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  • Windows XP-64 loses audio sounds, drive letters... why?

    - by Ira Baxter
    Until sometime in early December, I had a wonderfully functioning XP-64 system. It was configured to auto download/install MS patches. I occassionally update the software on it, e.g. Open Office, Adobe Reader, Skype, but I don't fetch hundreds of tools or anything much beyond what I just mentioned. In December, suddenly my audio stopped, and drive letters assigned to various mount points on other machines quit being available. Apparantly, the services that support these (and some others) are now not starting up when I boot/login. There isn't anything obvious in the event log. If I manually restart the associated services, these facilities come back on line and work for awhile (a day) but pretty soon the problem reappears. I don't reboot very often, nor do I log out out much. Hints?

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  • How to send Bluetooth audio to non-Bluetooth speakers?

    - by wonsungi
    In short, I am looking for a Bluetooth-3.5 mini stereo converter. What is this type of device called, and what are some of the best models (is there a difference in audio quality/lag)? I wish to connect some speakers (Altec Lansing inMotion IM7), which does not support Bluetooth, to my laptop (Lenovo X301) wirelessly. Currently, I can connect my laptop's headphone jack to the AUX jack on my speakers via a mini stereo cable. How do I replace this cable with some type of Bluetooth setup? I am not sure what this Bluetooth device is called. I thought I found something, but it actually does the opposite of what I need (3.5 mini stereo-Bluetooth). (My OS is Vista Enterprise, if that matters)

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  • No audio driver for ECS 7050-M v1.0a for Windows 7?

    - by NudeRaider
    I have an Elitegroup 7050-M v1.0a motherboard and I've used the Realtek HD audio driver it came with for years under Windows XP. There was a nice mixer panel, an equalizer and sound profiles that added effects. All of these menus are gone now that I've moved on to Windows 7 64 Bit. All I have is the basic settings under system control - sound, but they are not nearly as detailed. To clarify: Sound works, but the advanced settings I got when using older OS's are gone. Older drivers don't seem to work under Win7 and I can't find one for Win7. So how can I fully utilize the capabilities of my HD sound chip under Win7?

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  • Steganography : Encoded audio and video file not being played, getting corrupted. What is the issue

