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  • Reading part of system file contents with PHP

    - by zx
    Hi, I have a config file, in etc/ called 1.conf Here is the contents.. [2-main] exten => s,1,Macro(speech,"hi {$VAR1} how is your day going?") exten => s,4,Macro(dial,2,555555555) exten => s,2,Macro(speech,"lkqejqe;j") exten => s,3,Macro(speech,"hi there") exten => s,5,Macro(speech,"this is a test ") exten => s,6,Macro(speech,"testing 2") exten => s,7,Macro(speech,"this is a test") exten => 7,1,Goto(2-tester2,s,1) exten => 1,1,Goto(2-aa,s,1) [2-tester] [2-aa] exten => 1,1,Goto(2-main,s,1) How can I read the content in between speech for example.. exten => s,6,Macro(speech,"testing 2") Just get "testing 2" from that. Thank you in advance!

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  • Conceal packet loss in PCM stream

    - by ZeroDefect
    I am looking to use 'Packet Loss Concealment' to conceal lost PCM frames in an audio stream. Unfortunately, I cannot find a library that is accessible without all the licensing restrictions and code bloat (...up for some suggestions though). I have located some GPL code written by Steve Underwood for the Asterisk project which implements PLC. There are several limitations; although, as Steve suggests in his code, his algorithm can be applied to different streams with a bit of work. Currently, the code works with 8kHz 16-bit signed mono streams. Variations of the code can be found through a simple search of Google Code Search. My hope is that I can adapt the code to work with other streams. Initially, the goal is to adjust the algorithm for 8+ kHz, 16-bit signed, multichannel audio (all in a C++ environment). Eventually, I'm looking to make the code available under the GPL license in hopes that it could be of benefit to others... Attached is the code below with my efforts. The code includes a main function that will "drop" a number of frames with a given probability. Unfortunately, the code does not quite work as expected. I'm receiving EXC_BAD_ACCESS when running in gdb, but I don't get a trace from gdb when using 'bt' command. Clearly, I'm trampimg on memory some where but not sure exactly where. When I comment out the *amdf_pitch* function, the code runs without crashing... int main (int argc, char *argv[]) { std::ifstream fin("C:\\cc32kHz.pcm"); if(!fin.is_open()) { std::cout << "Failed to open input file" << std::endl; return 1; } std::ofstream fout_repaired("C:\\cc32kHz_repaired.pcm"); if(!fout_repaired.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } std::ofstream fout_lossy("C:\\cc32kHz_lossy.pcm"); if(!fout_lossy.is_open()) { std::cout << "Failed to open output repaired file" << std::endl; return 1; } audio::PcmConcealer Concealer; Concealer.Init(1, 16, 32000); //Generate random numbers; srand( time(NULL) ); int value = 0; int probability = 5; while(!fin.eof()) { char arr[2]; fin.read(arr, 2); //Generate's random number; value = rand() % 100 + 1; if(value <= probability) { char blank[2] = {0x00, 0x00}; fout_lossy.write(blank, 2); //Fill in data; Concealer.Fill((int16_t *)blank, 1); fout_repaired.write(blank, 2); } else { //Write data to file; fout_repaired.write(arr, 2); fout_lossy.write(arr, 2); Concealer.Receive((int16_t *)arr, 1); } } fin.close(); fout_repaired.close(); fout_lossy.close(); return 0; } PcmConcealer.hpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #ifndef __PCMCONCEALER_HPP__ #define __PCMCONCEALER_HPP__ /** 1. What does it do? The packet loss concealment module provides a suitable synthetic fill-in signal, to minimise the audible effect of lost packets in VoIP applications. It is not tied to any particular codec, and could be used with almost any codec which does not specify its own procedure for packet loss concealment. Where a codec specific concealment procedure exists, the algorithm is usually built around knowledge of the characteristics of the particular codec. It will, therefore, generally give better results for that particular codec than this generic concealer will. 2. How does it work? While good packets are being received, the plc_rx() routine keeps a record of the trailing section of the known speech signal. If a