check if anything is connected to my computer's HDMI input, and if so write a frame of video to bitmap once in a while. Can anyone point me to resources regarding grabbing audio/video from HDMI on windows?

    Read the article

  • How to change the audio output device in Firefox or any other modern browser?

    - by Zanami Zani
    I'm trying to play music through Ventrilo and currently I use Virtual Audio Cable. The way it works is that in foobar2000 (a music playing program) I set the output device in preferences to Virtual Audio Cable. Then in Ventrilo I log in to another name and set the input device to Virtual Audio Cable. This routes the music through the Virtual Audio Cable and allows me to play the music through Ventrilo. However, I would also like to change the output device for Firefox (or any other browser) or "Plugin Container for Firefix" to Virtual Audio Cable so that I could play music from Pandora or YouTube on to Ventrilo. Unfortunately I could not find an option for this anywhere.

    Read the article

  • Is Windows Media Player able to play DTS audio?

    - by rolgae
    I'm trying to play DTS audio with Windows Media Player 12 on Windows 7. For a MPEG-TS file with video and DTS audio, only video is played. A file containing only a DTS audio stream is rejected. But: WMP is able to play the DTS audio stream of a DVD. So, Is Windows Media Player able to play DTS audio, or not? And if: How do I make him play my DTS files? I did not find any good resources of the supported codecs. Just things like "WMP can play .mpg files, ..." VLC is able to play all of the above files. I do not want to install third party codec packs, thats not the question!

    Read the article

  • Any screen capture software that captures webcam, microphone inputs too ?

    - by mohanr
    I am going to conduct a user study. Apart from capturing the screen while the user is interacting with the system, I also want to capture the video/audio of the user. Is there any software that in addition to capturing the screen also overlays it with the webcam/microphone inputs. The goal is to capture the complete experience of the user: key/mouse interactions with the system along with their facial/vocal responses. I know that I can maybe run a screen-capture software and also run a software for capturing webcam audio/video alongside and try to sync/overlay both these streams with timestamps. But I am going to be dealing with probably several hundred hours of data. So I am looking for a tool that can streamline the process for me amap and help me keep my sanity at end of the process. Thanks,

    Read the article

  • Silverlight 4 - encoding PCM data from the microphone

    - by Richard
    Hi I've written a basic SL4 application to capture audio data from the microphone using CaptureSource. The trouble is, it's raw PCM output - which means huge and uncompressed. Given that I need this application to run purely within a SL4 environment, how can I compress the PCM audio data into something that can be delivered to a remote server more easily? In conversation, people have suggested Speex and WMA for instance, but I haven't found any libraries or examples that work without requiring reference to DLL's that won't work in a SL4 project. Thanks, Richard.

    Read the article

  • After playing a MediaElement, how can I play it again?

    - by Edward Tanguay
    I have a variable MediaElement variable named TestAudio in my Silverlight app. When I click the button, it plays the audio correctly. But when I click the button again, it does not play the audio. How can I make the MediaElement play a second time? None of the tries below to put position back to 0 worked: private void Button_Click_PlayTest(object sender, RoutedEventArgs e) { //TestAudio.Position = new TimeSpan(0, 0, 0); //TestAudio.Position = TestAudio.Position.Add(new TimeSpan(0, 0, 0)); //TestAudio.Position = new TimeSpan(0, 0, 0, 0, 0); //TestAudio.Position = TimeSpan.Zero; TestAudio.Play(); }

    Read the article

  • 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?

    Read the article

  • Changing volume in Java when using JLayer.

    - by Penchant
    I'm using JLayer to play an inputstream of mp3 data from the internet. How do i change the volume of the output? I'm using this code to play it: URL u = new URL(s); URLConnection conn = u.openConnection(); conn.setConnectTimeout(Searcher.timeoutms); conn.setReadTimeout(Searcher.timeoutms); bitstream = new Bitstream(conn.getInputStream()/*new FileInputStream(quick_file)*/); System.out.println(bitstream); decoder = new Decoder(); decoder.setEqualizer(equalizer); audio = FactoryRegistry.systemRegistry().createAudioDevice(); audio.open(decoder); for(int i = quick_positions[0]; i > 0; i--){ Header h = bitstream.readFrame(); if (h == null){ return; } bitstream.closeFrame();

    Read the article

  • Play a beep that loop and change the frequency/speed

    - by Bono
    Hi all, I am creating an iphone application that use audio. I want to play a beep sound that loop indefinitely. I found an easy way to do that using the upper layer AVAudioPlayer and the numberOfLoops set to "-1". It works fine. But now I want to play this audio and be able to change the rate / speed. It may works like the sound played by a car when approaching an obstacle. At the beginning the beep has a low frequency and this frequency accelerate till reaching a continuous sound biiiiiiiiiiiip ... It seems this is not feasible using the high layer AVAudioPlayer, but even looking at AudioToolBox I found no solution. Does anybody have informations about how to do that? Thanks a lot for helping me!