    - by Shantanu Gupta
    I have made a steganography program to encrypt/Decrypt some text under image audio and video. I used image as bmp(54 byte header) file, audio as wav(44 byte header) file and video as avi(56 byte header) file formats. When I tries to encrypt text under all these file then it gets encrypted successfully and are also getting decrypted correctly. But it is creating a problem with audio and video i.e these files are not being played after encrypted result. What can be the problem. I am working on Turbo C++ compiler. I know it is super outdated compiler but I have to do it in this only. Here is my code to encrypt. int Binary_encode(char *txtSourceFileName, char *binarySourceFileName, char *binaryTargetFileName,const short headerSize) { long BinarySourceSize=0,TextSourceSize=0; char *Buffer; long BlockSize=10240, i=0; ifstream ReadTxt, ReadBinary; //reads ReadTxt.open(txtSourceFileName,ios::binary|ios::in);//file name, mode of open, here input mode i.e. read only if(!ReadTxt) { cprintf("\nFile can not be opened."); return 0; } ReadBinary.open(binarySourceFileName,ios::binary|ios::in);//file name, mode of open, here input mode i.e. read only if(!ReadBinary) { ReadTxt.close();//closing opened file cprintf("\nFile can not be opened."); return 0; } ReadBinary.seekg(0,ios::end);//setting pointer to a file at the end of file. ReadTxt.seekg(0,ios::end); BinarySourceSize=(long )ReadBinary.tellg(); //returns the position of pointer TextSourceSize=(long )ReadTxt.tellg(); //returns the position of pointer ReadBinary.seekg(0,ios::beg); //sets the pointer to the begining of file ReadTxt.seekg(0,ios::beg); //sets the pointer to the begining of file if(BinarySourceSize<TextSourceSize*50) //Minimum size of an image should be 50 times the size of file to be encrypted { cout<<"\n\n"; cprintf("Binary File size should be bigger than text file size."); ReadBinary.close(); ReadTxt.close(); return 0; } cout<<"\n"; cprintf("\n\nSize of Source Image/Audio File is : "); cout<<(float)BinarySourceSize/1024; cprintf("KB"); cout<<"\n"; cprintf("Size of Text File is "); cout<<TextSourceSize; cprintf(" Bytes"); cout<<"\n"; getch(); //write header to file without changing else file will not open //bmp image's header size is 53 bytes Buffer=new char[headerSize]; ofstream WriteBinary; // writes to file WriteBinary.open(binaryTargetFileName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists ReadBinary.read(Buffer,headerSize);//reads no of bytes and stores them into mem, size contains no of bytes in a file WriteBinary.write(Buffer,headerSize);//writes header to 2nd image delete[] Buffer;//deallocate memory /* Buffer = new char[sizeof(long)]; Buffer = (char *)(&TextSourceSize); cout<<Buffer; */ WriteBinary.write((char *)(&TextSourceSize),sizeof(long)); //writes no of byte to be written in image immediate after header ends //to decrypt file if(!(Buffer=new char[TextSourceSize])) { cprintf("Enough Memory could not be assigned."); return 0; } ReadTxt.read(Buffer,TextSourceSize);//read all data from text file ReadTxt.close();//file no more needed WriteBinary.write(Buffer,TextSourceSize);//writes all text file data into image delete[] Buffer;//deallocate memory //replace Tsize+1 below with Tsize and run the program to see the change //this is due to the reason that 50-54 byte no are of colors which we will be changing ReadBinary.seekg(TextSourceSize+1,ios::cur);//move pointer to the location-current loc i.e. 53+content of text file //write remaining image content to image file while(i<BinarySourceSize-headerSize-TextSourceSize+1) { i=i+BlockSize; Buffer=new char[BlockSize]; ReadBinary.read(Buffer,BlockSize);//reads no of bytes and stores them into mem, size contains no of bytes in a file WriteBinary.write(Buffer,BlockSize); delete[] Buffer; //clear memory, else program can fail giving correct output } ReadBinary.close(); WriteBinary.close(); //Encoding Completed return 0; } Code to decrypt int Binary_decode(char *binarySourceFileName, char *txtTargetFileName, const short headerSize) { long TextDestinationSize=0; char *Buffer; long BlockSize=10240; ifstream ReadBinary; ofstream WriteText; ReadBinary.open(binarySourceFileName,ios::binary|ios::in);//file will be appended if(!ReadBinary) { cprintf("File can not be opened"); return 0; } ReadBinary.seekg(headerSize,ios::beg); Buffer=new char[4]; ReadBinary.read(Buffer,4); TextDestinationSize=*((long *)Buffer); delete[] Buffer; cout<<"\n\n"; cprintf("Size of the File that will be created is : "); cout<<TextDestinationSize; cprintf(" Bytes"); cout<<"\n\n"; sleep(1); WriteText.open(txtTargetFileName,ios::binary|ios::out|ios::trunc);//file will be created if not exists else truncate its data while(TextDestinationSize>0) { if(TextDestinationSize<BlockSize) BlockSize=TextDestinationSize; Buffer= new char[BlockSize]; ReadBinary.read(Buffer,BlockSize); WriteText.write(Buffer,BlockSize); delete[] Buffer; TextDestinationSize=TextDestinationSize-BlockSize; } ReadBinary.close(); WriteText.close(); return 0; } int text_encode(char *SourcefileName, char *DestinationfileName) { ifstream fr; //reads ofstream fw; // writes to file char c; int random; clrscr(); fr.open(SourcefileName,ios::binary);//file name, mode of open, here input mode i.e. read only if(!fr) { cprintf("File can not be opened."); getch(); return 0; } fw.open(DestinationfileName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists while(fr) { int i; while(fr!=0) { fr.get(c); //reads a character from file and increments its pointer char ch; ch=c; ch=ch+1; fw<<ch; //appends character in c to a file } } fr.close(); fw.close(); return 0; } int text_decode(char *SourcefileName, char *DestinationName) { ifstream fr; //reads ofstream fw; // wrrites to file char c; int random; clrscr(); fr.open(SourcefileName,ios::binary);//file name, mode of open, here input mode i.e. read only if(!fr) { cprintf("File can not be opened."); return 0; } fw.open(DestinationName,ios::binary|ios::out|ios::trunc);//file will be created or truncated if already exists while(fr) { int i; while(fr!=0) { fr.get(c); //reads a character from file and increments its pointer char ch; ch=c; ch=ch-1; fw<<ch; //appends character in c to a file } } fr.close(); fw.close(); return 0; }

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  • How can I make my Perl Jabber bot an event-driven program?

