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  • Key to mute my Skype mic?

    - by lo_fye
    I'm using a Mac. Is there a keyboard shortcut for muting the mic during a call? I can't find one, and I often need to toggle mute/unmute. Clicking the button on the Skype interface is very inconvenient :(

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  • how to record mic input and pipe the output to another program

    - by acrs
    Hi everyone Im trying to follow a tutorial on generating truly random bits How To Generate Truly Random Bits This is the command from the tutorial but it does not work rec -c 1 -d /dev/dsp -r 8000 -t wav -s w - | ./noise-filter >bits I know i can record my mic input using rec -c 1 no.wav this is the command i tried using rec -c 1 -r 8000 -t wav -s noise.wav | ./noise-filter >bits but i get root@xxc:~/cc# rec -c 1 -r 8000 -t wav -s noise.wav - | ./noise-filter >bits rec WARN formats: can't set sample rate 8000; using 48000 rec FAIL sox: Input files must have the same sample-rate I have complied noise-filter noise-filter I think the tutorial is using an older version of SOX and REC I'm using sox: SoX v14.3.2 on Ubuntu 12.04 server Can someone please help me ?

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  • Game of Thrones : l'arme secrète de George R. R. Martin contre les virus, un PC sous DOS avec Wordstar 4.0 et sans internet

    Game of Thrones : L'arme secrète de George R. R. Martin contre les virus George R. R. MARTIN, auteur de la saga Game of Thrones et co-producteur de la série du même nom a révélé à un Talk-show américain avoir une arme secrète contre les virus qui pourraient attaquer son ordinateur et détruire ses documents. Pour écrire ses livres, il se sert d'un ordinateur non relié à internet et qui fonctionne sous DOS et comme traitement de texte Wordstar 4,0 ! À la question « pourquoi rester avec ce vieux...

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  • Permission denied (publickey,gssapi-with-mic,password) ssh error

    - by zentenk
    Heads up I'm a noob with linux and networking. I set up a ubuntu server and I have a static ip for my network. When I try to connect to the server at home (external), it prompts me to log in. I supply the correct password (or incorrect pw), I get the error Permission denied, please try again. and after 3 times I get Permission denied (publickey,gssapi-with-mic,password) I am however able to connect with SSH from another computer in the same network with ssh < internal ip of server > I'm connecting with mac os x and my config file is vanilla. Note: During installation of ubuntu it says I don't have a default route or something while doing auto network configuration, but I ignored it and continued the installation, could this be the problem? EDIT: I have tried the below, I have nothing in hosts.allow and also iptables shows the ports that I have allowed, which is 22. I checked the auth.log, and there is nothing when I connect to it remotely (even when it says permission denied). I have tried connecting to it internally and the correct authentication logs show. Any idea whats wrong?

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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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  • No mic activity with setLoopBack set to false - AS3

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

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  • iPhone SDK: How to get mic volume

    - by TheGambler
    I want to get the volume or even how much noise is coming through the mic. So someone is talking or some noise is going on in the background I want to know how much. Which framework would I use: Audio Toolbox, Audio Unit, AV Foundation, and Core Audio

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  • Strange sound from Laptop, Microphone, Speakers or Soundcard?

    - by David Schilling
    I have recently purchased a second hand computer. There didn't seem to be anything wrong with it on purchasing, however recently I have noticed a small issue. Any time I play sound or listen through my headphones, I can hear what seems to be the CPU working (ie when beginning a new task, there is a small crackling). Also if I tap my laptop I can hear the sound louder, as if it is being picked up by the mic and then playing that sound in real time through the speakers or headphones. I have tried reading other topics but cannot find an answer to this question. These are the specs I am running on; Windows 7 Home Premium 32bit Pentium Dual Core CPU T4300 @ 2.10Ghz 2.10Ghz I hope someone can help me with this issue, it would be much appreciated.

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  • Do We Indeed Have a Future? George Takei on Star Wars.

