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  • Looping music with intro in XNA using SoundEffect

    - by Jordan Roher
    I have two sound files: Sound A is an 18 second intro designed to be played once Sound B is a 1 minute looping track I'd like to play Sound A once, then once Sound A is done, immediately play Sound B and keep looping Sound B until I tell it to stop. This is supposed to be looping town music in an RPG. I've tried doing this in code using just SoundEffect, but there's a tiny yet noticeable gap between the end of Sound A and the beginning of Sound B. Even if I put monitoring code watching Sound A's SoundEffectInstance.State in the Update() function, I haven't been able to start Sound B exactly when Sound A finishes so that it's seamless. I'd prefer to use SoundEffect because I can load WMA files rather than being stuck with WAVs in XACT.

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  • Determining what frequencies correspond to the x axis in aurioTouch sample application

    - by eagle
    I'm looking at the aurioTouch sample application for the iPhone SDK. It has a basic spectrum analyzer implemented when you choose the "FFT" option. One of the things the app is lacking is X axis labels (i.e. the frequency labels). In the aurioTouchAppDelegate.mm file, in the function - (void)drawOscilloscope at line 652, it has the following code: if (displayMode == aurioTouchDisplayModeOscilloscopeFFT) { if (fftBufferManager->HasNewAudioData()) { if (fftBufferManager->ComputeFFT(l_fftData)) [self setFFTData:l_fftData length:fftBufferManager->GetNumberFrames() / 2]; else hasNewFFTData = NO; } if (hasNewFFTData) { int y, maxY; maxY = drawBufferLen; for (y=0; y<maxY; y++) { CGFloat yFract = (CGFloat)y / (CGFloat)(maxY - 1); CGFloat fftIdx = yFract * ((CGFloat)fftLength); double fftIdx_i, fftIdx_f; fftIdx_f = modf(fftIdx, &fftIdx_i); SInt8 fft_l, fft_r; CGFloat fft_l_fl, fft_r_fl; CGFloat interpVal; fft_l = (fftData[(int)fftIdx_i] & 0xFF000000) >> 24; fft_r = (fftData[(int)fftIdx_i + 1] & 0xFF000000) >> 24; fft_l_fl = (CGFloat)(fft_l + 80) / 64.; fft_r_fl = (CGFloat)(fft_r + 80) / 64.; interpVal = fft_l_fl * (1. - fftIdx_f) + fft_r_fl * fftIdx_f; interpVal = CLAMP(0., interpVal, 1.); drawBuffers[0][y] = (interpVal * 120); } cycleOscilloscopeLines(); } } From my understanding, this part of the code is what is used to decide which magnitude to draw for each frequency in the UI. My question is how can I determine what frequency each iteration (or y value) represents inside the for loop. For example, if I want to know what the magnitude is for 6kHz, I'm thinking of adding a line similar to the following: if (yValueRepresentskHz(y, 6)) NSLog(@"The magnitude for 6kHz is %f", (interpVal * 120)); Please note that although they chose to use the variable name y, from what I understand, it actually represents the x-axis in the visual graph of the spectrum analyzer, and the value of the drawBuffers[0][y] represents the y-axis.

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  • Am I "wasting" my time learning C and other low level stuff ?

    - by Andreas Grech
    I have just recently started learning C and the reason I did that was because frankly, I consider myself to be of a "less-developer" than the people who know and work with C. Thus I planned to start learning ASM, C, C++ and bought the K&R book and started pushing myself to learn the C Programming Language and up till now I'm doing great...learning about arrays the low level way (ie the pointer + offset thing), pointers and all that and obviously asking questions on stackoverflow for guidance. My problem is that sometimes I get thinking if instead of learning this low level stuff, maybe I should maybe spend more time learning newer, more widely used technologies...basically, more web stuff. Now I am well versed with both C# and ASP.Net and currently that's what I do for a living, but still there exists Microsoft technologies that I haven't quite touched upon...such as ASP.Net MVC, The Entity Framework etc... And those are only Microsoft Technologies...obviously there are other stuff that I would like to touch upon...stuff like Ruby, which would lead me to Ruby on Rails, or Python for Django or even Java and J2EE, or maybe even PHP; ie, basically mainly Web Stuff. Mind you, I did touch upon some of the stuff I mentioned earlier on, such as PHP and Java but I am still not quite versed in them as I am in C# and ASP.Net...but still, I think that by learning other languages that are used in the web environment will broaden my horizons...both as a developer who loves learning, and also Career wise. My point is, am I really using up my time correctly by learning older, lower level stuff? Stuff that for my current line of work, will most probably never use, but still is interesting to know ? To be frankly honest, I am also learning C so that I could, maybe someday, get into Electronics and Micro-controller programming but that is a whole new world for me and, if I choose to go there, will take some time to get adjusted to. And even then, I don't know if I can get a career in working in that line of work. ...but I still wonder about this question over and over...Am I doing the right thing by learning C instead of something (Web-stuff) that will most probably be more useful for me career-wise? I'm sorry for such asking such a long and most probably a boring question, but I feel as if this is the only place where I can ask such a question and get an honest answer from experts in the field. Thank you for your time.

