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  • Modify audio pitch of recorded clip (m4v)

    - by devcube
    I'm writing an app in which I'm trying to change the pitch of the audio when I'm recording a movie (.m4v). Or by modifying the audio pitch of the movie afterwards. I want the end result to be a movie (.m4v) that has the original length (i.e. same visual as original) but with modified sound pitch, e.g. a "chipmunk voice". A realtime conversion is to prefer if possible. I've read alot about changing audio pitch in iOS but most examples focus on playback, i.e. playing the sound with a different pitch. In my app I'm recording a movie (.m4v / AVFileTypeQuickTimeMovie) and saving it using standard AVAssetWriter. When saving the movie I have access to the following elements where I've tried to manipulate the audio (e.g. modify the pitch): audio buffer (CMSampleBufferRef) audio input writer (AVAssetWriterAudioInput) audio input writer options (e.g. AVNumberOfChannelsKey, AVSampleRateKey, AVChannelLayoutKey) asset writer (AVAssetWriter) I've tried to hook into the above objects to modify the audio pitch, but without success. I've also tried with Dirac as described here: Real Time Pitch Change In iPhone Using Dirac And OpenAL with AL_PITCH as described here: Piping output from OpenAL into a buffer And the "BASS" library from un4seen: Change Pitch/Tempo In Realtime I haven't found success with any of the above libs, most likely because I don't really know how to use them, and where to hook them into the audio saving code. There seems to be alot of librarys that have similar effects but focuses on playback or custom recording code. I want to manipulate the audio stream I've already got (AVAssetWriterAudioInput) or modify the saved movie clip (.m4v). I want the video to be unmodifed visually, i.e. played at the same speed. But I want the audio to go faster (like a chipmunk) or slower (like a ... monster? :)). Do you have any suggestions how I can modify the pitch in either real time (when recording the movie) or afterwards by converting the entire movie (.m4v file)? Should I look further into Dirac, OpenAL, SoundTouch, BASS or some other library? I want to be able to share the movie to others with modified audio, that's the reason I can't rely on modifying the pitch for playback only. Any help is appreciated, thanks!

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  • Limit the amount a camera can pitch

    - by ChocoMan
    I'm having problems trying to limit the range my camera can pitch. Currently my camera can pitch around a model without restriction, but having a hard time trying to find the value of the degree/radian the camera is currently at after pitching. Here is what I got so far: // Moves camera with thumbstick Pitch = pController.ThumbSticks.Right.Y * MathHelper.ToRadians(speedAngleMAX); // Pitch Camera around model public void cameraPitch(float pitch) { pitchAngle = ModelLoad.camTarget - ModelLoad.CameraPos; axisPitch = Vector3.Cross(Vector3.Up, pitchAngle); // pitch constrained to model's orientation axisPitch.Normalize(); ModelLoad.CameraPos = Vector3.Transform(ModelLoad.CameraPos - ModelLoad.camTarget, Matrix.CreateFromAxisAngle(axisPitch, pitch)) + ModelLoad.camTarget; } I've tried restraining the Y-camera position of ModelLoad.CameraPos.Y, but doing so gave me some unwanted results.

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  • Camera wont stay behind model after pitch, then rotation

    - by ChocoMan
    I have a camera position behind a model. Currently, if I push the left thumbstick making my model move forward, backward, or strafe, the camera stays with the model. If I push the right thumbstick left or right, the model rotates in those directions fine along with the camera rotating while maintaining its position relatively behind the model. But when I pitch the model up or down, then rotate the model afterwards, the camera moves slightly rotates in a clock-like fashion behind the model. If I do a few rotations of the model and try to pitch the camera, the camera will eventually be looking at the side, then eventually the front of the model while also rotating in a clock-like fashion. My question is, how do I keep the camera to pitch up and down behind the model no matter how much the model has rotated? Here is what I got: // Rotates model and pitches camera on its own axis public void modelRotMovement(GamePadState pController) { // Rotates Camera with model Yaw = pController.ThumbSticks.Right.X * MathHelper.ToRadians(angularSpeed); // Pitches Camera around model Pitch = pController.ThumbSticks.Right.Y * MathHelper.ToRadians(angularSpeed); AddRotation = Quaternion.CreateFromYawPitchRoll(Yaw, 0, 0); ModelLoad.MRotation *= AddRotation; MOrientation = Matrix.CreateFromQuaternion(ModelLoad.MRotation); } // Orbit (yaw) Camera around with model (only seeing back of model) public void cameraYaw(Vector3 axisYaw, float yaw) { ModelLoad.CameraPos = Vector3.Transform(ModelLoad.CameraPos - ModelLoad.camTarget, Matrix.CreateFromAxisAngle(axisYaw, yaw)) + ModelLoad.camTarget; } // Raise camera above or below model's shoulders public void cameraPitch(Vector3 axisPitch, float pitch) { ModelLoad.CameraPos = Vector3.Transform(ModelLoad.CameraPos - ModelLoad.camTarget, Matrix.CreateFromAxisAngle(axisPitch, pitch)) + ModelLoad.camTarget; } // Call in update method public void updateCamera() { cameraYaw(Vector3.Up, Yaw); cameraPitch(Vector3.Right, Pitch); } NOTE: I tried to use addPitch just like addRotation but it didn't work...