    - by Bil Simser
    George Takei (rhymes with Okay), probably best known for playing Hikaru Sulu on the original Star Trek, has always had deep concerns for the present and the future. Whether on Earth or among the stars, he has the welfare of humanity very much at heart. I was digging through my old copies of Famous Monsters of Filmland, a great publication on monster and films that I grew up with, and came across this. This was his reaction to STAR WARS from issue 139 of Famous Monsters of Filmland and was written June 6, 1977. It is reprinted here without permission but I hope since the message is still valid to this day and has never been reprinted anywhere, nobody will mind me sharing it. STAR WARS is the most pre-posterously diverting galactic escape and at the same time the most hideously credible portent of the future yet.While I thrilled to the exploits that reminded me of the heroics of Errol Flynn as Robin Hood, Burt Lancaster as the Crimson Pirate and Buster Crabbe as Flash Gordon, I was at the same time aghast at the phantasmagoric violence technology can place at our disposal. STAR WARS raised in my mind the question - do we indeed have a future?It seems to me what George Lucas has done is to masterfully guide us on a journey through space and time and bring us back face to face with today's reality. STAR WARS is more than science fiction, I think it is science fictitious reality.Just yesterday, June 7, 1977, I read that the United States will embark on the production of a neutron bomb - a bomb that will kill people on a gigantic scale but will not destroy buildings. A few days before that, I read that the Pentagon is fearful that the Soviets may have developed a warhead that could neutralize ours that have a capacity for that irrational concept overkill to the nth power. Already, it seems we have the technology to realize the awesome special effects simulations that we saw in the film.The political scene of STAR WARS is that of government by force and power, of revolutions based on some unfathomable grievance, survival through a combination of cunning and luck and success by the harnessing of technology -  a picture not very much at variance from the political headlines that we read today.And most of all, look at the people; both the heroes in the film and the reaction of the audience. First, the heroes; Luke Skywalker is a pretty but easily led youth. Without any real philosophy to guide him, he easily falls under the influence of a mystical old man believed previously to be an eccentric hermit. Recognize a 1960's hippie or a 1970's moonie? Han Solo has a philosophy coupled with courage and skill. His philosophy is money. His proficiency comes for a price - the highest. Solo is a thoroughly avaricious mercenary. And the Princess, a decisive, strong, self-confident and chilly woman. The audience cheered when she wielded a gun. In all three, I missed qualities that could be called humane - love, kindness, yes, I missed sensuality. I also missed a sense of ideals and faith. In this regard the machines seemed more human. They demonstrated real affection for each other and an occasional poutiness. They exhibited a sense of fidelity and constancy. The machines were humanized and the humans conversely seemed mechanical.As a member of the audience, I was swept up by the sheer romantic escapsim of it all. The deering-dos, the rope swing escape across the pit, the ray gun battles and especially the swash buckle with the ray swords. Great fun!But I just hope that we weren't too intoxicated by the escapism to be able to focus on the recognizable. I hope the beauty of the effects didn't narcotize our sensitivity to violence. I hope the people see through the fantastically well done futuristic mirrors to the disquieting reflection of our own society. I hope they enjoy STAR WARS without being "purely entertained".

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  • iPhone SDK Question with Audio/Mic

    - by Henry D'Andrea
    I am trying to do an app, to where when it launches, it will detect audio, and then play it back automatically. NO BUTTONS, nothing to press. Just a picture of something then, it listens for audio, then plays it back. Similar to the Talking Carl app in the App Store. Any ideas/help? Would appreciate it, if i could use the code with IB.

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  • Why does my mic boost automatically go to 100 on every boot?

