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  • Should I unit test the model returned by DefaultModelBinder?

    - by Byron Sommardahl
    I'm having some trouble unit testing the model returned by DefaultModelBinder. I want to feed in a fake form collection and check the model that it returns to make sure model properties are being bound properly. In my research, I'm not turning up -any- resources on testing the DefaultModelBinder. Maybe I'm missing something. Maybe I shouldn't be testing this part of MVC? Your thoughts?

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  • Rendering order in ContentControl - WPF

    - by Eden
    Hi, There is something that I don't understand regarding ContentControl: I've a class that inherits from ContentControl, and it overrides the OnRender method. And although I don't call base.OnRender(drawingContext) still the content is being rendered... How come? What am I missing? Thanks, Eden

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  • Hudson build on URL token

    - by Ritesh M Nayak
    I configured a hudson instance and have created jobs. While creating builds, I was able to see this option "Trigger the build by accessing this URL + SecretTOKEN" option. Now, I am unable to see that for any new jobs I create. Am I missing some setting or a configuration? The only change I made was running the servlet container from Root to a regular user.

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  • WCF RIA Services Silverlight 3.0

    - by John
    Hi, I have downloaded WCF RIA Services Beta from the following website: WCF RIA Services Beta for Visual Studio 2008 SP1 http://www.microsoft.com/downloads/details.aspx?FamilyID=76bb3a07-3846-4564-b0c3-27972bcaabce&displaylang=en#filelist But I am unable to add a reference to the following assembly : system.Windows.Ria.Data I searched at the downloaded location c:\Program files\Microsoft SDK's\RIA Services but i am unable to find this dll. Would appreciate if you could point me what I am missing here.

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  • triggering onchange event

    - by Ravi
    Hello All, I'm not able to get the below code to work. $("#grid_table td:nth-child(10) input").live("onchange",function () { alert("changed"); }); am I missing something here? Thanks. Ravi

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  • Unable to import nltk in NetBeans

    - by afs
    Hello all, I am trying to import NLTK in my python code and I get this error: Traceback (most recent call last): File "/home/afs/NetBeansProjects/NER/getNE_followers.py", line 7, in import nltk ImportError: No module named nltk I am using NetBeans: 6.7.1, Python 2.6 NLTK. My NLTK module is installed in /usr/local/lib/python2.6/dist-packages/nltk/ and I have added this in Python paths in Netbeans. What am I missing here? Thanks in advance.

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  • HTML Comment-out Add-In

    - by Velika
    Here is an old add in to quickly comment out HTML code. Maybe I am missing it, but it seems like there is a shortcut in VS2010 to scratch your tail with one click but commenting out HTML code is still awkward as hell. What's the easiest way to get a function like this working? Can I expect any add-ins that were written for older versions of VS to work in VS2010 w/o an upgrade?

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  • How to add XML elements into the toolbox in Visual Studio 2010 RC

    - by Alex Marshall
    I'm trying to edit an XML schema in Visual Studio 2010 Ultimate RC, but when I go to the toolbox (with the schema open and focused) there's absolutely nothing in the Toolbox view, even when every tutorial out there that I've read tells me that there should be. I've tried using the context menu option for resetting the Toolbox to no effect. Is there something I'm missing ? Something I need to install to get this feature of Visual Studio going ?

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  • Unable to find CFNetwork in XCode

    - by Baishampayan Ghose
    Hello, I am trying to use CFNetwork in my app so I tried adding CFNetwork.framework from the Edit Target dialog in XCode. The interesting thing is that CFNetwork is not visible in the dialog box at all. Am I missing anything? Do I have to add/install CFNetwork in some other way? Screenshot - Thanks in advance.

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  • Converting regex statment for sentance extraction to Ruby

    - by DavidP6
    I found this regex statement on the wiki (http://en.wikipedia.org/wiki/Sentence_boundary_disambiguation) for Sentence boundary disambiguation, but am not able to use it in a Ruby split statment. I'm not too good with regex so maybe I am missing something? This is statment: ((?<=[a-z0-9)][.?!])|(?<=[a-z0-9][.?!]\"))(\s|\r\n)(?=\"?[A-Z]) and this is what I tried in Ruby, but no go: text.split("((?<=[a-z0-9)][.?!])|(?<=[a-z0-9][.?!]\"))(\s|\r\n)(?=\"?[A-Z])")

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  • XSL ignores my whitespace even with the <xsl:text> tag

    - by danielle
    I'm making a header in my XSL code that includes multiple fields of information, i.e. "Name: Bob Birthdate: January 1 1900," etc. I enclosed them in tags as such: <xsl:text> Gender: Male </xsl:text> But on the page, the whitespace around Gender/Male is being ignored. Is there something I'm missing? Thanks in advance.