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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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  • Pitch camera around model

    - by ChocoMan
    Currently, my camera rotates with my model's Y-Axis (yaw) perfectly. What I'm having trouble with is rotating the X-Axis (pitch) along with it. I've tried the same method for cameraYaw() in the form of cameraPitch(), while adjusting the axis to Vector.Right, but the camera wouldn't pitch at all in accordance to the Y-Axes of the controller. Is there a way similar to this to get the same effect for pitching the camera around the model? // Rotates model on its own Y-axis public void modelRotMovement(GamePadState pController) { Yaw = pController.ThumbSticks.Right.X * MathHelper.ToRadians(speedAngleMAX); AddRotation = Quaternion.CreateFromYawPitchRoll(Yaw, 0, 0); ModelLoad.MRotation *= AddRotation; MOrientation = Matrix.CreateFromQuaternion(ModelLoad.MRotation); } // Orbit (yaw) Camera around model public void cameraYaw(Vector3 axis, float yaw, float pitch) { Pitch = pController.ThumbSticks.Right.Y * MathHelper.ToRadians(speedAngleMAX); ModelLoad.CameraPos = Vector3.Transform(ModelLoad.CameraPos - ModelLoad.camTarget, Matrix.CreateFromAxisAngle(axis, yaw)) + ModelLoad.camTarget; } public void updateCamera() { cameraYaw(Vector3.Up, Yaw); }

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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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  • MIDI - Continous control of pitch outside the standard pitch bend range?

    - by user1423893
    After I send a note on message I can control the pitch of a note within a ±2 semitone range using the pitch bend channel command. How can I continuously update a pitch of a note outside of the normal pitch bend range without retriggering the note (i.e. sending another note on message with the new pitch?) In other words the current note is still sounding after a note on message and its envelope has not reached the end of its release stage. I would like to change the pitch outside the pitch bend range, preferably anywhere within the audible frequency range.

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  • C# Audio - How to time stretch (different tempo, same pitch)

    - by heath
    I'm trying to make a winform app in C# (VS2008) that can load an mp3 (other formats would be nice, but mp3 at a minimum) and be able to adjust the playback speed (tempo) without affecting pitch. I really don't need any other audio effects. I tried using DirectShow but that doesn't seem to offer time stretch capabilities. I was able to incorporate irrklang but that does not seem to have the time stretch capability either. So now I've moved on to SoundTouch. That certainly has the capabilities but I'm very unclear on how to implement in C#. After a few days of this, about all I've accomplished is using DLLImport on the SoundTouch DLL and am able to successfully retrieve a version number. At this point, I'm not even sure if I can do what I'm trying to do with SoundTouch. Can anyone offer some guidance either on how to implement SoundTouch or a different library with the capabilities that I'm looking for? Thank you.

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  • RockBand-like voice app for PC/OSX / Real time pitch display software

    - by Sai Emrys
    I played Rock Band 2 for the first time a little while ago (at Notacon). One thing I enjoyed about it was getting real-time feedback about my singing. I think it'd be neat to have something like that to run alongside my usual music, so that I can sing to random stuff in my music collection and know when I'm hitting the notes. Is there something like this for PC - ideally for OSX, and ideally that can just operate on arbitrary songs? I don't really care if it's game-like (though that's neat too); I just want it for the singing feedback. And I have no need for pitch correction - ideally what I'd see is just the pitches of the notes in the music and (on the same scale, differently displayed) of the live microphone. I tried to STFW but got no salient hits. :-/ Thanks!