    - by Ben
    When my computer turns on, it automatically sets the "mic boost" sound setting to 100. This causes a loud static sound in my speakers. I can manually go to alsamixer and turn the mic boost down manually, but I would prefer it if I didn't have to do this every time I turn the computer on. I've tried running sudo alsactl store after fixing the settings, and this does save them, but I have to run sudo alsactl restore to restore the settings. This means that I have to manually fix the sound every time I start the computer anyway, so it isn't really a fix. I tried putting sudo alsactl restore in my startup programs, but that didn't seem to fix anything. I'm running Ubuntu 12.04, but I started having this problem before upgrading from 11.10. I'm using a Sony Vaio laptop. I'm not really sure what made it start; it seemed like I just started having the problem randomly one day. Any help would be appreciated! Edit: here is the output from running amixer: http://paste.ubuntu.com/1060080/

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  • ALSA samples capture: cannot open device

    - by Randagio
    I'm quite new to Linux (Lubuntu 12.04 for sake of precision) and ALSA programming at all. I'm trying to write a C program to capture audio from internal PC microphone for processing it. So as first step I google a bit and I found this article for capturing audio samples A tutorial on using the ALSA Audio API but when I compile it and execute it with: ./capture "default" or ./capture "hw:0,0" and all the possible variants on theme it always raises the error: cannot open device hw:0,0 (no such file or directory). So the issue is: what is the name of the mic audio device to pass as parameter to record the audio from mic ? The mic is working ok because the Sound Recorder program records sounds perfectly and I can playback them. The output of the aplay -l is the following : **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 and this is the amixer output (cut) Simple mixer control 'Master',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined penum Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 4 [13%] [-40.50dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch penum Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [off] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive penum Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 22 [71%] [-1.50dB] [on] Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mic Boost (+20dB)',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined penum Playback channels: Mono Mono: Playback [off] so for aplay it seems I have no recording device, but for amixer I've got the mic, a mic boost and mic stereo as well with all those gorgeous stuffs on their place !!. If so, how could my Sound Recorder record the audio without any problem at all ?!?! For sure I'm giving the wrong device name to the command line for capturing audio but I'm loosing the hope for finding the correct one ! Please help....before I tear my hair out !!!

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  • What can I do to enhance MacBook Pro internal mic sound quality in iMovie?

    - by gaearon
    After using MacBook for two years, I bought 17'' MacBook Pro. I'm pretty happy with it, performance and all, but I also was going to record some music videos for YouTube. I play guitar and sing. However I was extremely disappointed with the sound quality that comes by default. I'm 100% sure my 13'' MacBook mic was much better at recording music and singing. Currently mic can't event handle acoustic guitar, outputting sound you'd think was recorded 5 years ago in ARM format on a Nokia phone on a loud concert. It totally feels like some lame filter is cutting low and high frequencies. I want to know what settings (visible or hidden) in iMovie or Mac OS itself I might want to tweak in order to get my MBP mic record clean sound.

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  • How do I fix a noisy input device (Internal Mic)? snd_hda_intel - debug included

    - by hazrpg
    As the title says, I'm having trouble with a very noisy audio input device - the internal mic or any plugged in mics. So far I've narrowed it down to an problem in ALSA since my debug info is showing a lot of "null" values. Can anyone help? Debug Info: http://www.alsa-project.org/db/?f=e0c6fb7e10624bf7691aa2b405cf0d3968e56c63 Exert from the debug: model : (null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null),(null)

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  • VBoxManage modifyhd --resize doesn't exist?

    - by George Korac
    I'm trying to increase the size of a VirtualBox Win7 .vdi disk on Ubuntu 10.04 but when I try executing VBoxManage modifyhd /path/disk.vdi --resize 15360 it returns Syntax error: unknown option: --resize. I'm unsure as to why this is happening because I've used it before and it's still listed under valid options for VBoxManage modifyhd in the VirualBox User Manual . Cheers, George @maniat1k george@george-laptop:~$ VBoxManage modifyhd '/home/george/.VirtualBox/HardDisks/Windows 7 64bit.vdi' --resize 15360 Sun VirtualBox Command Line Management Interface Version 3.1.6_OSE (C) 2005-2010 Sun Microsystems, Inc. All rights reserved. Usage: VBoxManage modifyhd | [--type normal|writethrough|immutable] [--autoreset on|off] [--compact] Syntax error: unknown option: --resize

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  • Core Audio on iPhone - any way to change the microphone gain (either for speakerphone mic or headpho