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  • How to open wav file with Lua

    - by Pete Webbo
    Hello, I am trying to do some wav processing using Lua, but have fallen a the first hurdle! I cannot find a function or library that will allow me to load a wav file and access the raw data. There is one library, but it onl allows playing of wavs, not access to the raw data. Are there any out there? Cheers, Pete.

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  • Text to speech on iPhone

    - by lostInTransit
    Hi Is there any way we can convert text to speech in an iPhone app? Is it possible using the SDK? Thanks Are there any third-party TTS engines available for the iPhone? (AFAIK Acapela is not yet released)

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  • Zero Downtime with Hibernate

    - by Stephan Schmidt
    What changes to a database (MySQL in this case) does Hibernate survive (data, schema, ...)? I ask this because of zero downtime with Hibernate. Change database, split app servers into two clusters, redeploy the application on one of the clusters and switch application. Thanks Stephan

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  • lowest latency, least overhead app server?

    - by Mark Harrison
    I'm designing an application which will have a network interface for feeding out large numbers of very small metadata requests. The application code itself is very fast, basically looking up data cached in memory and sending it to the client. What's the absolute lowest latency I can get for a network application server running on a linux box? This will be an internal app running on gigE with no authentication. Any language/framework considered, with a preference for C, C++, or Python. Likewise for protocol, although HTTP would be nice.

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  • AudioQueue in-memory playback example

    - by Jonesy
    Does anybody know of any examples using AudioQueue that play from an in-memory source? All the examples I can find play from files (using AudioFileReadPackets) but in my particular case I am generating the data myself in realtime so ideally, I want to enqueue the data myself rather than sucking it out of a file using the callback. Any help much appreciated.

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  • loading mp3 from file using random access to flash.media.Sound

    - by Irfan Mulic
    We are migrating application from Delphi to Flex (Air) that plays mp3 files from random access big file. it has positions and sizes to extract mp3 data to FileStream-MemoryStream and then we use bass.dll to play it from memory stream. Now I have to play those same mp3's in flex but I am not sure how... I was reading something similar for reading/writing data using ByteArray from here but how to apply it to flash.media.Sound ? http://livedocs.adobe.com/flex/3/html/help.html?content=ByteArrays_2.html Any help?

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  • If I want to play the same sound 10 times per second, must I have 10 copies of that sound in memory?

    - by mystify
    I have a sound that needs to get played 10 times per second. The sound is 1 second long. So it does overlap like 10 times. However, as far as I understand the Finch sound library, I would need 10 different instances of a sound in place so that I can play it 10 times at almost the same time. When I have just one instance, the sound would stop and play from the beginning on every iteration, but not overlap with itself. How to do that?

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  • How to sound audible bell from crontab

    - by user1526251
    The command line: /bin/echo -e "\007" in bash will ring the bell. With the line: /bin/echo -e "\007" in my crontab I expected the bell to ring every minute, but it's silent. I know crontab is working because the line: /bin/touch $HOME/jkjkjk updates the file jkjkjk every minute as it should. I found a posting some years ago suggesting that standard output should be directed to /dev/tty1 in crontab. But the line: /bin/echo "\007" /dev/tty1 Still fails. What to try next?

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  • How to produce precisely-timed tone and silence?

    - by Bob Denny
    I have a C# project that plays Morse code for RSS feeds. I write it using Managed DirectX, only to discover that Managed DirectX is old and deprecated. The task I have is to play pure sine wave bursts interspersed with silence periods (the code) which are precisely timed as to their duration. I need to be able to call a function which plays a pure tone for so many milliseconds, then Thread.Sleep() then play another, etc. At its fastest, the tones and spaces can be as short as 40ms. It's working quite well in Managed DirectX. To get the precisely timed tone I create 1 sec. of sine wave into a secondary buffer, then to play a tone of a certain duration I seek forward to within x milliseconds of the end of the buffer then play. I've tried System.Media.SoundPlayer. It's a loser because you have to Play(), Sleep(), then Stop() for arbitrary tone lengths. The result is a tone that is too long, variable by CPU load. It takes an indeterminate amount of time to actually stop the tone. I then embarked on a lengthy attempt to use NAudio 1.3. I ended up with a memory resident stream providing the tone data, and again seeking forward leaving the desired length of tone remaining in the stream, then playing. This worked OK on the DirectSoundOut class for a while (see below) but the WaveOut class quickly dies with an internal assert saying that buffers are still on the queue despite PlayerStopped = true. This is odd since I play to the end then put a wait of the same duration between the end of the tone and the start of the next. You'd think that 80ms after starting Play of a 40 ms tone that it wouldn't have buffers on the queue. DirectSoundOut works well for a while, but its problem is that for every tone burst Play() it spins off a separate thread. Eventually (5 min or so) it just stops working. You can see thread after thread after thread exiting in the Output window while running the project in VS2008 IDE. I don't create new objects during playing, I just Seek() the tone stream then call Play() over and over, so I don't think it's a problem with orphaned buffers/whatever piling up till it's choked. I'm out of patience on this one, so I'm asking in the hopes that someone here has faced a similar requirement and can steer me in a direction with a likely solution.

