Search Results

Search found 4165 results on 167 pages for 'pulse audio'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 65/167 | < Previous Page | 61 62 63 64 65 66 67 68 69 70 71 72  | Next Page >

  • Preinitialize BackgroundAudioPlayer in WP7?

    - by kgrevehagen
    When I am using the BackgroundAudioPlayer in my Windows Phone 7 application, it takes a lot of time to load the first time I want to play a song. Is there any way of preinitializing the BackgroundAudioPlayer before playing the first track, so that when I start playing, it starts right along? I have googled it, but no luck. I am just using BackgroundAudioPlayer.Instance when I e.g. want to play, pause, stop etc an audiotrack. Is there something other i could do to fix this? Thanks in advance!

    Read the article

  • Drawing a waveform in C#

    - by user488792
    Hi! I want to be able to display a WaveForm in C#, along with some simple features such as zooming and selection. I already have the data as a short[] of amplitude values. However, I am an amateur when it comes to hardcoding GUI. I have already found a possible helper class WaveFormClass that may help me achieve this but as a backup, I want to learn how to manually do it. So may I ask for some methods and possibly some links that will help? Thanks!

    Read the article

  • Getting Frequency Components with FFT

    - by ruhig brauner
    so I was able to solv my last problem but i stubmled upon the next already. So I want to make a simple spectrogram but in oder to do so I want to understand how FFT-libaries work and what they actually calculate and return. (FFT and Signal Processing is the number 1 topic I will get into as soon as I have time but right now, I only have time for some programming exercises in the evening. ;) ) Here I just summarized the most important parts: int framesPerSecond; int samplesPerSecond; int samplesPerCycle; // right now i want to refresh the spectogram every DoubleFFT_1D fft; WAVReader audioIn; double audioL[], audioR[]; double fftL[], fftR[]; ..... framesPerSecond = 30; audioIn= new WAVReader("Strobe.wav"); int samplesPerSecond = (int)audioIn.GetSampleRate(); samplesPerCycle = (int)(audioIn.GetSampleRate()/framesPerSecond); audioL = new double[samplesPerCycle*2]; audioR = new double[samplesPerCycle*2]; fftL = new double[samplesPerCycle]; fftR = new double[samplesPerCycle]; for(int i = 0; i < samplesPerCycle; i++) { // don't even know why,... fftL[i] = 0; fftR[i] = 0; } fft = new DoubleFFT_1D(samplesPerCycle); ..... for(int i = 0; i < samplesPerCycle; i++) { audioIn.GetStereoSamples(temp); audioL[i]=temp[0]; audioR[i]=temp[1]; } fft.realForwardFull(audioL); //still stereo fft.realForwardFull(audioR); System.out.println("Check"); for(int i = 0; i < samplesPerCycle; i++) { //storing the magnitude in the fftL/R arrays fftL[i] = Math.sqrt(audioL[2*i]*audioL[2*i] + audioL[2*i+1]*audioL[2*i+1]); fftR[i] = Math.sqrt(audioR[2*i]*audioR[2*i] + audioR[2*i+1]*audioR[2*i+1]); } So the question is, if I want to know, what frequencys are in the sampled signal, how do I calculate them? (When I want to print the fftL / fftR arrays, I get some exponential formes at both ends of the array.) Thx :)

    Read the article

  • sound not playing when i press the button and how to fix overlapping sounds

    - by alfredjunco
    the code is giving me an error"Unused variable'path'" and when i press a button there is no sound playing how do i fix this the aSound is in the h file - (void)playOnce:(NSString *)aSound; - (IBAction) beatButton50 { [self playOnce:@"racecars"]; } - (void)playOnce:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if([theAudio isPlaying]) { [theAudio stop]; } } - (void)playLooped:(NSString *)aSound { NSString *path = [[NSBundle mainBundle] pathForResource:aSound ofType:@"caf"]; if (!theAudio) { theAudio = [[AVAudioPlayer alloc] initWithContentsOfURL: [NSURL fileURLWithPath: path] error: NULL]; } [theAudio setDelegate: self]; // loop indefinitely [theAudio setNumberOfLoops: -1]; [theAudio setVolume: 1.0]; [theAudio play]; } - (void)stopAudio { [theAudio stop]; [theAudio setCurrentTime:0]; }

    Read the article

  • Uninterrupted mp3 play on a website?

    - by Kevin
    Client is requesting a single track to be heard across the website. Generally I advise against it, but they insist. So, what is the most straightforward way of having a flash player embedded in a site, and when a user goes to another page there isn't a gap/interruption? I am thinking an iframe is required.. I am using a flash player that has autoresume, but that only solves picking up where you last left off on the song before going to another page. I tried searching SO for an answer..