packet is missed, plc_fillin() is called to produce a synthetic replacement for the real speech signal. The average mean difference function (AMDF) is applied to the last known good signal, to determine its effective pitch. Based on this, the last pitch period of signal is saved. Essentially, this cycle of speech will be repeated over and over until the real speech resumes. However, several refinements are needed to obtain smooth pleasant sounding results. - The two ends of the stored cycle of speech will not always fit together smoothly. This can cause roughness, or even clicks, at the joins between cycles. To soften this, the 1/4 pitch period of real speech preceeding the cycle to be repeated is blended with the last 1/4 pitch period of the cycle to be repeated, using an overlap-add (OLA) technique (i.e. in total, the last 5/4 pitch periods of real speech are used). - The start of the synthetic speech will not always fit together smoothly with the tail of real speech passed on before the erasure was identified. Ideally, we would like to modify the last 1/4 pitch period of the real speech, to blend it into the synthetic speech. However, it is too late for that. We could have delayed the real speech a little, but that would require more buffer manipulation, and hurt the efficiency of the no-lost-packets case (which we hope is the dominant case). Instead we use a degenerate form of OLA to modify the start of the synthetic data. The last 1/4 pitch period of real speech is time reversed, and OLA is used to blend it with the first 1/4 pitch period of synthetic speech. The result seems quite acceptable. - As we progress into the erasure, the chances of the synthetic signal being anything like correct steadily fall. Therefore, the volume of the synthesized signal is made to decay linearly, such that after 50ms of missing audio it is reduced to silence. - When real speech resumes, an extra 1/4 pitch period of sythetic speech is blended with the start of the real speech. If the erasure is small, this smoothes the transition. If the erasure is long, and the synthetic signal has faded to zero, the blending softens the start up of the real signal, avoiding a kind of "click" or "pop" effect that might occur with a sudden onset. 3. How do I use it? Before audio is processed, call plc_init() to create an instance of the packet loss concealer. For each received audio packet that is acceptable (i.e. not including those being dropped for being too late) call plc_rx() to record the content of the packet. Note this may modify the packet a little after a period of packet loss, to blend real synthetic data smoothly. When a real packet is not available in time, call plc_fillin() to create a sythetic substitute. That's it! */ /*! Minimum allowed pitch (66 Hz) */ #define PLC_PITCH_MIN(SAMPLE_RATE) ((double)(SAMPLE_RATE) / 66.6) /*! Maximum allowed pitch (200 Hz) */ #define PLC_PITCH_MAX(SAMPLE_RATE) ((SAMPLE_RATE) / 200) /*! Maximum pitch OLA window */ //#define PLC_PITCH_OVERLAP_MAX(SAMPLE_RATE) ((PLC_PITCH_MIN(SAMPLE_RATE)) >> 2) /*! The length over which the AMDF function looks for similarity (20 ms) */ #define CORRELATION_SPAN(SAMPLE_RATE) ((20 * (SAMPLE_RATE)) / 1000) /*! History buffer length. The buffer must also be at leat 1.25 times PLC_PITCH_MIN, but that is much smaller than the buffer needs to be for the pitch assessment. */ //#define PLC_HISTORY_LEN(SAMPLE_RATE) ((CORRELATION_SPAN(SAMPLE_RATE)) + (PLC_PITCH_MIN(SAMPLE_RATE))) namespace audio { typedef struct { /*! Consecutive erased samples */ int missing_samples; /*! Current offset into pitch period */ int pitch_offset; /*! Pitch estimate */ int pitch; /*! Buffer for a cycle of speech */ float *pitchbuf;//[PLC_PITCH_MIN]; /*! History buffer */ short *history;//[PLC_HISTORY_LEN]; /*! Current pointer into the history buffer */ int buf_ptr; } plc_state_t; class PcmConcealer { public: PcmConcealer(); ~PcmConcealer(); void Init(int channels, int bit_depth, int sample_rate); //Process a block of received audio samples. int Receive(short amp[], int frames); //Fill-in a block of missing audio samples. int Fill(short amp[], int frames); void Destroy(); private: int amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames); void