    - by Halle
    After much searching the answer seems to be no, but I thought I'd ask here before giving up. For a project I'm working on that includes recording sound, the input levels sound a little quiet both when the route is external mic + speaker and when it's headphone mic + headphones. Does anyone know definitively whether it is possible to programmatically change mic gain levels on the iPhone in any part of Core Audio? If not, is it possible that I'm not really in "speakerphone" mode (with the external mic at least) but only think I am? Here is my audio session init code: OSStatus error = AudioSessionInitialize(NULL, NULL, audioQueueHelperInterruptionListener, r); [...some error checking of the OSStatus...] UInt32 category = kAudioSessionCategory_PlayAndRecord; // need to play out the speaker at full volume too so it is necessary to change default route below error = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(category), &category); if (error) printf("couldn't set audio category!"); UInt32 doChangeDefaultRoute = 1; error = AudioSessionSetProperty (kAudioSessionProperty_OverrideCategoryDefaultToSpeaker, sizeof (doChangeDefaultRoute), &doChangeDefaultRoute); if (error) printf("couldn't change default route!"); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioRouteChange, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); UInt32 inputAvailable = 0; UInt32 size = sizeof(inputAvailable); error = AudioSessionGetProperty(kAudioSessionProperty_AudioInputAvailable, &size, &inputAvailable); if (error) printf("ERROR GETTING INPUT AVAILABILITY! %d\n", (int)error); error = AudioSessionAddPropertyListener(kAudioSessionProperty_AudioInputAvailable, audioQueueHelperPropListener, r); if (error) printf("ERROR ADDING AUDIO SESSION PROP LISTENER! %d\n", (int)error); error = AudioSessionSetActive(true); if (error) printf("AudioSessionSetActive (true) failed"); Thanks very much for any pointers.

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  • Building a project in VS that depends on a static and dynamic library