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  • Playing an arbitrary tone with Android.

    - by fiXedd
    Is there any way to make Android emit a sound of arbitrary frequency (meaning, I don't want to have pre-recorded sound files)? I've looked around and ToneGenerator was the only thing I was able to find that was even close, but it seems to only be capable of outputting the standard DTMF tones. Any ideas?

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  • No mic activity with setLoopBack set to false - AS3

    - by Franky
    Trying to figure out why setloopback needs to be set to true for microphone activity to be detected. The problem is the echo feedback when using a macbook with a built in mic. If anyone has some ideas about this let me know. Right now I'm experimenting with toggling gain, depending on activity to simulate echo reduction. Not optimal though. @lessfame

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  • Squid handling of concurrent cache misses

    - by Oliver H-H
    We're using a Squid cache to off-load traffic from our web servers, ie. it's setup as a reverse-proxy responding to inbound requests before they hit our web servers. When we get blitzed with concurrent requests for the same request that's not in the cache, Squid proxies all the requests through to our web ("origin") servers. For us, this behavior isn't ideal: our origin servers gets bogged down trying to fulfill N identical requests concurrently. Instead, we'd like the first request to proxy through to the origin server, the rest of the requests to queue at the Squid layer, and then all be fulfilled by Squid when the origin server has responded to that first request. Does anyone know how to configure Squid to do this? We've read through the documentation multiple times and thoroughly web-searched the topic, but can't figure out how to do it. We use Akamai too and, interestingly, this is its default behavior. (However, Akamai has so many nodes that we still see lots of concurrent requests in certain traffic spike scenarios, even with Akamai's super-node feature enabled.) This behavior is clearly configurable for some other caches, eg. the Ehcache documentation offers the option "Concurrent Cache Misses: A cache miss will cause the filter chain, upstream of the caching filter to be processed. To avoid threads requesting the same key to do useless duplicate work, these threads block behind the first thread." Some folks call this behavior a "blocking cache," since the subsequent concurrent requests block behind the first request until it's fulfilled or timed-out. Thx for looking over my noob question! Oliver

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  • Piping SoX in Python - subprocess alternative?

    - by Cochise Ruhulessin
    I use SoX in an application. The application uses it to apply various operations on audiofiles, such as trimming. This works fine: from subprocess import Popen, PIPE kwargs = {'stdin': PIPE, 'stdout': PIPE, 'stderr': PIPE} pipe = Popen(['sox','-t','mp3','-', 'test.mp3','trim','0','15'], **kwargs) output, errors = pipe.communicate(input=open('test.mp3','rb').read()) if errors: raise RuntimeError(errors) This will cause problems on large files hower, since read() loads the complete file to memory; which is slow and may cause the pipes' buffer to overflow. A workaround exists: from subprocess import Popen, PIPE import tempfile import uuid import shutil import os kwargs = {'stdin': PIPE, 'stdout': PIPE, 'stderr': PIPE} tmp = os.path.join(tempfile.gettempdir(), uuid.uuid1().hex + '.mp3') pipe = Popen(['sox','test.mp3', tmp,'trim','0','15'], **kwargs) output, errors = pipe.communicate() if errors: raise RuntimeError(errors) shutil.copy2(tmp, 'test.mp3') os.remove(tmp) So the question stands as follows: Are there any alternatives to this approach, aside from writing a Python extension to the Sox C API?

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  • What is the best way to merge mp3 files?

    - by Dan Williams
    I've got many, many mp3 files that I would like to merge into a single file. I've used the command line method copy /b 1.mp3+2.mp3 3.mp3 but it's a pain when there's a lot of them and their namings are inconsistent. The time never seems to come out right either.

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  • Extracting note onset from MIDI

    - by Dolphin
    Hi I need to extract musical features (note details-pitch, duration, rhythm, loudness, note start time) from a polyphonic (having 2 scores for treble and bass - bass may also have chords) MIDI file. I'm using the jMusic API to extract these details from a MIDI file. My approach is to go through each score, into parts, then phrases and finally notes and extract the details. With my approach, it's reading all the treble notes first and then the bass notes - but chords are not captured (i.e. only a single note of the chord is taken), and I cannot identify from which point onwards are the bass notes. So what I tried was to get the note onsets (i.e. the start time of note being played) - since the starting time of both the treble and bass notes at the start of the piece should be same - But I cannot extract the note onset using jMusic API. Each time it shows 0.0. Is there any way I can identify the voice (treble or bass) of a note? And also all the notes of a chord? How is the voice or note onset for each note stored in MIDI? Is this different for each MIDI file? Any insight is greatly appreciated. Thanks in advance

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  • 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

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  • 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.**

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  • 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); } }

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  • 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... ^^

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