Search Results

Search found 4461 results on 179 pages for 'pic audio'.

Page 63/179 | < Previous Page | 59 60 61 62 63 64 65 66 67 68 69 70  | Next Page >

  • Pitch detection and change java

    - by omegas27
    Hello, I'm french so I'm sorry if you have trouble to understand some of my sentences. Aniways, I saw in some topics that the pitch could be fetected thanks to the Fourier transform but I didn't really understand how to implement it. Moreover, I didn't find how to change the pitch of a wav file and if possibl ,a mp3 file I am listening to music using javaSound for the wav and JLayer for the mp3. Thanks

    Read the article

  • Playing Multiple sounds at the same time in Android

    - by Wrapper
    I am unable to use the following to code to play multiple sounds/beeps simultaneously. In my onclicklistener I have added ... public void onClick(View v) { mSoundManager.playSound(1); mSoundManager.playSound(2); } ... But this plays only one sound at a time, sound with index 1 followed by sound with index 2. How can I play atleast 2 sounds simultaneously using this code whenever there is an onClick() event? public class SoundManager { private SoundPool mSoundPool; private HashMap<Integer, Integer> mSoundPoolMap; private AudioManager mAudioManager; private Context mContext; public SoundManager() { } public void initSounds(Context theContext) { mContext = theContext; mSoundPool = new SoundPool(4, AudioManager.STREAM_MUSIC, 0); mSoundPoolMap = new HashMap<Integer, Integer>(); mAudioManager = (AudioManager)mContext.getSystemService(Context.AUDIO_SERVICE); } public void addSound(int Index,int SoundID) { mSoundPoolMap.put(1, mSoundPool.load(mContext, SoundID, 1)); } public void playSound(int index) { int streamVolume = mAudioManager.getStreamVolume(AudioManager.STREAM_MUSIC); mSoundPool.play(mSoundPoolMap.get(index), streamVolume, streamVolume, 1, 0, 1f); } public void playLoopedSound(int index) { int streamVolume = mAudioManager.getStreamVolume(AudioManager.STREAM_MUSIC); mSoundPool.play(mSoundPoolMap.get(index), streamVolume, streamVolume, 1, -1, 1f); } }

    Read the article

  • Getting following exception javax.sound.sampled.LineUnavailableException: line with format ULAW 800

    - by angelina
    Dear All, I tried to play and get duration of a wave file using code below but got following exception.please resolve.I m using a wave file format. URL url = new URL("foo.wav"); Clip clip = AudioSystem.getClip(); AudioInputStream ais = AudioSystem.getAudioInputStream(url); clip.open(ais); System.out.println(clip.getMicrosecondLength()); **javax.sound.sampled.LineUnavailableException: line with format ULAW 8000.0 Hz, 8 bit, mono, 1 bytes/frame, not supported.**

    Read the article

  • Android PCM Bytes

    - by Pintac
    Hi I am using the AudioRecord class to analize raw pcm bytes as it comes in the mic. So thats working nicely. Now i need convert the pcm bytes into decibel. I have a formula that takes sound presure in Pa into db. db = 20 * log10(Pa/ref Pa) So the question is the bytes i am getting from audiorecorder from the buffer what is it is it amplitude pascal sound pressure or what. I tried to putting the value into te formula but it comes back with very hight db so i do not think its right thanks

    Read the article

  • AVAudioPlayer currentTime problem

    - by StrAbZ
    Hi, I'm trying to use the audioplayer with a slider in order to seek into a track (nothing complicated). But I have a weird behavior... for some value of currentTime (between 0 and trackDuration), the player stop playing the track, and goes into "audioPlayerDidFinishPlaying:successfully:" with successfully to NO. And it did not go into "audioPlayerDecodeErrorDidOccur:error:" It's like it can't read the time i'm giving to it. For exemple the duration of the track is: 295.784424 seconds i set the currentTime to 55.0s (ie: 54.963878 or 54.963900 or 54.987755, etc... when printed as %f). The "crashes" always happen when the currentTime is 54.987755... and I really don't understand why... So if you have any idea... ^^

    Read the article

  • How can I get latency info from Android's AudioTrack class?

    - by Ryan
    I've noticed that the C++ classes underlying the AudioTrack and AudioRecord APIs in Android both have a latency() method that is not exposed via JNI. As far as I can see, the latency() method in AudioRecord still does not take into account the hardware latency (they have a TODO comment for that), but the latency() method in AudioTrack does add in the hardware latency. I absolutely need to get this latency value from AudioTrack. Is there any possible way I can do this? I don't care what kind of crazy hack is needed as long as it doesn't require a rooted phone (the resulting code must still be packaged as an app on the market).

    Read the article

  • 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.