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  • Autocorrelation returns random results with mic input (using a high pass filter)

    - by Niall
    Hello, Sorry to ask a similar question to the one i asked before (FFT Problem (Returns random results)), but i've looked up pitch detection and autocorrelation and have found some code for pitch detection using autocorrelation. Im trying to do pitch detection of a users singing. Problem is, it keeps returning random results. I've got some code from http://code.google.com/p/yaalp/ which i've converted to C++ and modified (below). My sample rate is 2048, and data size is 1024. I'm detecting pitch of both a sine wave and mic input. The frequency of the sine wave is 726.0, and its detecting it to be 722.950820 (which im ok with), but its detecting the pitch of the mic as a random number from around 100 to around 1050. I'm now using a High pass filter to remove the DC offset, but it's not working. Am i doing it right, and if so, what else can i do to fix it? Any help would be greatly appreciated! double* doHighPassFilter(short *buffer) { // Do FFT: int bufferLength = 1024; float *real = malloc(bufferLength*sizeof(float)); float *real2 = malloc(bufferLength*sizeof(float)); for(int x=0;x<bufferLength;x++) { real[x] = buffer[x]; } fft(real, bufferLength); for(int x=0;x<bufferLength;x+=2) { real2[x] = real[x]; } for (int i=0; i < 30; i++) //Set freqs lower than 30hz to zero to attenuate the low frequencies real2[i] = 0; // Do inverse FFT: inversefft(real2,bufferLength); double* real3 = (double*)real2; return real3; } double DetectPitch(short* data) { int sampleRate = 2048; //Create sine wave double *buffer = malloc(1024*sizeof(short)); double amplitude = 0.25 * 32768; //0.25 * max length of short double frequency = 726.0; for (int n = 0; n < 1024; n++) { buffer[n] = (short)(amplitude * sin((2 * 3.14159265 * n * frequency) / sampleRate)); } doHighPassFilter(data); printf("Pitch from sine wave: %f\n",detectPitchCalculation(buffer, 50.0, 1000.0, 1, 1)); printf("Pitch from mic: %f\n",detectPitchCalculation(data, 50.0, 1000.0, 1, 1)); return 0; } // These work by shifting the signal until it seems to correlate with itself. // In other words if the signal looks very similar to (signal shifted 200 data) than the fundamental period is probably 200 data // Note that the algorithm only works well when there's only one prominent fundamental. // This could be optimized by looking at the rate of change to determine a maximum without testing all periods. double detectPitchCalculation(double* data, double minHz, double maxHz, int nCandidates, int nResolution) { //-------------------------1-------------------------// // note that higher frequency means lower period int nLowPeriodInSamples = hzToPeriodInSamples(maxHz, 2048); int nHiPeriodInSamples = hzToPeriodInSamples(minHz, 2048); if (nHiPeriodInSamples <= nLowPeriodInSamples) printf("Bad range for pitch detection."); if (1024 < nHiPeriodInSamples) printf("Not enough data."); double *results = new double[nHiPeriodInSamples - nLowPeriodInSamples]; //-------------------------2-------------------------// for (int period = nLowPeriodInSamples; period < nHiPeriodInSamples; period += nResolution) { double sum = 0; // for each sample, find correlation. (If they are far apart, small) for (int i = 0; i < 1024 - period; i++) sum += data[i] * data[i + period]; double mean = sum / 1024.0; results[period - nLowPeriodInSamples] = mean; } //-------------------------3-------------------------// // find the best indices int *bestIndices = findBestCandidates(nCandidates, results, nHiPeriodInSamples - nLowPeriodInSamples - 1); //note findBestCandidates modifies parameter // convert back to Hz double *res = new double[nCandidates]; for (int i=0; i < nCandidates;i++) res[i] = periodInSamplesToHz(bestIndices[i]+nLowPeriodInSamples, 2048); double pitch2 = res[0]; free(res); free(results); return pitch2; } /// Finds n "best" values from an array. Returns the indices of the best parts. /// (One way to do this would be to sort the array, but that could take too long. /// Warning: Changes the contents of the array!!! Do not use result array afterwards. int* findBestCandidates(int n, double* inputs,int length) { //int length = inputs.Length; if (length < n) printf("Length of inputs is not long enough."); int *res = new int[n]; double minValue = 0; for (int c = 0; c < n; c++) { // find the highest. double fBestValue = minValue; int nBestIndex = -1; for (int i = 0; i < length; i++) { if (inputs[i] > fBestValue) { nBestIndex = i; fBestValue = inputs[i]; } } // record this highest value res[c] = nBestIndex; // now blank out that index. if(nBestIndex!=-1) inputs[nBestIndex] = minValue; } return res; } int hzToPeriodInSamples(double hz, int sampleRate) { return (int)(1 / (hz / (double)sampleRate)); } double periodInSamplesToHz(int period, int sampleRate) { return 1 / (period / (double)sampleRate); } Thanks, Niall. Edit: Changed the code to implement a high pass filter with a cutoff of 30hz (from What Are High-Pass and Low-Pass Filters?, can anyone tell me how to convert the low-pass filter using convolution to a high-pass one?) but it's still returning random results. Plugging it into a VST host and using VST plugins to compare spectrums isn't an option to me unfortunately.