    - by TheGNUGuy
    I'm trying to make a Jabber bot and I am having trouble keeping it running while waiting for messages. How do I get my script to continuously run? I have tried calling a subroutine that has a while loop that I, in theory, have set up to check for any messages and react accordingly but my script isn't behaving that way. Here is my source: http://pastebin.com/03Habbvh # set jabber bot callbacks $jabberBot-SetMessageCallBacks(chat=\&chat); $jabberBot-SetPresenceCallBacks(available=\&welcome,unavailable=\&killBot); $jabberBot-SetCallBacks(receive=\&prnt,iq=\&gotIQ); $jabberBot-PresenceSend(type="available"); $jabberBot-Process(1); sub welcome { print "Welcome!\n"; $jabberBot-MessageSend(to=$jbrBoss-GetJID(),subject="",body="Hello There!",type="chat",priority=10); &keepItGoing; } sub prnt { print $_[1]."\n"; } #$jabberBot-MessageSend(to=$jbrBoss-GetJID(),subject="",body="Hello There! Global...",type="chat",priority=10); #$jabberBot-Process(5); #&keepItGoing; sub chat { my ($sessionID,$msg) = @_; $dump-pl2xml($msg); if($msg-GetType() ne 'get' && $msg-GetType() ne 'set' && $msg-GetType() ne '') { my $jbrCmd = &trimSpaces($msg-GetBody()); my $dbQry = $dbh-prepare("SELECT command,acknowledgement FROM commands WHERE message = '".lc($jbrCmd)."'"); $dbQry-execute(); if($dbQry-rows() 0 && $jbrCmd !~ /^insert/si) { my $ref = $dbQry-fetchrow_hashref(); $dbQry-finish(); $jabberBot-MessageSend(to=$msg-GetFrom(),subject="",body=$ref-{'acknowledgement'},type="chat",priority=10); eval $ref-{'command'}; &keepItGoing; } else { $jabberBot-MessageSend(to=$msg-GetFrom(),subject="",body="I didn't understand you!",type="chat",priority=10); $dbQry-finish(); &keepItGoing; } } } sub gotIQ { print "iq\n"; } sub trimSpaces { my $string = $_[0]; $string =~ s/^\s+//; #remove leading spaces $string =~ s/\s+$//; #remove trailing spaces return $string; } sub keepItGoing { print "keepItGoing!\n"; my $proc = $jabberBot-Process(1); while(defined($proc) && $proc != 1) { $proc = $jabberBot-Process(1); } } sub killBot { print "killing\n"; $jabberBot-MessageSend(to=$_[0]-GetFrom(),subject="",body="Logging Out!",type="chat",priority=10); $jabberBot-Process(1); $jabberBot-Disconnect(); exit; }

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  • C sockets, chat server and client, problem echoing back.