save_history(plc_state_t *s, short *buf, int channel_index, int frames); void normalise_history(plc_state_t *s); /** Holds the states of each of the channels **/ std::vector< plc_state_t * > ChannelStates; int plc_pitch_min; int plc_pitch_max; int plc_pitch_overlap_max; int correlation_span; int plc_history_len; int channel_count; int sample_rate; bool Initialized; }; } #endif PcmConcealer.cpp /* * Code adapted from Steve Underwood of the Asterisk Project. This code inherits * the same licensing restrictions as the Asterisk Project. */ #include "audio/PcmConcealer.hpp" /* We do a straight line fade to zero volume in 50ms when we are filling in for missing data. */ #define ATTENUATION_INCREMENT 0.0025 /* Attenuation per sample */ #if !defined(INT16_MAX) #define INT16_MAX (32767) #define INT16_MIN (-32767-1) #endif #ifdef WIN32 inline double rint(double x) { return floor(x + 0.5); } #endif inline short fsaturate(double damp) { if (damp > 32767.0) return INT16_MAX; if (damp < -32768.0) return INT16_MIN; return (short)rint(damp); } namespace audio { PcmConcealer::PcmConcealer() : Initialized(false) { } PcmConcealer::~PcmConcealer() { Destroy(); } void PcmConcealer::Init(int channels, int bit_depth, int sample_rate) { if(Initialized) return; if(channels <= 0 || bit_depth != 16) return; Initialized = true; channel_count = channels; this->sample_rate = sample_rate; ////////////// double min = PLC_PITCH_MIN(sample_rate); int imin = (int)min; double max = PLC_PITCH_MAX(sample_rate); int imax = (int)max; plc_pitch_min = imin; plc_pitch_max = imax; plc_pitch_overlap_max = (plc_pitch_min >> 2); correlation_span = CORRELATION_SPAN(sample_rate); plc_history_len = correlation_span + plc_pitch_min; ////////////// for(int i = 0; i < channel_count; i ++) { plc_state_t *t = new plc_state_t; memset(t, 0, sizeof(plc_state_t)); t->pitchbuf = new float[plc_pitch_min]; t->history = new short[plc_history_len]; ChannelStates.push_back(t); } } void PcmConcealer::Destroy() { if(!Initialized) return; while(ChannelStates.size()) { plc_state_t *s = ChannelStates.at(0); if(s) { if(s->history) delete s->history; if(s->pitchbuf) delete s->pitchbuf; memset(s, 0, sizeof(plc_state_t)); delete s; } ChannelStates.erase(ChannelStates.begin()); } ChannelStates.clear(); Initialized = false; } //Process a block of received audio samples. int PcmConcealer::Receive(short amp[], int frames) { if(!Initialized) return 0; int j = 0; for(int k = 0; k < ChannelStates.size(); k++) { int i; int overlap_len; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples) { /* Although we have a real signal, we need to smooth it to fit well with the synthetic signal we used for the previous block */ /* The start of the real data is overlapped with the next 1/4 cycle of the synthetic data. */ pitch_overlap = s->pitch >> 2; if (pitch_overlap > frames) pitch_overlap = frames; gain = 1.0 - s->missing_samples * ATTENUATION_INCREMENT; if (gain < 0.0) gain = 0.0; new_step = 1.0/pitch_overlap; old_step = new_step*gain; new_weight = new_step; old_weight = (1.0 - new_step)*gain; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->pitchbuf[s->pitch_offset] + new_weight * amp[index]); if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->missing_samples = 0; } save_history(s, amp, j, frames); j++; } return frames; } //Fill-in a block of missing audio samples. int PcmConcealer::Fill(short amp[], int frames) { if(!Initialized) return 0; int j =0; for(int k = 0; k < ChannelStates.size(); k++) { short *tmp = new short[plc_pitch_overlap_max]; int i; int pitch_overlap; float old_step; float new_step; float old_weight; float new_weight; float gain; short *orig_amp; int orig_len; orig_amp = amp; orig_len = frames; plc_state_t *s = ChannelStates.at(k); if (s->missing_samples == 0) { // As the gap in real speech starts we need to assess the last known pitch, //and prepare the synthetic data we will use for fill-in normalise_history(s); s->pitch = amdf_pitch(plc_pitch_min, plc_pitch_max, s->history + plc_history_len - correlation_span - plc_pitch_min, j, correlation_span); // We overlap a 1/4 wavelength pitch_overlap = s->pitch >> 2; // Cook up a single cycle of pitch, using a