    Read the article

  • CMake: Mac OS X: ld: unknown option: -soname

    - by Alex Ivasyuv
    I try to build my app with CMake on Mac OS X, I get the following error: Linking CXX shared library libsml.so ld: unknown option: -soname collect2: ld returned 1 exit status make[2]: *** [libsml.so] Error 1 make[1]: *** [CMakeFiles/sml.dir/all] Error 2 make: *** [all] Error 2 This is strange, as Mac has .dylib extension instead of .so. There's my CMakeLists.txt: cmake_minimum_required(VERSION 2.6) PROJECT (SilentMedia) SET(SourcePath src/libsml) IF (DEFINED OSS) SET(OSS_src ${SourcePath}/Media/Audio/SoundSystem/OSS/DSP/DSP.cpp ${SourcePath}/Media/Audio/SoundSystem/OSS/Mixer/Mixer.cpp ) ENDIF(DEFINED OSS) IF (DEFINED ALSA) SET(ALSA_src ${SourcePath}/Media/Audio/SoundSystem/ALSA/DSP/DSP.cpp ${SourcePath}/Media/Audio/SoundSystem/ALSA/Mixer/Mixer.cpp ) ENDIF(DEFINED ALSA) SET(SilentMedia_src ${SourcePath}/Utils/Base64/Base64.cpp ${SourcePath}/Utils/String/String.cpp ${SourcePath}/Utils/Random/Random.cpp ${SourcePath}/Media/Container/FileLoader.cpp ${SourcePath}/Media/Container/OGG/OGG.cpp ${SourcePath}/Media/PlayList/XSPF/XSPF.cpp ${SourcePath}/Media/PlayList/XSPF/libXSPF.cpp ${SourcePath}/Media/PlayList/PlayList.cpp ${OSS_src} ${ALSA_src} ${SourcePath}/Media/Audio/Audio.cpp ${SourcePath}/Media/Audio/AudioInfo.cpp ${SourcePath}/Media/Audio/AudioProxy.cpp ${SourcePath}/Media/Audio/SoundSystem/SoundSystem.cpp ${SourcePath}/Media/Audio/SoundSystem/libao/AO.cpp ${SourcePath}/Media/Audio/Codec/WAV/WAV.cpp ${SourcePath}/Media/Audio/Codec/Vorbis/Vorbis.cpp ${SourcePath}/Media/Audio/Codec/WavPack/WavPack.cpp ${SourcePath}/Media/Audio/Codec/FLAC/FLAC.cpp ) SET(SilentMedia_LINKED_LIBRARY sml vorbisfile FLAC++ wavpack ao #asound boost_thread-mt boost_filesystem-mt xspf gtest ) INCLUDE_DIRECTORIES( /usr/include /usr/local/include /usr/include/c++/4.4 /Users/alex/Downloads/boost_1_45_0 ${SilentMedia_SOURCE_DIR}/src ${SilentMedia_SOURCE_DIR}/${SourcePath} ) #link_directories( # /usr/lib # /usr/local/lib # /Users/alex/Downloads/boost_1_45_0/stage/lib #) IF(LibraryType STREQUAL "static") ADD_LIBRARY(sml-static STATIC ${SilentMedia_src}) # rename library from libsml-static.a => libsml.a SET_TARGET_PROPERTIES(sml-static PROPERTIES OUTPUT_NAME "sml") SET_TARGET_PROPERTIES(sml-static PROPERTIES CLEAN_DIRECT_OUTPUT 1) ELSEIF(LibraryType STREQUAL "shared") ADD_LIBRARY(sml SHARED ${SilentMedia_src}) # change compile optimization/debug flags # -Werror -pedantic IF(BuildType STREQUAL "Debug") SET_TARGET_PROPERTIES(sml PROPERTIES COMPILE_FLAGS "-pipe -Wall -W -ggdb") ELSEIF(BuildType STREQUAL "Release") SET_TARGET_PROPERTIES(sml PROPERTIES COMPILE_FLAGS "-pipe -Wall -W -O3 -fomit-frame-pointer") ENDIF() SET_TARGET_PROPERTIES(sml PROPERTIES CLEAN_DIRECT_OUTPUT 1) ENDIF() ### TEST ### IF(Test STREQUAL "true") ADD_EXECUTABLE (bin/TestXSPF ${SourcePath}/Test/Media/PlayLists/XSPF/TestXSPF.cpp) TARGET_LINK_LIBRARIES (bin/TestXSPF ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/test1 ${SourcePath}/Test/test.cpp) TARGET_LINK_LIBRARIES (bin/test1 ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/TestFileLoader ${SourcePath}/Test/Media/Container/FileLoader/TestFileLoader.cpp) TARGET_LINK_LIBRARIES (bin/TestFileLoader ${SilentMedia_LINKED_LIBRARY}) ADD_EXECUTABLE (bin/testMixer ${SourcePath}/Test/testMixer.cpp) TARGET_LINK_LIBRARIES (bin/testMixer ${SilentMedia_LINKED_LIBRARY}) ENDIF (Test STREQUAL "true") ### TEST ### ADD_CUSTOM_TARGET(doc COMMAND doxygen ${SilentMedia_SOURCE_DIR}/doc/Doxyfile) There was no error on Linux. Build process: cmake -D BuildType=Debug -D LibraryType=shared . make I found, that incorrect command generate in CMakeFiles/sml.dir/link.txt. But why, as the goal of CMake is cross-platforming.. How to fix it?