    - by wretrOvian
    Hi This is my chat server : #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <sys/types.h> #include <sys/socket.h> #include <netdb.h> #include <string.h> #define LISTEN_Q 20 #define MSG_SIZE 1024 struct userlist { int sockfd; struct sockaddr addr; struct userlist *next; }; int main(int argc, char *argv[]) { // declare. int listFD, newFD, fdmax, i, j, bytesrecvd; char msg[MSG_SIZE], ipv4[INET_ADDRSTRLEN]; struct addrinfo hints, *srvrAI; struct sockaddr_storage newAddr; struct userlist *users, *uptr, *utemp; socklen_t newAddrLen; fd_set master_set, read_set; // clear sets FD_ZERO(&master_set); FD_ZERO(&read_set); // create a user list users = (struct userlist *)malloc(sizeof(struct userlist)); users->sockfd = -1; //users->addr = NULL; users->next = NULL; // clear hints memset(&hints, 0, sizeof hints); // prep hints hints.ai_family = AF_INET; hints.ai_socktype = SOCK_STREAM; hints.ai_flags = AI_PASSIVE; // get srver info if(getaddrinfo("localhost", argv[1], &hints, &srvrAI) != 0) { perror("* ERROR | getaddrinfo()\n"); exit(1); } // get a socket if((listFD = socket(srvrAI->ai_family, srvrAI->ai_socktype, srvrAI->ai_protocol)) == -1) { perror("* ERROR | socket()\n"); exit(1); } // bind socket bind(listFD, srvrAI->ai_addr, srvrAI->ai_addrlen); // listen on socket if(listen(listFD, LISTEN_Q) == -1) { perror("* ERROR | listen()\n"); exit(1); } // add listfd to master_set FD_SET(listFD, &master_set); // initialize fdmax fdmax = listFD; while(1) { // equate read_set = master_set; // run select if(select(fdmax+1, &read_set, NULL, NULL, NULL) == -1) { perror("* ERROR | select()\n"); exit(1); } // query all sockets for(i = 0; i <= fdmax; i++) { if(FD_ISSET(i, &read_set)) { // found active sockfd if(i == listFD) { // new connection // accept newAddrLen = sizeof newAddr; if((newFD = accept(listFD, (struct sockaddr *)&newAddr, &newAddrLen)) == -1) { perror("* ERROR | select()\n"); exit(1); } // resolve ip if(inet_ntop(AF_INET, &(((struct sockaddr_in *)&newAddr)->sin_addr), ipv4, INET_ADDRSTRLEN) == -1) { perror("* ERROR | inet_ntop()"); exit(1); } fprintf(stdout, "* Client Connected | %s\n", ipv4); // add to master list FD_SET(newFD, &master_set); // create new userlist component utemp = (struct userlist*)malloc(sizeof(struct userlist)); utemp->next = NULL; utemp->sockfd = newFD; utemp->addr = *((struct sockaddr *)&newAddr); // iterate to last node for(uptr = users; uptr->next != NULL; uptr = uptr->next) { } // add uptr->next = utemp; // update fdmax if(newFD > fdmax) fdmax = newFD; } else { // existing sockfd transmitting data // read if((bytesrecvd = recv(i, msg, MSG_SIZE, 0)) == -1) { perror("* ERROR | recv()\n"); exit(1); } msg[bytesrecvd] = '\0'; // find out who sent? for(uptr = users; uptr->next != NULL; uptr = uptr->next) { if(i == uptr->sockfd) break; } // resolve ip if(inet_ntop(AF_INET, &(((struct sockaddr_in *)&(uptr->addr))->sin_addr), ipv4, INET_ADDRSTRLEN) == -1) { perror("* ERROR | inet_ntop()"); exit(1); } // print fprintf(stdout, "%s\n", msg); // send to all for(j = 0; j <= fdmax; j++) { if(FD_ISSET(j, &master_set)) { if(send(j, msg, strlen(msg), 0) == -1) perror("* ERROR | send()"); } } } // handle read from client } // end select result handle } // end looping fds } // end while return 0; } This is my client: #include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <sys/types.h> #include <sys/socket.h> #include <netdb.h> #include <string.h> #define MSG_SIZE 1024 int main(int argc, char *argv[]) { // declare. int newFD, bytesrecvd, fdmax; char msg[MSG_SIZE]; fd_set master_set, read_set; struct addrinfo hints, *srvrAI; // clear sets FD_ZERO(&master_set); FD_ZERO(&read_set); // clear hints memset(&hints, 0, sizeof hints); // prep hints hints.ai_family = AF_INET; hints.ai_socktype = SOCK_STREAM; hints.ai_flags = AI_PASSIVE; // get srver info if(getaddrinfo(argv[1], argv[2], &hints, &srvrAI) != 0) { perror("* ERROR | getaddrinfo()\n"); exit(1); } // get a socket if((newFD = socket(srvrAI->ai_family, srvrAI->ai_socktype, srvrAI->ai_protocol)) == -1) { perror("* ERROR | socket()\n"); exit(1); } // connect to server if(connect(newFD, srvrAI->ai_addr, srvrAI->ai_addrlen) == -1) { perror("* ERROR | connect()\n"); exit(1); } // add to master, and add keyboard FD_SET(newFD, &master_set); FD_SET(STDIN_FILENO, &master_set); // initialize fdmax if(newFD > STDIN_FILENO) fdmax = newFD; else fdmax = STDIN_FILENO; while(1) { // equate read_set = master_set; if(select(fdmax+1, &read_set, NULL, NULL, NULL) == -1) { perror("* ERROR | select()"); exit(1); } // check server if(FD_ISSET(newFD, &read_set)) { // read data if((bytesrecvd = recv(newFD, msg, MSG_SIZE, 0)) < 0 ) { perror("* ERROR | recv()"); exit(1); } msg[bytesrecvd] = '\0'; // print fprintf(stdout, "%s\n", msg); } // check keyboard if(FD_ISSET(STDIN_FILENO, &read_set)) { // read data from stdin if((bytesrecvd = read(STDIN_FILENO, msg, MSG_SIZE)) < 0) { perror("* ERROR | read()"); exit(1); } msg[bytesrecvd] = '\0'; // send if((send(newFD, msg, bytesrecvd, 0)) == -1) { perror("* ERROR | send()"); exit(1); } } } return 0; } The problem is with the part where the server recv()s data from an FD, then tries echoing back to all [send() ]; it just dies, w/o errors, and my client is left looping :(

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  • project hierarchy

    - by Noona
    Is there a difference between a package and a folder in eclipse? for example, if I have this hierarchy requirement: java –classpath C:\ChatCompany\BackendChatServer\ -Djava.security.policy=c:\HW2\permissions.policy hw2.chat.backend.main.ChatBackendServer when the package's name is: "hw2.chat.backend.main" and "ChatCompany\BackendChatServer\" is the folder name, then how can I make this separation between a package and a folder in eclipse, so that I can write "package hw2.chat.backend.main" and not "package ChatCompany.BackendChatServer.hw2.chat.backend.main"? thanks

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  • xvidcap: Error accessing sound input from /dev/dsp