single of the real signal with 1/4 //cycle OLA'ed to make the ends join up nicely // The first 3/4 of the cycle is a simple copy for (i = 0; i < s->pitch - pitch_overlap; i++) s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]; // The last 1/4 of the cycle is overlapped with the end of the previous cycle new_step = 1.0/pitch_overlap; new_weight = new_step; for ( ; i < s->pitch; i++) { s->pitchbuf[i] = s->history[plc_history_len - s->pitch + i]*(1.0 - new_weight) + s->history[plc_history_len - 2*s->pitch + i]*new_weight; new_weight += new_step; } // We should now be ready to fill in the gap with repeated, decaying cycles // of what is in pitchbuf // We need to OLA the first 1/4 wavelength of the synthetic data, to smooth // it into the previous real data. To avoid the need to introduce a delay // in the stream, reverse the last 1/4 wavelength, and OLA with that. gain = 1.0; new_step = 1.0/pitch_overlap; old_step = new_step; new_weight = new_step; old_weight = 1.0 - new_step; for (i = 0; i < pitch_overlap; i++) { int index = (i * channel_count) + j; amp[index] = fsaturate(old_weight * s->history[plc_history_len - 1 - i] + new_weight * s->pitchbuf[i]); new_weight += new_step; old_weight -= old_step; if (old_weight < 0.0) old_weight = 0.0; } s->pitch_offset = i; } else { gain = 1.0 - s->missing_samples*ATTENUATION_INCREMENT; i = 0; } for ( ; gain > 0.0 && i < frames; i++) { int index = (i * channel_count) + j; amp[index] = s->pitchbuf[s->pitch_offset]*gain; gain -= ATTENUATION_INCREMENT; if (++s->pitch_offset >= s->pitch) s->pitch_offset = 0; } for ( ; i < frames; i++) { int index = (i * channel_count) + j; amp[i] = 0; } s->missing_samples += orig_len; save_history(s, amp, j, frames); delete [] tmp; j++; } return frames; } void PcmConcealer::save_history(plc_state_t *s, short *buf, int channel_index, int frames) { if (frames >= plc_history_len) { /* Just keep the last part of the new data, starting at the beginning of the buffer */ //memcpy(s->history, buf + len - plc_history_len, sizeof(short)*plc_history_len); int frames_to_copy = plc_history_len; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + frames - plc_history_len)) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = 0; return; } if (s->buf_ptr + frames > plc_history_len) { /* Wraps around - must break into two sections */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*(plc_history_len - s->buf_ptr)); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = plc_history_len - s->buf_ptr; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } frames -= (plc_history_len - s->buf_ptr); //memcpy(s->history, buf + (plc_history_len - s->buf_ptr), sizeof(short)*len); frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * (i + (plc_history_len - s->buf_ptr))) + channel_index; s->history[i] = buf[index]; } s->buf_ptr = frames; return; } /* Can use just one section */ //memcpy(s->history + s->buf_ptr, buf, sizeof(short)*len); short *hist_ptr = s->history + s->buf_ptr; int frames_to_copy = frames; for(int i = 0; i < frames_to_copy; i ++) { int index = (channel_count * i) + channel_index; hist_ptr[i] = buf[index]; } s->buf_ptr += frames; } void PcmConcealer::normalise_history(plc_state_t *s) { short *tmp = new short[plc_history_len]; if (s->buf_ptr == 0) return; memcpy(tmp, s->history, sizeof(short)*s->buf_ptr); memcpy(s->history, s->history + s->buf_ptr, sizeof(short)*(plc_history_len - s->buf_ptr)); memcpy(s->history + plc_history_len - s->buf_ptr, tmp, sizeof(short)*s->buf_ptr); s->buf_ptr = 0; delete [] tmp; } int PcmConcealer::amdf_pitch(int min_pitch, int max_pitch, short amp[], int channel_index, int frames) { int i; int j; int acc; int min_acc; int pitch; pitch = min_pitch; min_acc = INT_MAX; for (i = max_pitch; i <= min_pitch; i++) { acc = 0; for (j = 0; j < frames; j++) { int index1 = (channel_count * (i+j)) + channel_index; int index2 = (channel_count * j) + channel_index; //std::cout << "Index 1: " << index1 << ", Index 2: " << index2 << std::endl; acc += abs(amp[index1] - amp[index2]); } if (acc < min_acc) { min_acc = acc; pitch = i; } } std::cout << "Pitch: " << pitch << std::endl; return pitch; } } P.S. - I must confess that digital audio is not my forte...

