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  • Where to get streaming (live) video and audio from camera example app for Android?

    - by Ole Jak
    Where to get streaming (live) video and audio from camera example for Android? Suppose I want to create some live video streaming service app so I'll have some cool server at the back end. And I know how to do that part. Suppose I have some stand alone app for PCs now I want to go on to mobile devices. So I want to see some sample app grabing audio and video streams from Phone, Synchronizing them, encoding somehow, and sending LIVE stream to server. I need any Open-Source sample that will do this or something like this. Where can I get one?

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  • How do I write the audio stream to a memory buffer instead of a file using DirectShow?

    - by yngvedh
    Hi, I have made a sample application which constructs a filter graph to capture audio from the microphone and stream it to a file. Is there any filter which allows me to stream to a memory buffer instead? I'm following the approach outlined in an article on msdn and are currently using the CLSID_FileWriter object to write the audio to file. This works nicely, but I cannot figure out how to write to a memory buffer. Is there such a memory sink filter or do I have to create it myself? (I would prefer one which is bundled with windows XP)

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  • What open source C/C++ audio compression options are there besides LAME MP3?

    - by Ole Jak
    Are there any C/C++ open source audio encoder besides LAME MP3? It doesn't need to be exactly mp3 format, I need a "compressed digital audio file". I do not want to use Lame because it is too big while no programmer can answer a simple question on it (share simple but easily downloadable and readable project containing only needed 2 simple functions... So I'm tired of searching for help with it.. I need something fresh powerful but more readable than this lib I found (mp3stego) ) "I don't want LAME because I am a fighter with its monopoly" Haha..

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  • Sentiment analysis with NLTK python for sentences using sample data or webservice?

    - by Ke
    I am embarking upon a NLP project for sentiment analysis. I have successfully installed NLTK for python (seems like a great piece of software for this). However,I am having trouble understanding how it can be used to accomplish my task. Here is my task: I start with one long piece of data (lets say several hundred tweets on the subject of the UK election from their webservice) I would like to break this up into sentences (or info no longer than 100 or so chars) (I guess i can just do this in python??) Then to search through all the sentences for specific instances within that sentence e.g. "David Cameron" Then I would like to check for positive/negative sentiment in each sentence and count them accordingly NB: I am not really worried too much about accuracy because my data sets are large and also not worried too much about sarcasm. Here are the troubles I am having: All the data sets I can find e.g. the corpus movie review data that comes with NLTK arent in webservice format. It looks like this has had some processing done already. As far as I can see the processing (by stanford) was done with WEKA. Is it not possible for NLTK to do all this on its own? Here all the data sets have already been organised into positive/negative already e.g. polarity dataset http://www.cs.cornell.edu/People/pabo/movie-review-data/ How is this done? (to organise the sentences by sentiment, is it definitely WEKA? or something else?) I am not sure I understand why WEKA and NLTK would be used together. Seems like they do much the same thing. If im processing the data with WEKA first to find sentiment why would I need NLTK? Is it possible to explain why this might be necessary? I have found a few scripts that get somewhat near this task, but all are using the same pre-processed data. Is it not possible to process this data myself to find sentiment in sentences rather than using the data samples given in the link? Any help is much appreciated and will save me much hair! Cheers Ke

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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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  • Overriding Object.Equals() instance method in C#; now Code Analysis / FxCop warning CA2218: "should

    - by Chris W. Rea
    I've got a complex class in my C# project on which I want to be able to do equality tests. It is not a trivial class; it contains a variety of scalar properties as well as references to other objects and collections (e.g. IDictionary). For what it's worth, my class is sealed. To enable a performance optimization elsewhere in my system (an optimization that avoids a costly network round-trip), I need to be able to compare instances of these objects to each other for equality – other than the built-in reference equality – and so I'm overriding the Object.Equals() instance method. However, now that I've done that, Visual Studio 2008's Code Analysis a.k.a. FxCop, which I keep enabled by default, is raising the following warning: warning : CA2218 : Microsoft.Usage : Since 'MySuperDuperClass' redefines Equals, it should also redefine GetHashCode. I think I understand the rationale for this warning: If I am going to be using such objects as the key in a collection, the hash code is important. i.e. see this question. However, I am not going to be using these objects as the key in a collection. Ever. Feeling justified to suppress the warning, I looked up code CA2218 in the MSDN documentation to get the full name of the warning so I could apply a SuppressMessage attribute to my class as follows: [SuppressMessage("Microsoft.Naming", "CA2218:OverrideGetHashCodeOnOverridingEquals", Justification="This class is not to be used as key in a hashtable.")] However, while reading further, I noticed the following: How to Fix Violations To fix a violation of this rule, provide an implementation of GetHashCode. For a pair of objects of the same type, you must ensure that the implementation returns the same value if your implementation of Equals returns true for the pair. When to Suppress Warnings ----- Do not suppress a warning from this rule. [arrow & emphasis mine] So, I'd like to know: Why shouldn't I suppress this warning as I was planning to? Doesn't my case warrant suppression? I don't want to code up an implementation of GetHashCode() for this object that will never get called, since my object will never be the key in a collection. If I wanted to be pedantic, instead of suppressing, would it be more reasonable for me to override GetHashCode() with an implementation that throws a NotImplementedException? Update: I just looked this subject up again in Bill Wagner's good book Effective C#, and he states in "Item 10: Understand the Pitfalls of GetHashCode()": If you're defining a type that won't ever be used as the key in a container, this won't matter. Types that represent window controls, web page controls, or database connections are unlikely to be used as keys in a collection. In those cases, do nothing. All reference types will have a hash code that is correct, even if it is very inefficient. [...] In most types that you create, the best approach is to avoid the existence of GetHashCode() entirely. ... that's where I originally got this idea that I need not be concerned about GetHashCode() always.