    Read the article

  • SoundPool repeating issue for Samsung Galaxy S3

    - by Alaa Eldin
    I'm trying to play a background sound for my application, I use SoundPool class, my problem is that, sound plays well only when I set the loop parameter with zero value, but it doesn't work for any other value. My code for initialization is: soundpool = new SoundPool(4, AudioManager.STREAM_MUSIC, 0); soundsMap = new HashMap<Integer, Integer>(); soundsMap.put(1, soundpool.load(this, R.raw.soundfile_1, 1)); soundsMap.put(2, soundpool.load(this, R.raw.soundfile_2, 1)); my code for playing is soundpool.play(1, 0.9f, 0.9f, 1, -1, 1f); as I mentioned sound works when I put (0) instead of (-1) for the loop value, anyone has any idea why (-1) or any value other than (0) doesn't work (there is no output sound) ?

    Read the article

  • Open Source sound engine

    - by Steph Thirion
    When I started using SoundEngine (from CrashLanding and TouchFighter), I had read about a few people recommending not to use it, for it was, according to them, not stable enough. Still it was the only solution I knew of to play sounds with pitch and position control without learning C++ and OpenAL, so I ignored the warnings and went on with it. But now I'm starting to worry. The 2.2 SDK introduced AVFoundation. Using both SoundEngine from CrashLanding (for sounds) and AVAudioPlayer (for music), I found out SoundEngine behaves strangely when the only existing AVAudioPlayer is released (all sounds stop until a new AVAudioPlayer is initiated). Around the same time as the 2.2 SDK came out, the CrashLanding sample code was mysteriously removed from the ADC site. I'm worried there are more bad surprises to come. My question is, is anyone aware of an Open Source alternative to SoundEngine? Maybe even a C++ library that uses OpenAL?

    Read the article

  • Is it possible to detect when the system is recording a sound and then perform some action on Python

    - by Jorge
    I began learning Python a few days ago, and i was wondering about a practical use for a program. Then i came up with the following: if my brother is in his room recording himself playing guitar, a led plugged to the usb and wired so it's outside his door lights up, and then i'll know he's recording and i'll take care not to make any noises. The main questions are: How Python can detect any recording going on in the system? How would i interface with the usb so i can actually turn the led on?

    Read the article

  • A way to enable a LaunchDaemon to output sound?

    - by Varun Mehta
    I have a small Foundation application that checks a website and plays a sound if it sees a certain value. This application successfully plays a sound when I run it as my user from the Terminal. I've configured this app to run as a LaunchDaemon, with the following plist: <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> <plist version="1.0"> <dict> <key>Label</key> <string>org.myorg.appidentifier</string> <key>ProgramArguments</key> <array> <string>/Users/varunm/path/to/cli/application</string> </array> <key>KeepAlive</key> <true/> <key>RunAtLoad</key> <true/> </dict> </plist> When I have this service launched I can see it successfully read in and log values from the website, but it never generates any sound. The sound files are located in the same directory as the binary, and I use the following code: NSSound *soundToPlay = [[NSSound alloc] initWithContentsOfFile:@"sound.wav" byReference:NO]; [soundToPlay setDelegate:stopper]; [soundToPlay play]; while (g_keepRunning) { [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1.0]]; } [soundToPlay setCurrentTime:0.0]; Is there any way to get my LaunchDaemon application to play sound? This machine gets run by different people, and sometimes has no one logged in, which is why I have to configure it as a LaunchDaemon.

    Read the article

  • How to record sound from a microphone in VB6?

    - by Clay Nichols
    We've been recording sound for over a decade using what seems like a very clunky method using the Winmm.dll and the MCIsendString. I've read that this doesn't set the recording quality value correctly (not sure if that article was ever true or is still true). I was wondering if there is any better way to record sound.

    Read the article

  • can the python wave module accept StringIO object

    - by user368005
    i'm trying to use the wave module to read wav files in python. whats not typical of my applications is that I'm NOT using a file or a filename to read the wav file, but instead i have the wav file in a buffer. And here's what i'm doing import StringIO buffer = StringIO.StringIO() buffer.output(wav_buffer) file = wave.open(buffer, 'r') but i'm getting a EOFError when i run it... File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 493, in open return Wave_read(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 163, in __init__ self.initfp(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 128, in initfp self._file = Chunk(file, bigendian = 0) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/chunk.py", line 63, in __init__ raise EOFError i know the StringIO stuff works for creation of wav file and i tried the following and it works import StringIO buffer = StringIO.StringIO() audio_out = wave.open(buffer, 'w') audio_out.setframerate(m.getRate()) audio_out.setsampwidth(2) audio_out.setcomptype('NONE', 'not compressed') audio_out.setnchannels(1) audio_out.writeframes(raw_audio) audio_out.close() buffer.flush() # these lines do not work... # buffer.output(wav_buffer) # file = wave.open(buffer, 'r') # this file plays out fine in VLC file = open(FILE_NAME + ".wav", 'w') file.write(buffer.getvalue()) file.close() buffer.close()

    Read the article

  • Correct way to Convert 16bit PCM Wave data to float

    - by fredley
    I have a wave file in 16bit PCM form. I've got the raw data in a byte[] and a method for extracting samples, and I need them in float format, i.e. a float[] to do a Fourier Transform. Here's my code, does this look right? I'm working on Android so javax.sound.sampled etc. is not available. private static short getSample(byte[] buffer, int position) { return (short) (((buffer[position + 1] & 0xff) << 8) | (buffer[position] & 0xff)); } ... float[] samples = new float[samplesLength]; for (int i = 0;i<input.length/2;i+=2){ samples[i/2] = (float)getSample(input,i) / (float)Short.MAX_VALUE; }

    Read the article

  • Why do calls to waveOutGetPosition hang?