    - by fg nu
    Noob noobin'. I would appreciate some very careful handholding in setting up an example in Visual Studio 2010 Professional where I am trying to build a project which links: a previously built static library, for which the VS project folder is "C:\libjohnpaul\" a previously built dynamic library, for which the VS project folder is "C:\libgeorgeringo\" These are listed as Recipes 1.11, 1.12 and 1.13 in the C++ Cookbook. The project fails to compile for me with unresolved dependencies (see details below), and I can't figure out why. Project 1: Static Library The following are the header and source files that were compiled in this project. I was able to compile this project fine in VS2010, to the named standard library "libjohnpaul.lib" which lives in the folder ("C:/libjohnpaul/Release/"). // libjohnpaul/john.hpp #ifndef JOHN_HPP_INCLUDED #define JOHN_HPP_INCLUDED void john( ); // Prints "John, " #endif // JOHN_HPP_INCLUDED // libjohnpaul/john.cpp #include <iostream> #include "john.hpp" void john( ) { std::cout << "John, "; } // libjohnpaul/paul.hpp #ifndef PAUL_HPP_INCLUDED #define PAUL_HPP_INCLUDED void paul( ); // Prints " Paul, " #endif // PAUL_HPP_INCLUDED // libjohnpaul/paul.cpp #include <iostream> #include "paul.hpp" void paul( ) { std::cout << "Paul, "; } // libjohnpaul/johnpaul.hpp #ifndef JOHNPAUL_HPP_INCLUDED #define JOHNPAUL_HPP_INCLUDED void johnpaul( ); // Prints "John, Paul, " #endif // JOHNPAUL_HPP_INCLUDED // libjohnpaul/johnpaul.cpp #include "john.hpp" #include "paul.hpp" #include "johnpaul.hpp" void johnpaul( ) { john( ); paul( ); Project 2: Dynamic Library Here are the header and source files for the second project, which also compiled fine with VS2010, and the "libgeorgeringo.dll" file lives in the directory "C:\libgeorgeringo\Debug". // libgeorgeringo/george.hpp #ifndef GEORGE_HPP_INCLUDED #define GEORGE_HPP_INCLUDED void george( ); // Prints "George, " #endif // GEORGE_HPP_INCLUDED // libgeorgeringo/george.cpp #include <iostream> #include "george.hpp" void george( ) { std::cout << "George, "; } // libgeorgeringo/ringo.hpp #ifndef RINGO_HPP_INCLUDED #define RINGO_HPP_INCLUDED void ringo( ); // Prints "and Ringo\n" #endif // RINGO_HPP_INCLUDED // libgeorgeringo/ringo.cpp #include <iostream> #include "ringo.hpp" void ringo( ) { std::cout << "and Ringo\n"; } // libgeorgeringo/georgeringo.hpp #ifndef GEORGERINGO_HPP_INCLUDED #define GEORGERINGO_HPP_INCLUDED // define GEORGERINGO_DLL when building libgerogreringo.dll # if defined(_WIN32) && !defined(__GNUC__) # ifdef GEORGERINGO_DLL # define GEORGERINGO_DECL _ _declspec(dllexport) # else # define GEORGERINGO_DECL _ _declspec(dllimport) # endif # endif // WIN32 #ifndef GEORGERINGO_DECL # define GEORGERINGO_DECL #endif // Prints "George, and Ringo\n" #ifdef __MWERKS__ # pragma export on #endif GEORGERINGO_DECL void georgeringo( ); #ifdef __MWERKS__ # pragma export off #endif #endif // GEORGERINGO_HPP_INCLUDED // libgeorgeringo/ georgeringo.cpp #include "george.hpp" #include "ringo.hpp" #include "georgeringo.hpp" void georgeringo( ) { george( ); ringo( ); } Project 3: Executable that depends on the previous libraries Lastly, I try to link the aforecompiled static and dynamic libraries into one project called "helloBeatlesII" which has the project directory "C:\helloBeatlesII" (note that this directory does not nest the other project directories). The linking process that I did is described below: To the "helloBeatlesII" solution, I added the solutions "libjohnpaul" and "libgeorgeringo"; then I changed the properties of the "helloBeatlesII" project to additionally point to the include directories of the other two projects on which it depends ("C:\libgeorgeringo\libgeorgeringo" & "C:\libjohnpaul\libjohnpaul"); added "libgeorgeringo" and "libjohnpaul" to the project dependencies of the "helloBeatlesII" project and made sure that the "helloBeatlesII" project was built last. Trying to compile this project gives me the following unsuccessful build: 1------ Build started: Project: helloBeatlesII, Configuration: Debug Win32 ------ 1Build started 10/13/2012 5:48:32 PM. 1InitializeBuildStatus: 1 Touching "Debug\helloBeatlesII.unsuccessfulbuild". 1ClCompile: 1 helloBeatles.cpp 1ManifestResourceCompile: 1 All outputs are up-to-date. 1helloBeatles.obj : error LNK2019: unresolved external symbol "void __cdecl georgeringo(void)" (?georgeringo@@YAXXZ) referenced in function _main 1helloBeatles.obj : error LNK2019: unresolved external symbol "void __cdecl johnpaul(void)" (?johnpaul@@YAXXZ) referenced in function _main 1E:\programming\cpp\vs-projects\cpp-cookbook\helloBeatlesII\Debug\helloBeatlesII.exe : fatal error LNK1120: 2 unresolved externals 1 1Build FAILED. 1 1Time Elapsed 00:00:01.34 ========== Build: 0 succeeded, 1 failed, 2 up-to-date, 0 skipped ========== At this point I decided to call in the cavalry. I am new to VS2010, so in all likelihood I am missing something straightforward.

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  • No external microphone Acer AO722

    - by Leeghwater
    The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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  • How to play from mic to speakers in Ubuntu Karmic?

    - by vava
    Ok, I have a silly problem. I have few bluetooth headsets lying around and I want to make DYI baby monitor out of them. All I need is for some application to listen to the mic and send it to the speakers. Loopback (even though it doesn't work) is not good enough, it'll send sound from the mic to the speakers in the same device but I need it to go across devices. So does anyone know some application that can do that? I'm looking for something small and easy to use, not jackd or similar.