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  • How does text-to-speech shortcut read any highlighted text?

    - by TP
    Hi, I am trying to develop a cocoa application that requires to read highlighted text from any application. But so far I cannot find any decent solution because the accessibility API isn't always work. (e.g. Firefox) Does anyone know it is implemented in text-to-speech included in Leopard? I have found this SO question but it isn't solved.

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  • How do I get the best quality screenshot for OCR (Optical Character Recognition) and what tool would

    - by GiH
    I'm trying to get some data into a text file from screenshots. Apparently screenshots don't work very well with OCR because they are 75dpi and the minimum for good quality OCR is 150dpi. Does anyone know how what the best way to be taking screenshots for OCR would be? And also, what the best OCR software for doing this would be? Right now I'm getting pretty good results with the free online tools such as new-ocr but it does make mistakes that I have to correct every now and then. So I'd like some tips. UPDATE: I tested out ABBYY screenshot and it was pretty bad... the online tools are better

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  • Doubt with c# handlers?

    - by aF
    I have this code in c# public void startRecognition(string pName) { presentationName = pName; if (WaveNative.waveInGetNumDevs() > 0) { string grammar = System.Environment.GetEnvironmentVariable("PUBLIC") + "\\SoundLog\\Presentations\\" + presentationName + "\\SpeechRecognition\\soundlog.cfg"; /* if (File.Exists(grammar)) { File.Delete(grammar); } executeCommand();*/ recContext = new SpSharedRecoContextClass(); recContext.CreateGrammar(0, out recGrammar); if (File.Exists(grammar)) { recGrammar.LoadCmdFromFile(grammar, SPLOADOPTIONS.SPLO_STATIC); recGrammar.SetGrammarState(SPGRAMMARSTATE.SPGS_ENABLED); recGrammar.SetRuleIdState(0, SPRULESTATE.SPRS_ACTIVE); } recContext.Recognition += new _ISpeechRecoContextEvents_RecognitionEventHandler(handleRecognition); //recContext.RecognitionForOtherContext += new _ISpeechRecoContextEvents_RecognitionForOtherContextEventHandler(handleRecognition); //System.Windows.Forms.MessageBox.Show("olari"); } } private void handleRecognition(int StreamNumber, object StreamPosition, SpeechLib.SpeechRecognitionType RecognitionType, SpeechLib.ISpeechRecoResult Result) { System.Windows.Forms.MessageBox.Show("entrei"); string temp = Result.PhraseInfo.GetText(0, -1, true); _recognizedText = ""; foreach (string word in recognizedWords) { if (temp.Contains(word)) { _recognizedText = word; } } } public void run() { if (File.Exists(System.Environment.GetEnvironmentVariable("PUBLIC") + "\\SoundLog\\Serialization\\Voices\\identifiedVoicesDLL.txt")) { deserializer = new XmlSerializer(_identifiedVoices.GetType()); FileStream fs = new FileStream(System.Environment.GetEnvironmentVariable("PUBLIC") + "\\SoundLog\\Serialization\\Voices\\identifiedVoicesDLL.txt", FileMode.Open); Object o = deserializer.Deserialize(fs); fs.Close(); _identifiedVoices = (double[])o; } if (File.Exists(System.Environment.GetEnvironmentVariable("PUBLIC") + "\\SoundLog\\Serialization\\Voices\\deletedVoicesDLL.txt")) { deserializer = new XmlSerializer(_deletedVoices.GetType()); FileStream fs = new FileStream(System.Environment.GetEnvironmentVariable("PUBLIC") + "\\SoundLog\\Serialization\\Voices\\deletedVoicesDLL.txt", FileMode.Open); Object o = deserializer.Deserialize(fs); fs.Close(); _deletedVoices = (ArrayList)o; } myTimer.Interval = 5000; myTimer.Tick += new EventHandler(clearData); myTimer.Start(); if (WaveNative.waveInGetNumDevs() > 0) { _waveFormat = new WaveFormat(_samples, 16, 2); _recorder = new WaveInRecorder(-1, _waveFormat, 8192 * 2, 3, new BufferDoneEventHandler(DataArrived)); _scaleHz = (double)_samples / _fftLength; _limit = (int)((double)_limitVoice / _scaleHz); SoundLogDLL.MelFrequencyCepstrumCoefficients.calculateFrequencies(_samples, _fftLength); } } startRecognition is a method for Speech Recognition that load a grammar and makes the recognition handler here: recContext.Recognition += new _ISpeechRecoContextEvents_RecognitionEventHandler(handleRecognition); Now I have a problem, when I call the method startRecognition before method run, both handlers (the recognition one and the handler for the Tick) work well. If a word is recognized, handlerRecognition method is called. But, when I call the method run before the method startRecognition, both methods seem to run well but then the recognition Handler is never executed! Even when I see that words are recognized (because they happear on the Windows Speech Recognition app). What can I do for the recognition handler be allways called?

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  • Building a Universal iPad App - Where is the device recognition code?

    - by JustinXXVII
    I noticed that when I create a new project in XCode for a Universal iPad/iPhone application, the template comes with two separate App Delegate files, one for each device. I can't seem to locate the place in code where it tries to decide which app delegate to use. I have an existing iPhone project I'd like to port to iPad. My thinking was that if I went ahead and designed the iPad project, I could just import my iPhone classes and nibs, and then use the App Delegate and UIDevice to decide which MainWindow.xib to load. The process went like this: Create an iPad project coded as a split-view create brand new classes and nibs for the iPad import iPhone classes and nibs Change build/target settings in accordance with Universal Apps Use [[UIDevice currentDevice] model] in the AppDelegate to decide which MainWindow to load Will this work, or does the app just automatically know which device it's being deployed on? Thanks for any insight you can offer.

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  • Attempting to calculate width of Map Overlays on the fly

    - by Bloudermilk
    Hey all- I am working on an Android app that utilizes the Google Maps API MapView, MapController, MapActivity, and ItemizedOverlay. I am basically trying to recreate certain functionalities of the Maps app (damn Google for not providing speech bubbles—for lack of a better name—for items!), particularly those speech bubbles. I have an invisible XML structure for the speech bubble in the XML layout file containing my MapView. The first time I show a speech bubble I grab that XML and remove it from it's current parent, applying some ItemizedOverlay.LayoutParams to it, and add it to the MapView as an Overlay. I position it above the item that was selected, fill it with the proper text, then set it to visible. This all works great. The goal here, though, is to also automatically animate the map to reveal any parts of a speech bubble that may be off-screen when it opens. So I'm trying popup.getWidth() (popup is the instance of my LinearLayout that is the speech bubble) after I do all the manipulation to the bubble, even after I display it to the user. Problem is, popup.getWidth() is returning me the width of the previously displayed popup, not the currently displayed one. I can't figure out why this would be happening if I'm fetching the width after I set it to visible with its new dimensions (which, by the way, are relative when I'm setting them with LayoutParams: fill_content for both width and height).. I have even tried forcing both the MapView and the "popup" to invalidate() before trying to fetch the width. Any ideas why this may be happening? How can I force the View to settle into its new dimensions before trying to fetch them? Thanks! Nick

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  • Creating Java Neural Networks