    - by stivlo
    I'm running Ubuntu 11.10 and I'm trying xvidcap to record a screencast with audio from the microphone, however it can't record any sound: $ xvidcap --file appo.avi --cap_geometry 700x500-0+0 Error accessing sound input from /dev/dsp Sound disabled! Sure enough /dev/dsp doesn't even exist: $ sudo ls -lh /dev/dsp ls: cannot access /dev/dsp: No such file or directory I found a blog post about fixing xvidcap sound input, however if I try the suggestion I get: $ sudo modprobe snd-pcm-oss FATAL: Module snd_pcm_oss not found. So the question is, how can I create /dev/dsp? The problem behind the problem is: how can I record sound from the microphone with xvidcap? So workarounds are welcome too. UPDATE: I've followed the suggestion of James, and something has improved. The error accessing /dev/dsp is gone, however now I get: [oss @ 0x8e0c120] Estimating duration from bitrate, this may be inaccurate xtoffmpeg.c add_audio_stream(): Can't initialize fifo for audio recording Now when I record xvidcap appears in the recording tab of pavucontrol and I can choose Audio stream from Internal Audio Analog Stereo or Monitor of Internal Audio Analog Stereo, I tried both just in case, but the video is still mute. UPDATE 2: I found that "Monitor of" is the one to record application sounds, while for microphone, I should choose "Internal Audio Analog Stereo". To rule out other problems, such as with the microphone, I tried with gnome-sound-recorder and it works. Actually I jumped on my chair, since the volume was too high! :-)

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  • HTML5 media loading sometimes suspends or aborts: misconfigured Apache?

    - by Joan Botella
    Recently, some code that has been working fine for months started to run unexpectedly. That code is just a media files loading JavaScript function, that uses jQuery. It's pretty long, but in essence it is like this: var $audio=$('<audio>'); $audio.on('canplaythrough',function(e){ $audio[0].play(); }); $audio.attr('src','song.ogg'); Basically, the file only loads sometimes, and sometimes stops loading with a suspend or even an abort event. I have uploaded a little testing HTML to http://www.joanbotella.com/tests/loading , where you can see what's happening. You can download the test files from http://www.joanbotella.com/tests/loading/loadingTest.zip for local testing. I have just checked that opening the test index.html file directly into Firefox, and not through my localhost Apache server, makes the audio files perfectly playable. So, I assume, my hosting and I have the Apache server misconfigured for serving media files. My software versions are: Apache 2.2.22-1ubuntu1.7 , Mozilla Firefox 31.0 , Chromium 36.0.1985.125 and jQuery 1.11.0. Can you help me? Thanks in advance!

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  • Will proprietary software-based sound enhancements work with Ubuntu? (BeatsAudio, Dolby)

    - by LiveWireBT
    This question is targeted at mainstream or gamer-grade software-based audio/sound enhancements, found in highly integrated computing and entertainment systems like laptops, tablets and smartphones. These are mostly marketed with fancy badges of known audio-releated brands on the product or packaging, while being mostly uncertain about the actual implementation or components used and poorly differentiated from the general audio capabilities of the system or device. This question is not about actual hardware like speakers. If your headphones are not properly detected, your speakers are assigned wrong, work partially or not at all then your soundcard or chip is not properly detected and you should take a look at troubleshooting audio issues. This question is also not about enthusiast or recording-grade hardware like recording interfaces, amplifiers and DACs in a variety of formfactors. And this question is also not about audio encoding and playback of different audio formats like Dolby Digital, Dolby TrueHD and DTS. Most of these may be subject to patents and licensing, see restricted formats. If you are just searching for an equalizer, please take a look at this question: Is there any Sound enhancers/equalizer? Simply speaking: Every feature where you would flip a switch or check a box in a fancy looking interface in Windows that makes the sound change from neutral to fancy.

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  • I can't hear any sounds on ubuntu 11.10 on Dell inspiron N5010

    - by Ahmed
    I have a problem that I can't hear any sounds and I don't know where to start. I did the following : lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 5 Series/3400 Series Chipset High Definition Audio (rev 06) Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 48 Memory at fbf00000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel -- 01:00.1 Audio device: ATI Technologies Inc Manhattan HDMI Audio [Mobility Radeon HD 5000 Series] Subsystem: Dell Device 0447 Flags: bus master, fast devsel, latency 0, IRQ 49 Memory at fbe40000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel And It seems that I have 2 soundcards. Is that normal ?? I also did this: aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Generic [HD-Audio Generic], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 Also on the sound setting GUI. I have 2 hardware profiles for sound cards but none of them works when I test the speakers. Where should I start searching ?

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  • Writing a Makefile.am to invoke googletest unit tests

    - by jmglov
    I am trying to add my first unit test to an existing Open Source project. Specifically, I added a new class, called audio_manager: src/audio/audio_manager.h src/audio/audio_manager.cc I created a src/test directory structure that mirrors the structure of the implementation files, and wrote my googletest unit tests: src/test/audio/audio_manager.cc Now, I am trying to set up my Makefile.am to compile and run the unit test: src/test/audio/Makefile.am I copied Makefile.am from: src/audio/Makefile.am Does anyone have a simple recipe for me, or is it to the cryptic automake documentation for me? :)

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  • How to remove music/videos DRM protection and convert to Mobile Devices such as iPod, iPhone, PSP, Z