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  • Exception in MSAccess when adding datas with ' character

    - by Suryakavitha
    i am using windows forms application with MSAccess.... i got data from database table successfully but when i am trying to add data with (') character i got the exception that "OLEDB Exception: Syntax error(Missing Operator)inquery expression" ... Now how can i solve this problem? Plz tell me the solution of this problem.... all characters are accepted but apostrophe character only got error... Thanks in Advance

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  • Java GUI Overlay

    - by seurimas
    Hey, I want to make an little window like the sort of thing used by Teamspeak/Ventrillo or Steam/xFire where a window can be shown while still in a fullscreen game using Java. There was a similar question/answer ("How to create an overlay window in Java?") but that doesn't work for the particular game (EVE) whereas the previously mentioned overlays work just fine. What's the missing element? Or is it an entirely different method?

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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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  • Apache and file permissions

    - by Matthew
    I'm running LAMP on Ubuntu 8.04. Apache's username and group are www-data. I put my connection details and AES key in a file in a directory that's not web served. I chown-ed the files to www-data:www-data and set the permissions to 700. Still, the script that require()s these files will only run if I chmod the files to 755. What am I missing?

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  • Sending a JSON array to be received as a Dictionary<string,string>

    - by James Bond
    I have a method with the following signature: public ActionResult RenderFamilyTree(string name, Dictionary<string, string> children) I'm trying to call it from javascript using jQuery like this: $('#div_render').load( "<%= Url.Action("RenderFamilyTree") %>", { 'name': 'Raul', [ {'key':'key1','value':'value1'}, {'key':'key2','value':'value2'} ] }, function() { alert('Loaded'); } ); Am I missing something to get this to work?

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  • Query with Two Different DSN

    - by morant
    I have a query: The "X" tables are from one data source and the "Y" table is from another...but there is join between the data sources. I can't seem to figure out how to enter the two different connection strings for this to run. ConnectionString="Dsn=Xdb;uid=xxx;pwd=xxxxxxx" ProviderName="System.Data.Odbc" ConnectionString="Dsn=Ydb;uid=xxx;pwd=xxxxxxx" ProviderName="System.Data.Odbc" Is this possible...am I just missing something?

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  • Conditional or in C#

    - by vikitor
    Hello, I've been programming in JAVA and C all my years at Uni, but now I'm learning C# and building a small application, and I've found troubles with this: if (taxonType.Equals(null) ¦¦ taxonID == -1) I get a red underline for this conditional, and I don't really know why, because according to what I've seen that should be ok, but it is not. Is there something I'm missing? thank you all in advance, Victor

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  • FullCalendar Events: Confused how to enable resizing.

    - by Josiah Kiehl
    I'm using the FullCalendar plugin, and am very pleased with it, however I can't figure out how to enable resizing. I looked at the source of the calendar on the main FullCalendar page, and my code is no different save that I get my events from a source rather than explicitly define them. What am I missing? $('#schedule').fullCalendar({ events: '/campaigns/scheduled', editable: true });

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  • no sms deleted folder?

    - by Nicholas
    Hi, I'm new to Android and trying to convert a Windows Mobile app to java/android. In WM there are the followning standard folders: Inbox,Sent, dratfs, outbox, deleted. I'm able to access all folders except deleted with Uri.parse("content://sms/XXXXXX") 1) Is deleted folder missing? 2) Is it possible to create user folders, like "content://sms/My Folder" Thanks, Nicholas

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  • getaddrinfo is not statically compiled

    - by bobby
    I have included the header netdb.h, where getaddrinfo is included, but gcc issues this warning: warning: Using 'getaddrinfo' in statically linked applications requires at runtime the shared libraries from the glibc version used for linking gcc -m32 -static -s -O2 -std=c99 -D_POSIX_C_SOURCE=200112L myprogram.c How can I statically compile whatever file is missing ?

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