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  • C# - .WAV Playback Randomly High Pitch

    - by Nate Shoffner
    For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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  • A Great Idea I Never Had. Better Elevator Automation, no Buttons in Elevator.

    After my second year at engineering school (Cornell University), I spent the summer working in a dream job as an intern at the world famous Ford Design Center in Dearborn Michigan.  The group I worked in was an elite group of engineers whose responsibilities were to figure out better ways to build the cars that [...]...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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  • A Great Idea I Never Had. Better Elevator Automation, no Buttons in Elevator.

    After my second year at engineering school (Cornell University), I spent the summer working in a dream job as an intern at the world famous Ford Design Center in Dearborn Michigan.  The group I... This site is a resource for asp.net web programming. It has examples by Peter Kellner of techniques for high performance programming...Did you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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  • Cepstral Analysis for pitch detection

    - by Ohmu
    Hi! I'm looking to extract pitches from a sound signal. Someone on IRC just explain to me how taking a double FFT achieves this. Specifically: take FFT take log of square of absolute value (can be done with lookup table) take another FFT take absolute value I am attempting this using vDSP I can't understand how I didn't come across this technique earlier. I did a lot of hunting and asking questions; several weeks worth. More to the point, I can't understand why I didn't think of it. I am attempting to achieve this with vDSP library. it looks as though it has functions to handle all of these tasks. However, I'm wondering about the accuracy of the final result. I have previously used a technique which scours the frequency bins of a single FFT for local maxima. when it encounters one, it uses a cunning technique (the change in phase since the last FFT) to more accurately place the actual peak within the bin. I am worried that this precision will be lost with this technique I'm presenting here. I guess the technique could be used after the second FFT to get the fundamental accurately. But it kind of looks like the information is lost in step 2. as this is a potentially tricky process, could someone with some experience just look over what I'm doing and check it for sanity? also, I've heard there is an alternative technique involving fitting a quadratic over neighbouring bins. Is this of comparable accuracy? if so, I would favour it, as it doesn't involve remembering bin phases. so questions: does this approach makes sense? Can it be improved? I'm a bit worried about And the log square component; there seems to be a vDSP function to do exactly that: vDSP_vdbcon however, there is no indication it precalculates a log-table -- I assume it doesn't, as the FFT function requires an explicit pre-calculation function to be called and passed into it. and this function doesn't. Is there some danger of harmonics being picked up? is there any cunning way of making vDSP pull out the maxima, biggest first? Can anyone point me towards some research or literature on this technique? the main question: is it accurate enough? Can the accuracy be improved? I have just been told by an expert that the accuracy IS INDEED not sufficient. Is this the end of the line? Pi PS I get SO annoyed (npi) when I want to create tags, but cannot. :| I have suggested to the maintainers that SO keep track of attempted tags, but I'm sure I was ignored. we need tags for vDSP, accelerate framework, cepstral analysis

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  • Android library to get pitch from WAV file