    - by tonywesley
    The music/video files you purchased from online music stores like iTunes, Yahoo Music or Wal-Mart are under DRM protection. So you can't convert them to the formats supported by your own mobile devices such as Nokia phone, Creative Zen palyer, iPod, PSP, Walkman, Zune… You also can't share your purchased music/videos with your friends. The following step by step tutorial is dedicated to instructing music lovers to how to convert your DRM protected music/videos to mobile devices. Method 1: If you only want to remove DRM protection from your protected music, this method will not spend your money. Step 1: Burn your protected music files to CD-R/RW disc to make an audio CD Step 2: Find a free CD Ripper software to convert the audio CD track back to MP3, WAV, WMA, M4A, AAC, RA… Method 2: This guide will show you how to crack drm from protected wmv, wma, m4p, m4v, m4a, aac files and convert to unprotected WMV, MP4, MP3, WMA or any video and audio formats you like, such as AVI, MP4, Flv, MPEG, MOV, 3GP, m4a, aac, wmv, ogg, wav... I have been using Media Converter software, it is the quickest and easiest solution to remove drm from WMV, M4V, M4P, WMA, M4A, AAC, M4B, AA files by quick recording. It gets audio and video stream at the bottom of operating system, so the output quality is lossless and the conversion speed is fast . The process is as follows. Step 1: Download and install the software Step 2: Run the software and click "Add…" button to load WMA or M4A, M4B, AAC, WMV, M4P, M4V, ASF files Step 3: Choose output formats. If you want to convert protected audio files, please select "Convert audio to" list; If you want to convert protected video files, please select "Convert video to" list. Step 4: You can click "Settings" button to custom preference for output files. Click "Settings" button bellow "Convert audio to" list for protected audio files Click "Settings" button bellow "Convert video to" list for protected video files Step 5: Start remove DRM and convert your DRM protected music and videos by click on "Start" button. What is DRM? DRM, which is most commonly found in movies and music files, doesn't mean just basic copy-protection of video, audio and ebooks, but it basically means full protection for digital content, ranging from delivery to end user's ways to use the content. We can remove the Drm from video and audio files legally by quick recording.

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  • CodePlex Daily Summary for Friday, March 12, 2010

    CodePlex Daily Summary for Friday, March 12, 2010New Projects.NET DEPENDENCY INJECTION: Abel Perez Enterprise FrameworkAutodocs - WCF REST Automatic API Documentation Generator: Autodocs is an automatic API documentation generator for .NET applications that use Windows Communication Foundation (WCF) to establish REST API's.BlockBlock: Block Block is a free game. You know Lumines and you will like BlockBlock.C4F XNA ASCII Post-Processing: This is the source code for the Coding4Fun article "XNA Effects – ASCII Art in 3D"ChequePrinter: this is ChequePrinterCompiladores MSIL usando Phoenix (PLP 2008.1 - CIn/UFPE): Este projeto foi feito com o intuito de explorar a plataforma Microsoft Phoenix para a construção de compiladores para MSIL de duas linguagens de E...CRM External View: CRM External View enables more robust control over exposing Microsoft CRM data (in a form of views) for external parties. The solution uses web ser...CS Project2: This is for the projectDotNetNuke IM Module of Facebook Like Messenger: Help you integrate 123 Web Messenger into DotNetNuke, and add a powerful 1-to-1 IM Software named "Facebook Messenger Style Web Chat Bar" at the bo...DotNetNuke® RadPanelBar: DNNRadPanelBar makes it easy to add telerik RadPanelBar functionality to your module or skin. Licensing permits anyone to use the components (incl...DotNetNuke® Skin Blocks: A DotNetNuke Design Challenge skin package submitted to the "Modern Business" category by Armand Datema of Schwingsoft. This skin uses a bit of jQu...Drilltrough and filtering on SSAS-cubes in SSRS: We will describe a technique to create Reporting services (SSRS) reports that use Analysis services (SSAS) cubes as data sources, have a very intu...Ecosystem Diagnosis & Treatment: The Ecosystem DIagnosis & Treatment community provides tools, analyses and applications of the medical model to natural resource problems. 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It should be considered a beta test version as it has not been tested for very long on my device.Most Popular ProjectsMetaSharpWBFS ManagerRawrAJAX Control ToolkitMicrosoft SQL Server Product Samples: DatabaseSilverlight ToolkitWindows Presentation Foundation (WPF)ASP.NET Ajax LibraryASP.NETMicrosoft SQL Server Community & SamplesMost Active ProjectsUmbraco CMSRawrN2 CMSBlogEngine.NETFasterflect - A Fast and Simple Reflection APIjQuery Library for SharePoint Web Servicespatterns & practices – Enterprise LibraryFarseer Physics EngineCaliburn: An Application Framework for WPF and SilverlightSharePoint Team-Mailer

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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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  • Asterisk server firewall script allows 2-way audio from incoming calls, but not on outgoing?