    - by Tori Wieldt
    A new article on OTN/Java, titled “Neural Networks on the NetBeans Platform,” by Zoran Sevarac, reports on Neuroph Studio, an open source Java neural network development environment built on top of the NetBeans Platform. This article shows how to create Java neural networks for classification.From the article:“Neural networks are artificial intelligence (machine learning technology) suitable for ill-defined problems, such as recognition, prediction, classification, and control. This article shows how to create some Java neural networks for classification. Note that Neuroph Studio also has support for image recognition, text character recognition, and handwritten letter recognition...”“Neuroph Studio is a Java neural network development environment built on top of the NetBeans Platform and Neuroph Framework. It is an IDE-like environment customized for neural network development. Neuroph Studio is a GUI that sits on top of Neuroph Framework. Neuroph Framework is a full-featured Java framework that provides classes for building neural networks…”The author, Zoran Sevarac, is a teaching assistant at Belgrade University, Department for Software Engineering, and a researcher at the Laboratory for Artificial Intelligence at Belgrade University. He is also a member of GOAI Research Network. Through his research, he has been working on the development of a Java neural network framework, which was released as the open source project Neuroph.Brainy stuff. Read the article here.

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  • Did 12.04 just add multi-touch gesture support mid-release?

    - by adempewolff
    I was reviewing the updates I was about to download today and I noticed that a lot of them had to do with gesture support, noticed that many of these were new installs rather than upgrades. Has 12.04 just added multi-touch gesture support mid-release? If so, what are the capabilities that this adds? Which applications already support these capabilities and can I expect others to add support in the near future? Here are the packages that were installed: Install: libframe6:amd64 (2.2.4-0ubuntu0.12.04.1), libgeis1:amd64 (2.2.9.2-0ubuntu1), libgrail5:amd64 (3.0.6-0ubuntu0.12.04.01, automatic) And here are those that were upgraded (also including many with touch support): Upgrade: libgrip0:amd64 (0.3.4-0ubuntu2~ubuntu12.04.1, 0.3.5-0ubuntu1~12.04.1), eog:amd64 (3.4.2-0ubuntu1, 3.4.2-0ubuntu1.1), ginn:amd64 (0.2.4-0ubuntu1, 0.2.4.1-0ubuntu1) Of which the descriptions for the new installs are, libgeis1: Gesture engine interface support A common API for clients of a systemwide gesture recognition and propagation engine. libframe6: Touch Frame Library This library handles the buildup and synchronization of a set of simultaneous touches. The library is input agnostic, with bindings for mtdev, frame and XI2.1. libgrail5: Gesture Recognition And Instantiation Library This library consists of an interface and tools for handling gesture recognition and gesture instantiation. Applications can use the grail callbacks to receive gesture primitives and raw input events from the underlying kernel device. And the descriptions for the upgraded packages are, ligrip0: provides multitouch gestures to GTK+ apps Libgrip hooks gesture recognition into GTK+ applications. ginn: Gesture Injector: No-GEIS, No-Toolkits A daemon with jinn-like wish-granting capabilities: it gives applications the ability to support a subset of multi-touch gestures without having to integrate GEIS or multi-touch GTK/Qt libs. Adding in a ton of new libraries and upgrading the existing components makes me wonder if 12.04 is meant to start natively supporting gestures other than two finger scroll in the near future. I expected these capabilities to be introduced soon but I thought that they would only be rolled out in a new release, not as upgrades for an existing release. Anyone have any info about this?

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  • What program to use to create subtitles for a video?

    - by user4124
    I have a video that I want to create subtitles for. Is there a program that can perform rudimentary speech-to-text in order to set the correct start/stop of each individual subtitle create rudimentary text subtitles (using some sort of speech-to-text) I know about gnome-subtitles. However, it requires extensive effort to create those subtitles manually. You need to select yourself the start and stop for each sentence. Youtube has the above features (creates rudimentary text subtitles at the correct timings, using speech-to-text). However I would rather not upload the videos to Youtube just to get my subtitles. Is it possible to do the subtitles efficiently in Ubuntu? Update: I plan to use the .srt subtitles only, and do not need to hard code them on the videos. My biggest requirement is to have the program automatically find the start/stop for each sentence, so that I write the text in it.

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  • using a PHP print_r array result in javascript/jquery

    - by Phil Jackson
    Hello all.I have a simple jquery/ajax request to the server which returns the structure and data of an array. I was wondering if there was a quick way in which I can use this array structure and data using jquery; A simple request; var token = $("#token").val(); $.ajax({ type: 'POST', url: './', data: 'token=' + token + '&re=8', cache: false, timeout: 5000, success: function(html){ // do something here with the html var } }); the result ( actual result from PHP's print_r(); ); Array ( [0] => Array ( [username] => Emmalene [contents] => <ul><li class="name">ACTwebDesigns</li><li class="speech">helllllllo</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">sds</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">Sponge</li><li class="speech">dick</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">arghh</li></ul> ) ) I was thinking along the lines of var demo = Array(html); // and then do something with the demo var Not sure if that would work it just sprang to mind. Any help is much appreciated.