    - by Sakura
    I have a list of sampled data from the WAV file. I would like to pass in these values into a library and get the frequency of the music played in the WAV file. For now, I will have 1 frequency in the WAV file and I would like to find a library that is compatible with Android. I understand that I need to use FFT to get the frequency domain. Is there any good libraries for that? I found that [KissFFT][1] is quite popular but I am not very sure how compatible it is on Android. Is there an easier and good library that can perform the task I want? EDIT: I tried to use JTransforms to get the FFT of the WAV file but always failed at getting the correct frequency of the file. Currently, the WAV file contains sine curve of 440Hz, music note A4. However, I got the result as 441. Then I tried to get the frequency of G4, I got the result as 882Hz which is incorrect. The frequency of G4 is supposed to be 783Hz. Could it be due to not enough samples? If yes, how much samples should I take? //DFT DoubleFFT_1D fft = new DoubleFFT_1D(numOfFrames); double max_fftval = -1; int max_i = -1; double[] fftData = new double[numOfFrames * 2]; for (int i = 0; i < numOfFrames; i++) { // copying audio data to the fft data buffer, imaginary part is 0 fftData[2 * i] = buffer[i]; fftData[2 * i + 1] = 0; } fft.complexForward(fftData); for (int i = 0; i < fftData.length; i += 2) { // complex numbers -> vectors, so we compute the length of the vector, which is sqrt(realpart^2+imaginarypart^2) double vlen = Math.sqrt((fftData[i] * fftData[i]) + (fftData[i + 1] * fftData[i + 1])); //fd.append(Double.toString(vlen)); // fd.append(","); if (max_fftval < vlen) { // if this length is bigger than our stored biggest length max_fftval = vlen; max_i = i; } } //double dominantFreq = ((double)max_i / fftData.length) * sampleRate; double dominantFreq = (max_i/2.0) * sampleRate / numOfFrames; fd.append(Double.toString(dominantFreq)); Can someone help me out? EDIT2: I manage to fix the problem mentioned above by increasing the number of samples to 100000, however, sometimes I am getting the overtones as the frequency. Any idea how to fix it? Should I use Harmonic Product Frequency or Autocorrelation algorithms?

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  • Ray picking - get direction from pitch and yaw

    - by Isaac Waller
    I am attempting to cast a ray from the center of the screen and check for collisions with objects. When rendering, I use these calls to set up the camera: GL11.glRotated(mPitch, 1, 0, 0); GL11.glRotated(mYaw, 0, 1, 0); GL11.glTranslated(mPositionX, mPositionY, mPositionZ); I am having trouble creating the ray, however. This is the code I have so far: ray.origin = new Vector(mPositionX, mPositionY, mPositionZ); ray.direction = new Vector(?, ?, ?); My question is: what should I put in the question mark spots? I.e. how can I create the ray direction from the pitch and roll? Any help would be much appreciated!

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  • Having to check collisions twice per game tic

    - by user22241
    I have vertically moving elevators (3 solid tiles wide) and static solid tiles. Each are separate entities and therefore have their own respective collision routines (to check for, and resolve, collisions with the main character) I check my vertical collisions after characters vertical movements and then horizontal collisions after horizontal movements. The problem is that I want my platform to kill the player if it squashes him from the top, and also if he's on a moving platform (that is moving up) that squashes him into a solid block. Correct behaviour, player on solid blocks being squashed from above by decending elevator Here is what happens. Gravity pushes character into solid block, solid block collision routine corrects characters position and sits him on the solid block which pushes him into the moving elevator, elevator routine then checks for collision and kills player. This assumes I am checking solid blocks first, then elevator collisions. However, if it's the other way around, this happens.... Incorrect behaviour, player on accending elevator gets pushed into solid blocks above Player is on an elevator moving up, gravity pushes him into the elevator, solid block CD routine detects no collision, no action taken. Elevator CD routine detects character has been pushed into elevator by gravity, corrects this by moving character up and sitting him on the elevator and pushes him into the solid blocks above, however the solid block vertical routine has now already run for this tic, so the game continues and the next solid block collision that is encountered is the horizontal routine. This detects a collision and moves the character out of the collision to the left or right of the block which looks odd to say the least (character should get killed here). The only way I've managed to get this working correctly is by running the solid block CD, then the elevator CD, then the solid block CD again straight after. This is clearly wasteful but I can't figure out how else to do this. Any help would be appreciated.