    - by cappie
    I'm running an Asterisk PBX on a virtual machine directly connected to the Internet and I really want to prevent script kiddies, l33t h4x0rz and actual hackers access to my server. The basic way I protect my calling-bill now is by using 32 character passwords, but I would much rather have a way to protect The firewall script I'm currently using is stated below, however, without the established connection firewall rule (mentioned rule #1), I cannot receive incoming audio from the target during outgoing calls: #!/bin/bash # first, clean up! iptables -F iptables -X iptables -t nat -F iptables -t nat -X iptables -t mangle -F iptables -t mangle -X iptables -P INPUT ACCEPT iptables -P FORWARD DROP # we're not a router iptables -P OUTPUT ACCEPT # don't allow invalid connections iptables -A INPUT -m state --state INVALID -j DROP # always allow connections that are already set up (MENTIONED RULE #1) iptables -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT # always accept ICMP iptables -A INPUT -p icmp -j ACCEPT # always accept traffic on these ports #iptables -A INPUT -p tcp --dport 80 -j ACCEPT iptables -A INPUT -p tcp --dport 22 -j ACCEPT # always allow DNS traffic iptables -A INPUT -p udp --sport 53 -j ACCEPT iptables -A OUTPUT -p udp --dport 53 -j ACCEPT # allow return traffic to the PBX iptables -A INPUT -p udp -m udp --dport 50000:65536 -j ACCEPT iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT iptables -A INPUT -p udp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -p tcp --destination-port 5060:5061 -j ACCEPT iptables -A INPUT -m multiport -p udp --dports 10000:20000 iptables -A INPUT -m multiport -p tcp --dports 10000:20000 # IP addresses of the office iptables -A INPUT -s 95.XXX.XXX.XXX/32 -j ACCEPT # accept everything from the trunk IP's iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT iptables -A INPUT -s 195.XXX.XXX.XXX/32 -j ACCEPT # accept everything on localhost iptables -A INPUT -i lo -j ACCEPT # accept all outgoing traffic iptables -A OUTPUT -j ACCEPT # DROP everything else #iptables -A INPUT -j DROP I would like to know what firewall rule I'm missing for this all to work.. There is so little documentation on which ports (incoming and outgoing) asterisk actually needs.. (return ports included). Are there any firewall/iptables specialists here that see major problems with this firewall script? It's so frustrating not being able to find a simple firewall solution that enabled me to have a PBX running somewhere on the Internet which is firewalled in such a way that it can ONLY allows connections from and to the office, the DNS servers and the trunk(s) (and only support SSH (port 22) and ICMP traffic for the outside world). Hopefully, using this question, we can solve this problem once and for all.

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  • Using the Onboard VGA output with a PCIe video card. Both nVidia

    - by sebikul
    I have 2 video cards, one On board, a nVidia 6150SE nForce 430 and a PCIe nVidia GeForce GT 220 1GB DDR2 RAM I have already configured the PCIe card to use the dual monitor feature, using the VGA and HDMI ports, but now I want to add a third monitor, using the On board VGA port I have managed to enable the On board graphics processor, which is taking 400MB of ram, but I cant manage to use it, nvidia-settings does not detect it, like it's not usable (but is there) My questions are the following: How can I manage to get the On board VGA display to work together with the PCIe graphics card? If possible, how can I recover those 400 MB the on board card is taking (even without being used) or how can I get it to use the PCIe card available memory? System Details: Linux 2.6.35-28-generic i686 Ubuntu 10.10 (All updates installed) NVIDIA Driver Version: 260.19.06 (Official) If more info is needed please let me know. Here is the lspci output when the On board card is disabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) And this is when both are enabled: 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a3) 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) 00:0d.0 VGA compatible controller: nVidia Corporation C61 [GeForce 6150SE nForce 430] (rev a2) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:09.0 Ethernet controller: Intel Corporation 82557/8/9/0/1 Ethernet Pro 100 (rev 08) 02:00.0 VGA compatible controller: nVidia Corporation GT216 [GeForce GT 220] (rev a2) 02:00.1 Audio device: nVidia Corporation High Definition Audio Controller (rev a1) Output of lshw -class display: *-display description: VGA compatible controller product: GT216 [GeForce GT 220] vendor: nVidia Corporation physical id: 0 bus info: pci@0000:02:00.0 version: a2 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:18 memory:df000000-dfffffff memory:c0000000-cfffffff memory:da000000-dbffffff ioport:ef80(size=128) memory:def80000-deffffff *-display description: VGA compatible controller product: C61 [GeForce 6150SE nForce 430] vendor: nVidia Corporation physical id: d bus info: pci@0000:00:0d.0 version: a2 width: 64 bits clock: 66MHz capabilities: pm msi vga_controller bus_master cap_list rom configuration: driver=nvidia latency=0 resources: irq:22 memory:dd000000-ddffffff memory:b0000000-bfffffff memory:dc000000-dcffffff memory:deb40000-deb5ffff If what I'm looking for is not possible, please tell me, so I can disable the On board card and recover those 400MB of wasted RAM Thanks for your help!