    Read the article

  • 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]); }];

    Read the article

  • How Do I Convert text to a WAV file With Inaudible Waveform?

    - by Scott
    I am trying to create an audio watermarking system. I figure the best solution is to create an audio file (WAV) based on a unique string of text and then combine this with the original wav. The part that makes this tricky (for me anyway) is: How do I convert the text string to a wav? How do I ensure that the resulting WAV form is inaudible (or at least barely noticeable to the listener). I would prefer this be done server side (via PHP, etc) but if the processing load isn't too much then would be ok with something in Flash or Javascript. I'd be willing to pay someone to create me a workable solution (complete source code that functions as described). Thanks, Scott!

    Read the article

  • How to configure the framesize using AudioUnit.framework on iOS

    - by Piperoman
    I have an audio app i need to capture mic samples to encode into mp3 with ffmpeg First configure the audio: /** * We need to specifie our format on which we want to work. * We use Linear PCM cause its uncompressed and we work on raw data. * for more informations check. * * We want 16 bits, 2 bytes (short bytes) per packet/frames at 8khz */ AudioStreamBasicDescription audioFormat; audioFormat.mSampleRate = SAMPLE_RATE; audioFormat.mFormatID = kAudioFormatLinearPCM; audioFormat.mFormatFlags = kAudioFormatFlagIsPacked | kAudioFormatFlagIsSignedInteger; audioFormat.mFramesPerPacket = 1; audioFormat.mChannelsPerFrame = 1; audioFormat.mBitsPerChannel = audioFormat.mChannelsPerFrame*sizeof(SInt16)*8; audioFormat.mBytesPerPacket = audioFormat.mChannelsPerFrame*sizeof(SInt16); audioFormat.mBytesPerFrame = audioFormat.mChannelsPerFrame*sizeof(SInt16); The recording callback is: static OSStatus recordingCallback(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData) { NSLog(@"Log record: %lu", inBusNumber); NSLog(@"Log record: %lu", inNumberFrames); NSLog(@"Log record: %lu", (UInt32)inTimeStamp); // the data gets rendered here AudioBuffer buffer; // a variable where we check the status OSStatus status; /** This is the reference to the object who owns the callback. */ AudioProcessor *audioProcessor = (__bridge AudioProcessor*) inRefCon; /** on this point we define the number of channels, which is mono for the iphone. the number of frames is usally 512 or 1024. */ buffer.mDataByteSize = inNumberFrames * sizeof(SInt16); // sample size buffer.mNumberChannels = 1; // one channel buffer.mData = malloc( inNumberFrames * sizeof(SInt16) ); // buffer size // we put our buffer into a bufferlist array for rendering AudioBufferList bufferList; bufferList.mNumberBuffers = 1; bufferList.mBuffers[0] = buffer; // render input and check for error status = AudioUnitRender([audioProcessor audioUnit], ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, &bufferList); [audioProcessor hasError:status:__FILE__:__LINE__]; // process the bufferlist in the audio processor [audioProcessor processBuffer:&bufferList]; // clean up the buffer free(bufferList.mBuffers[0].mData); //NSLog(@"RECORD"); return noErr; } With data: inBusNumber = 1 inNumberFrames = 1024 inTimeStamp = 80444304 // All the time same inTimeStamp, this is strange However, the framesize that i need to encode mp3 is 1152. How can i configure it? If i do buffering, that implies a delay, but i would like to avoid this because is a real time app. If i use this configuration, each buffer i get trash trailing samples, 1152 - 1024 = 128 bad samples. All samples are SInt16.

    Read the article

  • Signal amplitude against time in Java

    - by wsr74ws84
    I'm racking my brain in order to solve a knotty problem (at least for me). While playing an audio file (using Java) I want the signal amplitude to be displayed against time. I mean I'd like to implement a small panel showing a sort of oscilloscope (spectrum analyzer). The audio signal should be viewed in the time domain (vertical axis is amplitude and the horizontal axis is time). Does anyone know how to do it? Is there a good tutorial I can rely on? Since I know very little about Java, I hope someone can help me.

    Read the article

< Previous Page | 61 62 63 64 65 66 67 68 69 70 71 72  | Next Page >