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  • IE adding a attribute 'done[number]' ??

    - by Phil Jackson
    Hi all im struggling to find an answer to my problem here. I've made a IM application the same as facebooks but it is having problems in IE. The problem started as I kept seeing rn at the beginnning of every post made via IE. That was due to stripslashes function. But as I was investigating I noticed my tag was being added an attribut 'done'; <li><UL done67="7">rn<LI class=name>ACTwebDesigns</LI>rn<LI class=speech>hello</LI></UL></li> <li><UL done1="4">rn<LI class=name>ACTwebDesigns</LI>rn<LI class=speech>foo</LI></UL></li> <li><UL done84="10">rn<LI class=name>ACTwebDesigns</LI>rn<LI class=speech>barr</LI>rn<LI class=speech ?>foobar</LI></UL></li> <li><UL done88="14">rn<LI class=name>ACTwebDesigns</LI>rn<LI class=speech>this is a test</LI></UL></li> does anyone know of a reason why IE would add this attribute? EDIT: function checkForm() { $(".chat_input").keydown(function(e){ if ( e.keyCode == 13 ) { var data = strip_tags($(this).val()); var username = $("#users_username").val(); var box = $(this).parents('div:eq(0)'); $(this).val(""); if( box.find('.conversation_box li.' + session_number ).length == 0 ) { var conversation_list = box.find('.conversation_box').html(); var insert_data = '<li class="' + session_number + '"><ul><li class="name">' + username + '</li><li class="speech">' + data + '</li></ul></li>'; box.find('.conversation_box').html(conversation_list + insert_data); bottom(); }else{ var conversation_list = box.find('.conversation_box li.' + session_number + ' ul').html(); var insert_data = '<li class="speech"">' + data + '</li>'; box.find('.conversation_box li.' + session_number + ' ul').html(conversation_list + insert_data); bottom(); } return false; } }); } function store_chat(){ try{ var token = $("#token").val(); var openedBoxes = $("li.conversation_list"); openedBoxes.each(function(){ var boxContainer = $(this).parents('div:eq(0)'); var amount = boxContainer.find('.conversation_box li').length; var p = boxContainer.find('.open_trigger').html(); var u = $("#users_username").val(); if( amount != 0 ){ if( $(this).parents('div:eq(0)').find('.conversation_box li.' + session_number ).length != 0 ) { var session_contents = $(this).parents('div:eq(0)').find('.conversation_box li.' + session_number ).html(); alert( session_contents ); $.ajax({ type: 'POST', url: './', data: 'token=' + token + '&re=7&s=' + amount + '&sd=' + session_contents + '&u=' + u + '&p=' + p, cache: false, timeout: 5000, success: function(html){ auth(html); boxContainer.find('.conversation_box').html(html); bottom(); } }); } } }); }catch(er){} }

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  • looping through JSON array

    - by Phil Jackson
    Hi all. I have recently posted another question which straight away users pointed me in the right direction. $.ajax({ type: 'POST', url: './', data: 'token=' + token + '&re=8', cache: false, timeout: 5000, success: function(html){ auth(html); var JSON_array = eval(html); alert(JSON_array[0].username); } }); this returns the data correctly but I want to perform a kind of 'foreach'. the array contains data about multiple incoming and outgoing Instant Messages. So if a user is talking to more than one person at a time i need to loop through. the array's structure is as follows. Array ( [0] => Array ( [username] => Emmalene [contents] => <ul><li class="name">ACTwebDesigns</li><li class="speech">helllllllo</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">sds</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">Sponge</li><li class="speech">dick</li></ul> <ul><li class="name">ACTwebDesigns</li><li class="speech">arghh</li></ul> ) ) any help very much appreciated.

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  • Jquery CPU usage

    - by nharry
    I am using Jquery to make an image scroll across my page horizontally. The only problem is that it uses a serious amount of cpu usage. Up to 100% on a single core laptop in firefox. What could cause this??? Jquery <script> jQuery(document).ready(function() { $(".speech").animate({backgroundPosition: "-6000px 0px"}, 400000, null); }); </script> CSS .speech { /*position:fixed;*/ top:0; left:0px; height:400px; width:100%; z-index:-1; background:url(/images/speech.png) -300px -500px repeat-x; margin-right: auto; margin-left: auto; position: fixed; } HTML <div class="speech"></div>

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  • User intentions analysis

    - by Mark Bramnik
    I'm going to work on some project that would do a user-action recognition based on what he/she does in the system. As far as I understand there are two main parts here: Intercept the user actions (say http traffic in web/ui interaction in thick-client) analysis of user intentions. While the first part is rather technical and therefor easy to implement, the second one is AI related and can be academic. So I was wondering whether someone knows some third-parties/academic projects that would implement the 'action-recognition' stuff?