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  • How can I synchronize a text with audio/sound in XNA/XACT?

    - by Omkar
    Hello Geeks, I wanted to display the text while sound is playing at background. In short if there is sound/audio for "What is this", I want to display the text "What is this" in text box synchronously. Is this possible with XNA/XACT? and can I use this in standard C# based WPF or Silverlight applications? Appreciating your help.

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  • How to stream audio from ASP.NET MVC controller when it's still encoding?

    - by kyrisu
    Background I have wave files on my server that I want to stream. Because of the size I want to encode them to mp3. I've tried to use FileStreamResult - but it doesn't work because as soon as program leaves the controller stream is closed and I get - "Cannot access a closed stream" FileContentResult - but it's not a stream and the user would need to wait for encoding to finish Question Is there a way to stream audio from the controller while it's still encoding?

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  • MessageListener didnt receive full message ASMACK Android

    - by Frank Junior
    i got problem when want to receive message, right now i am able to receive message, but some attribut is missing class MyMessageListener implements MessageListener { @Override public void processMessage(Chat chat, Message message) { Util.DebugLog("message->"+message.toXmlns()); } } what i got is <message to="[email protected]" type="chat" from="[email protected]/ff3b2485"><body asdf="asdf">aaa</body></message> talk_id and chat type inside message is missing. This is want i want when receive message <message to="[email protected]" type="chat" talk_id="304" chat_type="0" from="[email protected]/ff3b2485"><body asdf="asdf">aaa</body></message>

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  • convert decrypted .vobs to .avi with ffmpeg on ubuntu

    - by Arcath
    I have a .vob file that has bee ripped from a dvd, when I watch the .vob its very good quality video and 5.1 english audio but when I use ffmpeg it has rubbish video and mono french audio. That was using this command: ffmpeg -i /samba/ripping/vobs/12161840#2.vob -f avi /samba/ripping/avis/test.avi I've tried a few different variations on that but it never comes back with anything good just bigger files with bad video and incorrect sound. I know the videos good and the correct audio streams exist so how do I select a 5.1 track and get good video? ffmpeg gives the .vob details as: Input #0, mpeg, from '/samba/ripping/vobs/12161840#2.vob': Duration: 00:42:05.56, start: 0.287267, bitrate: 5738 kb/s Stream #0.0[0x1e0]: Video: mpeg2video, yuv420p, 720x576 [PAR 64:45 DAR 16:9], 8436 kb/s, 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0.1[0x80]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.2[0x81]: Audio: ac3, 48000 Hz, 5.1, s16, 384 kb/s Stream #0.3[0x82]: Audio: ac3, 48000 Hz, mono, s16, 192 kb/s Output #0, avi, to '/samba/ripping/avis/test.avi': Metadata: ISFT : Lavf52.64.2 Stream #0.0: Video: mpeg4, yuv420p, 720x576 [PAR 64:45 DAR 16:9], q=2-31, 200 kb/s, 25 tbn, 25 tbc Stream #0.1: Audio: mp2, 48000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.3 -> #0.1

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  • how to implement login and service features as in skype or msn chat in a wpf application

    - by black sensei
    Hello Good people! I'm building an WPF application that connect to web services for its operations.Things that i needed to be working are so far fine.Now i'll like to improve use experience by adding features like username editable combobox, sign me in when skype start and start when computer start. I have a fair idea about each feature but very small knowledge about their implementation. Question 1 username combobox : i use a combobox with isEditable set to true but i think it doesnt have the previous username, would that mean that i have to store every successful login username in a sqlite for example? Question 2 sign me in when skype start : i think about using sqlite after all to store the credentials and store the value (as in true or false) if autologin has to be performed. Question 3 start when computer start : i know it's about having is as service.but the process of using it as a service and removing its service when checkbox is checked or unckecked is a bit confusing to me. Question 4 Please wait(signing in) of skype if i want to do things like please wait at login(login is over webservice) in a WPF application should i use a animated gif in a grid that i can show when hiding the login combobox and passwordbox grid or i should use an animated object(for which i have no knowledge about for now) ? This post in mainly for you experts to either point me to the right resource and tell me what is done as best practice. things like dos and dons.Thanks for reading this and please let me have a clair idea about how to start implementing those features. thanks again

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  • What is your favorite programming discussion forum, channel, or chat?

    - by DV
    I think Stackoverflow is a great resource for programmers, but a lot of times free discussion and experimentation help find answers as quickly. I personally used Slicehost's forums a lot for setting up a Linux as a server. A quick search on Google turns up many results, but how to guess where to find the right people? In essence, are there communities out there where one can discuss and collaborate with great people similar to Stackoverflow audience? I am interested in multitude of topics, from PHP/ASP/Perl to C++/C#/Python, from SQL/XML/HTML/JSON to Javascript/Flash/Silverlight.

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