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  • Ubuntu Audio Pitch Shifting filter

    - by user777305
    I'm currently developing a video player software which intended to be an embeedded player. I'm using Java with VLCJ library for the video player. What i'm looking is a way, something to transform the audio output to make the output sound as oldman or a kid (pitch shifting, i guess is the name). VLC have this when enabling time stretch, but the video play speed is affected (slower to get oldman sound, but fast-forward-alike to get kid voice effect. Is there any solutions for this? I don't find this feature on VLC(J), so i think what i need is the audio output of the ubuntu itself (Ubuntu 12.04) to do this job. something like filtering audio output system wide. Any software or setting to do this? it also need to be controllable via command line to provide realtime effect changing. Thanks in advance.

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  • How to TDD Asynchronous Events?

    - by Padu Merloti
    The fundamental question is how do I create a unit test that needs to call a method, wait for an event to happen on the tested class and then call another method (the one that we actually want to test)? Here's the scenario if you have time to read further: I'm developing an application that has to control a piece of hardware. In order to avoid dependency from hardware availability, when I create my object I specify that we are running in test mode. When that happens, the class that is being tested creates the appropriate driver hierarchy (in this case a thin mock layer of hardware drivers). Imagine that the class in question is an Elevator and I want to test the method that gives me the floor number that the elevator is. Here is how my fictitious test looks like right now: [TestMethod] public void TestGetCurrentFloor() { var elevator = new Elevator(Elevator.Environment.Offline); elevator.ElevatorArrivedOnFloor += TestElevatorArrived; elevator.GoToFloor(5); //Here's where I'm getting lost... I could block //until TestElevatorArrived gives me a signal, but //I'm not sure it's the best way int floor = elevator.GetCurrentFloor(); Assert.AreEqual(floor, 5); } Edit: Thanks for all the answers. This is how I ended up implementing it: [TestMethod] public void TestGetCurrentFloor() { var elevator = new Elevator(Elevator.Environment.Offline); elevator.ElevatorArrivedOnFloor += (s, e) => { Monitor.Pulse(this); }; lock (this) { elevator.GoToFloor(5); if (!Monitor.Wait(this, Timeout)) Assert.Fail("Elevator did not reach destination in time"); int floor = elevator.GetCurrentFloor(); Assert.AreEqual(floor, 5); } }

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  • Einmal nach oben, bitte. Mit den Oracle Communities.

    - by A&C Redaktion
    Stellen Sie sich vor, Sie treffen im Fahrstuhl einen bekannten und vertrauten Geschäftspartner. Theoretisch bleibt Ihnen nur die Fahrtzeit vom Erdgeschoss in den dritten Stock, um eine wichtige Angelegenheit zu erklären. Das nennt man „Elevator Pitch". Anwar Behan und Bernhard Adelmann haben dieses beliebte Szenario für Sie nachgestellt und erklären so auf unterhaltsame Weise, welchen Mehrwert die Oracle Communities für den Oracle Partner haben. Mit den Oracle Communities geht es für Partner nach oben und hier zum Video.

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  • WebGL First Person Camera - Matrix issues