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  • Fix elements within objects at their position? (Prevent movement when resizing)

    - by Skadier
    I would like to know if it is possible to fix elements at their absolute position within custom elements in InDesign CS5? I created a kind of speech bubble and I would like to place a stripline within this bubble to separate two content areas. Just a little scheme to show the desired layout as Pseudo-Markup :D <speech-bubble> <textbox>HEADER SECTION</textbox> <stripline> <textbox>Some other text</textbox> </speech-bubble> I created something like this but with two separate elements which aren't connected. So I have to select both of them in order to move the whole bubble. Then I tried to connect them using Object->Paths->Create linked path but then the stripline moves and the HEADER SECTION moves too. All in all I would like to have a speech bubble which can be resized in order to hold more text but it shouldn't make the HEADER_SECTION larger or move the stripline. Hope you understand what I mean :D Thanks in advance!

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  • Use a custom domain and point to Tumblr blog

    - by jskye
    My domain mydomain.com is registered with GoDaddy. I wish to host my Tumblr blog on this domain with Nearly Free Speech hosting. My active nameservers at GoDaddy already point to my authoritative ones at Nearly Free Speech which is working. However I'm baffled as to how to get my correct configuration to point to my Tumblr. Preferably I'd like (A) my domain http://mydomain.com to host the blog and have http://www.mydomain.com redirect also to http://mydomain.com. If this is too difficult my next preference is (B) to have http://www.mydomain.com host the blog whilst http://mydomain.com redirects to http://www.mydomain.com My third preference is to have (C) a sub-domain like http://tumblr.mydomain.com or http://tumblr.mydomain.com to host the blog and I guess have http://mydomain.com and http://www.mydomain.com both redirect to it. I've tried having two aliases mydomain.com and www.mydomain.com pointing to my permanent Nearly Free Speech IP at mydomain.nfshost.com and when I try to add: (1) an A record pointing mydomain.com to the IP 66.6.44.4 as per Tumblr's instructions it tells me I already have the bare domain as an alias so I cant do that. (2) the A record on the www.mydomain.com alias. I can do this with either www.mydomain.com set as an alias or not. But when I tried this with mydomain.com set as the canonical name the result when visiting either mydomain.com or www.mydomain.com was both of them continually redirecting to each other until an error was thrown. So I was wondering if there is a ninja that could save me some hair-pulling and tell me the correct way to config A, or else B, or else C.

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  • Alternative methods to login to Windows

    - by jay
    I've always wanted some cool way to log into Windows like inserting a designated USB or voice recognition. Only recently did I discover http://www.luxand.com/blink/ which uses facial recognition to log into your PC. What other software lets you change the way you log in?

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  • Firefox add-on tab-specific buttons and scripts, similar to Page Actions in Google Chrome

    - by Chetan
    I want to write a Firefox extension that acts exactly like the built-in RSS feed scanner (as an exercise). It should do the following: On each new page / tab load, it should scan the content of the page for RSS feeds If there are RSS feeds in the page, it should put a button in the location bar that the user can click On clicking the button, a speech bubble should appear under the button (the way a speech bubble appears under the bookmarks star when you click on it), with information on the feeds and buttons to subscribe to them So my main questions are: What is the process to run specific content scripts for specific pages? What is the process to use the results of those scripts to update the speech bubble for each location bar button for each tab? Basically, I'm trying to figure out how to do in Firefox what Page Actions are in Google Chrome. Please help! :)

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  • JavaFX Threading issue - GUI freezing while method call ran.

    - by David Meadows
    Hi everyone, I hoped someone might be able to help as I'm a little stumped. I have a javafx class which runs a user interface, which includes a button to read some text out loud. When you press it, it invokes a Java object which uses the FreeTTS java speech synth to read out loud a String, which all works fine. The problem is, when the speech is being read out, the program stops completely until its completed. I'm not an expert on threaded applications, but I understand that usually if I extend the Thread class, and provided my implementation of the speech synth code inside an overridden run method, when I call start on the class it "should" create a new Thread, and run this code there, allowing the main thread which has the JavaFX GUI on to continue as normal. Any idea why this isn't the case? Thanks a lot in advance!

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  • visual c# 2010 communicating between two projects

    - by cake is a joke
    I am trying to create a windows form project, and use speech recognition for the Kinect with the Kinect to Windows SDK. I have the form application project (p1) and the Kinect speech project (p2) which is a command prompt. I made it a command prompt because it was the easiest way to do things. Anyway, I have read and found two things about this. 1)I found out how to run two projects at the same time in the same solution. 2) I also found out how to add references to get classes from each project to the other. So, how would I get variables from each project? Just by using project references, or something? P2 can recognize speech and save it to variables, if that counts for anything.

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