    - by MusiGenesis
    I'm using the winmm.dll API method waveOutGetPosition to get the current position of the playback of a WAV file. Sometimes this works as expected for me, but eventually one of the calls never returns and my application locks up. I found this thread with a few users who have experienced the same problem: http://social.msdn.microsoft.com/Forums/en-US/windowsgeneraldevelopmentissues/thread/c6a1e80e-4a18-47e7-af11-56a89f638ad7 but no solution. Has anyone run into this problem before?

    Read the article

  • What does LAME text does in MP3 file?

    - by Dims
    I see here http://en.wikipedia.org/wiki/MP3 that MP3 file consists of MP3 headers interchanged with MP3 data. MP3 header consist of few bytes. But here is my MP3 file dump with ID3 tag cut. Header is highlighted with blue. You can see that "LAME3.96" text is highlighted with green. What does it does there? Is this a part of MP3 elementary stream? Or this is the part of some headers I didn't tag?

    Read the article

  • 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!

    Read the article

  • Rapid calls to fread crashes the application

    - by Slynk
    I'm writing a function to load a wave file and, in the process, split the data into 2 separate buffers if it's stereo. The program gets to i = 18 and crashes during the left channel fread pass. (You can ignore the couts, they are just there for debugging.) Maybe I should load the file in one pass and use memmove to fill the buffers? if(params.channels == 2){ params.leftChannelData = new unsigned char[params.dataSize/2]; params.rightChannelData = new unsigned char[params.dataSize/2]; bool isLeft = true; int offset = 0; const int stride = sizeof(BYTE) * (params.bitsPerSample/8); for(int i = 0; i < params.dataSize; i += stride) { std::cout << "i = " << i << " "; if(isLeft){ std::cout << "Before Left Channel, "; fread(params.leftChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Left Channel, "; } else{ std::cout << "Before Right Channel, "; fread(params.rightChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Right Channel, "; offset += stride; std::cout << "After offset incr.\n"; } isLeft != isLeft; } } else { params.leftChannelData = new unsigned char[params.dataSize]; fread(params.leftChannelData, sizeof(BYTE), params.dataSize, file); }

    Read the article

  • byte[] to wav file

    - by John
    Hi, It would be great if you could tell me how I could save a byte[] to a wav file. Sometimes I need to set different samplerate, number of bits and channels. Thanks for your help.

    Read the article

  • DSP - Filter sweep effect

    - by Trap
    I'm implementing a 'filter sweep' effect (I don't know if it's called like that). What I do is basically create a low-pass filter and make it 'move' along a certain frequency range. To calculate the filter cut-off frequency at a given moment I use a user-provided linear function, which yields values between 0 and 1. My first attempt was to directly map the values returned by the linear function to the range of frequencies, as in cf = freqRange * lf(x). Although it worked ok it looked as if the sweep ran much faster when moving through low frequencies and then slowed down during its way to the high frequency zone. I'm not sure why is this but I guess it's something to do with human hearing perceiving changes in frequency in a non-linear manner. My next attempt was to move the filter's cut-off frequency in a logarithmic way. It works much better now but I still feel that the filter doesn't move at a constant perceived speed through the range of frequencies. How should I divide the frequency space to obtain a constant perceived sweep speed? Thanks in advance.

    Read the article

  • Simple sound effect loop using AudioToolKit

    - by Typeoneerror
    I've created a few sounds for use in my game. I can play them at certain events without issue: // create sounds CFBundleRef mainBundle; mainBundle = CFBundleGetMainBundle(); _soundFileShake = CFBundleCopyResourceURL(mainBundle, CFSTR("shake"), CFSTR("wav"), NULL); AudioServicesCreateSystemSoundID(_soundFileShake, &_soundIdShake); // later... AudioServicesPlaySystemSound(_soundIdShake); The game has a mechanism which allows you to shake the device to activate some functionality. I've got the shaking code done so I get get a "shaking started" and "shaking ended" message to my game. What I need to have happen is start playing "shave.wav" when shaking starts and loop it until it stops. Is there a way to do this with AudioToolbox/AudioServices? How could I do this if not?

    Read the article

< Previous Page | 59 60 61 62 63 64 65 66 67 68 69 70  | Next Page >