    - by Ryan Welsh
    I have been trying to make a WebGL FPS camera.I have all the inputs working correctly (I think) but when it comes to applying the position and rotation data to the view matrix I am a little lost. The results can be viewed here http://thistlestaffing.net/masters/camera/index.html and the code here var camera = { yaw: 0.0, pitch: 0.0, moveVelocity: 1.0, position: [0.0, 0.0, -70.0] }; var viewMatrix = mat4.create(); var rotSpeed = 0.1; camera.init = function(canvas){ var ratio = canvas.clientWidth / canvas.clientHeight; var left = -1; var right = 1; var bottom = -1.0; var top = 1.0; var near = 1.0; var far = 1000.0; mat4.frustum(projectionMatrix, left, right, bottom, top, near, far); viewMatrix = mat4.create(); mat4.rotateY(viewMatrix, viewMatrix, camera.yaw); mat4.rotateX(viewMatrix, viewMatrix, camera.pitch); mat4.translate(viewMatrix, viewMatrix, camera.position); } camera.update = function(){ viewMatrix = mat4.create(); mat4.rotateY(viewMatrix, viewMatrix, camera.yaw); mat4.rotateX(viewMatrix, viewMatrix, camera.pitch); mat4.translate(viewMatrix, viewMatrix, camera.position); } //prevent camera pitch from going above 90 and reset yaw when it goes over 360 camera.lockCamera = function(){ if(camera.pitch > 90.0){ camera.pitch = 90; } if(camera.pitch < -90){ camera.pitch = -90; } if(camera.yaw <0.0){ camera.yaw = camera.yaw + 360; } if(camera.yaw >360.0){ camera.yaw = camera.yaw - 0.0; } } camera.translateCamera = function(distance, direction){ //calculate where we are looking at in radians and add the direction we want to go in ie WASD keys var radian = glMatrix.toRadian(camera.yaw + direction); //console.log(camera.position[3], radian, distance, direction); //calc X coord camera.position[0] = camera.position[0] - Math.sin(radian) * distance; //calc Z coord camera.position[2] = camera.position [2] - Math.cos(radian) * distance; console.log(camera.position [2] - (Math.cos(radian) * distance)); } camera.rotateUp = function(distance, direction){ var radian = glMatrix.toRadian(camera.pitch + direction); //calc Y coord camera.position[1] = camera.position[1] + Math.sin(radian) * distance; } camera.moveForward = function(){ if(camera.pitch!=90 && camera.pitch!=-90){ camera.translateCamera(-camera.moveVelocity, 0.0); } camera.rotateUp(camera.moveVelocity, 0.0); } camera.moveBack = function(){ if(camera.pitch!=90 && camera.pitch!=-90){ camera.translateCamera(-camera.moveVelocity, 180.0); } camera.rotateUp(camera.moveVelocity, 180.0); } camera.moveLeft = function(){ camera.translateCamera(-camera.moveVelocity, 270.0); } camera.moveRight = function(){ camera.translateCamera(-camera.moveVelocity, 90.0); } camera.lookUp = function(){ camera.pitch = camera.pitch + rotSpeed; camera.lockCamera(); } camera.lookDown = function(){ camera.pitch = camera.pitch - rotSpeed; camera.lockCamera(); } camera.lookLeft = function(){ camera.yaw= camera.yaw - rotSpeed; camera.lockCamera(); } camera.lookRight = function(){ camera.yaw = camera.yaw + rotSpeed; camera.lockCamera(); } . If there is no problem with my camera then I am doing some matrix calculations within my draw function where a problem might be. //position cube 1 worldMatrix = mat4.create(); mvMatrix = mat4.create(); mat4.translate(worldMatrix, worldMatrix, [-20.0, 0.0, -30.0]); mat4.multiply(mvMatrix, worldMatrix, viewMatrix); setShaderMatrix(); gl.bindBuffer(gl.ARRAY_BUFFER, vertexBuffer); gl.vertexAttribPointer(shaderProgram.attPosition, 3, gl.FLOAT, false, 8*4,0); gl.vertexAttribPointer(shaderProgram.attTexCoord, 2, gl.FLOAT, false, 8*4, 3*4); gl.vertexAttribPointer(shaderProgram.attNormal, 3, gl.FLOAT, false, 8*4, 5*4); gl.activeTexture(gl.TEXTURE0); gl.bindTexture(gl.TEXTURE_2D, myTexture); gl.uniform1i(shaderProgram.uniSampler, 0); gl.useProgram(shaderProgram); gl.drawArrays(gl.TRIANGLES, 0, vertexBuffer.numItems); //position cube 2 worldMatrix = mat4.create(); mvMatrix = mat4.create(); mat4.multiply(mvMatrix, worldMatrix, viewMatrix); mat4.translate(worldMatrix, worldMatrix, [40.0, 0.0, -30.0]); setShaderMatrix(); gl.drawArrays(gl.TRIANGLES, 0, vertexBuffer.numItems); //position cube 3 worldMatrix = mat4.create(); mvMatrix = mat4.create(); mat4.multiply(mvMatrix, worldMatrix, viewMatrix); mat4.translate(worldMatrix, worldMatrix, [20.0, 0.0, -100.0]); setShaderMatrix(); gl.drawArrays(gl.TRIANGLES, 0, vertexBuffer.numItems); camera.update();

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  • How to rotate 3D axis(XYZ) using Yaw,Pitch,Roll angles in Opengl

    - by user3639338
    I am working Pose estimation with capturing from camera with Opencv. Now I had three angle(Yaw,Pitch,Roll) from each frame(Head) using my code.How to rotate 3D axis(XYZ) those three angle using opengl ? I draw 3D axis using opengl. I have Problem with rotate this axis for returning each frame(Head) using VideoCapture camera input from my code.I cant rotate continuously using returning three angle my code.

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