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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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  • 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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  • 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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  • Design considerations on JSON schema for scalars with a consistent attachment property

    - by casperOne
    I'm trying to create a JSON schema for the results of doing statistical analysis based on disparate pieces of data. The current schema I have looks something like this: { // Basic key information. video : "http://www.youtube.com/watch?v=7uwfjpfK0jo", start : "00:00:00", end : null, // For results of analysis, to be populated: // *** This is where it gets interesting *** analysis : { game : { value: "Super Street Fighter 4: Arcade Edition Ver. 2012", confidence: 0.9725 } teams : [ { player : { value : "Desk", confidence: 0.95, } characters : [ { value : "Hakan", confidence: 0.80 } ] } ] } } The issue is the tuples that are used to store a value and the confidence related to that value (i.e. { value : "some value", confidence : 0.85 }), populated after the results of the analysis. This leads to a creep of this tuple for every value. Take a fully-fleshed out value from the characters array: { name : { value : "Hakan", confidence: 0.80 } ultra : { value: 1, confidence: 0.90 } } As the structures that represent the values become more and more detailed (and more analysis is done on them to try and determine the confidence behind that analysis), the nesting of the tuples adds great deal of noise to the overall structure, considering that the final result (when verified) will be: { name : "Hakan", ultra : 1 } (And recall that this is just a nested value) In .NET (in which I'll be using to work with this data), I'd have a little helper like this: public class UnknownValue<T> { T Value { get; set; } double? Confidence { get; set; } } Which I'd then use like so: public class Character { public UnknownValue<Character> Name { get; set; } } While the same as the JSON representation in code, it doesn't have the same creep because I don't have to redefine the tuple every time and property accessors hide the appearance of creep. Of course, this is an apples-to-oranges comparison, the above is code while the JSON is data. Is there a more formalized/cleaner/best practice way of containing the creep of these tuples in JSON, or is the approach above an accepted approach for the type of data I'm trying to store (and I'm just perceiving it the wrong way)? Note, this is being represented in JSON because this will ultimately go in a document database (something like RavenDB or elasticsearch). I'm not concerned about being able to serialize into the object above, because I can always use data transfer objects to facilitate getting data into/out of my underlying data store.

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  • Learn How to Use Oracle’s Spatial and BI Tools for Location-aware Predictive Analytics

    - by Mandy Ho
    November 29, 2-3pm EST Are you a OBIEE (Oracle Business Intelligence Enterprise Edition) user? Have Location data you'd like to incorporate into your analysis as well? This is a great webinar for you! Join us, as Oracle experts from both teams show how to perform perdictive analytics, network analytics and spatial analysis, combined together, in real world scenarios. We will include demos evaluating airline on-time performance and retail establishment performance.  Learn how to: - Gain better business insights and improve ROI with Oracle Spatial and Graph, Oracle Advanced Analytics, and Oracle Business Intelligence Enterprise Edition (OBIEE). - Streamline and remove the complexity of building applications with OBIEE’s built-in location and analytics features. - Create the statistical model, build interactive reports and dashboards including location analysis and map visualization, and incorporate network analytics for geomarketing and site scoring. - Perform location analysis and processing such as proximity, containment, geocoding, aggregation of geographic regions, and more. Speakers include Jayant Sharma, Director, Product Management, Oracle Spatial and Mapping Technologies; Jean Ihm, Principal Product Manager, Oracle Spatial and Mapping Technologies; and Abhinav Agarwal, OBIEE Product Management. Who should attend This webinar is appropriate for CIOs, business and technical managers, developers, and analysts involved in design and management of analytic applications and solutions where spatial analysis can add insight and value to business processes. Click here, or the link below to sign up today! https://www2.gotomeeting.com/register/764677554

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  • Web application Project management methodologie

    - by dutchiexl
    I am looking to streamline my company's web development process. Including analysis. I myself am specialized in XP and Scrum. But we are building web application with a process cycle of 3-4 weeks and a lifetime of 1-4 months. When a project is sold, only then the project managers (= people who do analysis but know nothing about it = a small flow chart and some screen shots as analysis) What is happening is: A LOT of change requests Minimal development time Minimal analysis time NOW: the main question :) can you recommend me some methodologies and books to read for the entire project management ? Thanks in advance @Edit, I myself was looking at a combination of SCRUM for the management with flowcharts, + RAD/LD for development, and trying to distilate something from that.

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