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  • Ubuntu 11.10 - How can i stop self-feedback-loop, coming directly from my microphone to speaker?

    - by YumYumYum
    I have microphone audio, which comes instantly to my speaker. I am using default pulseaudio and alsa from the package. I have tried to setup: 1) PA /etc/pulse/default.pa /etc/asound.conf $ ls analog-input-aux.conf analog-input-fm.conf analog-input-mic.conf analog-input-tvtuner.conf analog-output-desktop-speaker.conf analog-output-mono.conf analog-input.conf analog-input-front-mic.conf analog-input-mic.conf.common analog-input-video.conf analog-output-headphones-2.conf analog-output-speaker.conf analog-input.conf.common analog-input-internal-mic.conf analog-input-mic-line.conf analog-output.conf analog-output-headphones.conf iec958-stereo-output.conf analog-input-dock-mic.conf analog-input-linein.conf analog-input-rear-mic.conf analog-output.conf.common analog-output-lfe-on-mono.conf 2) ALSA in lsmod to make sure no loopback modules are loaded etc but none is resolving it. And there are many less information available on this. Has anyone similar problem solution in Ubuntu 11.10? (this problem i have resolved in Ubuntu 11.04 by replacing the default pulseaudio version to latest source from git, but while trying the same in Ubuntu 11.10 does not worked). Any tips please?

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  • Super-Charge GIMP’s Image Editing Capabilities with G’MIC [Cross-Platform]

    - by Asian Angel
    Recently we showed you how to enhance GIMP’s image editing power and today we help you super-charge GIMP even more. G’MIC (GREYC’s Magic Image Converter) will add an impressive array of filters and effects to your GIMP installation for image editing goodness. Note: We applied the Contrast Swiss Mask filter to the image shown in the screenshot above to create a nice, warm sunset effect. To add the new PPA open the Ubuntu Software Center, go to the Edit Menu, and select Software Sources. Access the Other Software Tab in the Software Sources Window and add the first of the PPAs shown below (outlined in red). The second PPA will be automatically added to your system. Once you have the new PPAs set up, go back to the Ubuntu Software Center and do a search for “G’MIC”. You will find two listings available and can select either one to add G’MIC to your system (both work equally well). Click on More Info for the listing that you choose and scroll down to where Add-ons are listed. Make sure to select the Add-on listed, click Apply Changes when it appears, and then click Install. We have both shown here for your convenience… When you get ready to use G’MIC to enhance an image, go to the Filters Menu and select G’MIC. A new window will appear where you can select from an impressive array of filters available for your use. Have fun! Command Line Installation For those of you who prefer using the command line for installation use the following commands: sudo add-apt-repository ppa:ferramroberto/gimp sudo apt-get update sudo apt-get install gmic gimp-gmic Links Note: G’MIC is available for Linux, Windows, and Mac. G’MIC PPA at Launchpad [via Web Upd8] G’MIC Homepage at Sourceforge *Downloads for all three platforms available here. Bonus The anime wallpaper shown in the screenshots above can be found here: anime sport [DesktopNexus] Latest Features How-To Geek ETC Learn To Adjust Contrast Like a Pro in Photoshop, GIMP, and Paint.NET Have You Ever Wondered How Your Operating System Got Its Name? Should You Delete Windows 7 Service Pack Backup Files to Save Space? What Can Super Mario Teach Us About Graphics Technology? Windows 7 Service Pack 1 is Released: But Should You Install It? How To Make Hundreds of Complex Photo Edits in Seconds With Photoshop Actions Access and Manage Your Ubuntu One Account in Chrome and Iron Mouse Over YouTube Previews YouTube Videos in Chrome Watch a Machine Get Upgraded from MS-DOS to Windows 7 [Video] Bring the Whole Ubuntu Gang Home to Your Desktop with this Mascots Wallpaper Hack Apart a Highlighter to Create UV-Reactive Flowers [Science] Add a “Textmate Style” Lightweight Text Editor with Dropbox Syncing to Chrome and Iron

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