    Read the article

  • How to sync audio files with Logitech media server in MAC OS?

    - by Abhishek
    I want to customize the Logitech Media Server (web interface on localhost) so that N number of DIFFERENT audio files will start to play at the same time on N number of wifi receivers, each file on a different receiver. Currently, the server will sync only 1 track to N number(amount) of receivers. Is it possible with Logitech media server is open source. How can I able to do this? can you explain me sample code?

    Read the article

  • Ask HTG: Dealing with Windows 8 CP Expiry, Nintendo DS Save Backups, Jumbled Audio Tracks in Windows Media Player

    - by Jason Fitzpatrick
    Once a week we round up some great reader questions and share the answers with everyone. This week we’re looking at what to do when Windows 8 Consumer Preview expires, backing up your Nintendo DS saves, and how to sort out jumbled audio tracks in Windows Media Player movies. How To Be Your Own Personal Clone Army (With a Little Photoshop) How To Properly Scan a Photograph (And Get An Even Better Image) The HTG Guide to Hiding Your Data in a TrueCrypt Hidden Volume

    Read the article

  • How to record both audio, Where i have one music running and my microphone is in use?

    - by YumYumYum
    I have one music playing, and i have microphone open, already the microphone is used by other application. In such case, how can i record that music and the microphone audio to a file? (if possible with command line). Follow up: $ rec new-file.wav Input File : 'default' (alsa) Channels : 2 Sample Rate : 48000 Precision : 16-bit Sample Encoding: 16-bit Signed Integer PCM In:0.00% 00:00:25.94 [00:00:00.00] Out:1.24M [ | ] Clip:0 ^C $ sox -d new-file.wav

    Read the article

  • Join mp4 files in linux

    - by Jose Armando
    I want to join two mp4 files to create a single one. The video streams are encoded in h264 and the audio in aac. I can not re-encode the videos to another format due to computational reasons. Also, I cannot use any gui programs, all processing must be performed with linux command line utilities. FFmpeg cannot do this for mpeg4 files so instead I used MP4Box e.g. MP4Box -add video1.mp4 -cat video2.mp4 newvideo.mp4 unfortunately the audio gets all mixed up. I thought that the problem was that the audio was in aac so I transcoded it in mp3 and used again MP4Box. In this case the audio is fine for the first half of newvideo.mp4 (corresponding to video1.mp4) but then their is no audio and I cannot navigate in the video also. My next thought was that the audio and video streams had some small discrepancies in their lengths that I should fix. So for each input video I splitted the video and audio streams and then joined them with the -shortest option in ffmpeg. thus for the first video I ran avconv -y -i video1.mp4 -c copy -map 0:0 videostream1.mp4 avconv -y -i video1.mp4 -c copy -map 0:1 audiostream1.m4a avconv -y -i videostream1.mp4 -i audiostream1.m4a -c copy -shortest video1_aligned.mp4 similarly for the second video and then used MP4Box as previously. Unfortunately this didn't work either. The only success I had was when I joined the video streams separetely (i.e. videostream1.mp4 and videostream2.mp4) and the audio streams (i.e. audiostream1.m4a and audiostream2.m4a) and then joined the video and audio in a final file. However, the synchronization is lost for the second half of the video. Concretelly, there is a 1 sec delay of audio and video. Any suggestions are really welcome.

    Read the article

  • How to upload binary (audio) data from a Flash AS3 client to .NET server (WCF/REST/HTTP/?)?

    - by Bobby
    Simply stated: I'm trying to record audio in a browser, and get that data back up to the server. I originally tried to capture, encode and upload the audio using Silverlight, but because of the lack of suitable client-side encoding options, I'm now giving Flash a shot (Flash has baked-in support for encoding to Speex). I think I've figured out how to capture and encode the audio... But now what was easy in Silverlight, is the challenge in Flash. My server-side is .NET: MVC2- I'm open to receiving the audio in whatever manner is best- REST, WCF.. So that's my question: How could one upload binary data from Flash, to a .NET server-side endpoint. If the answer is WCF: then how would one setup the client-side proxies to communicate with the service? If the answer is REST or HTTP Post, then how would one construct this HTTP request and pass along the data? I've been reading up on AS3, but am new to Flash dev... Thanks for any help!

    Read the article

  • MPlayer does not work

    - by Soham Pal
    Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

    Read the article

  • alsa - sound issues on ubuntu 12.04

    - by tam_ubuuser
    i am having an sony E series laptop.i have an HDMI port .at this stage ,i have tested my sound card , which provides audio out on my laptop i.e i could hear songs .my laptop has two sound cards amd 5450 and an intel-hda(alsamixer shows that as s/pdif) . i decided to connect HDMI output to my new HD-TV.but, i could get only visuals on my TV,NO AUDIO OUTPUT ( HDMI cable works fine with win 7).my laptop has two sound cards.but i couldn't switch output to other card.( i don't know ,how to do that) i decided to update alsa. complied the following code in terminal. sudo apt-add-repository ppa:ubuntu-audio-dev/alsa-daily sudo apt-get update sudo apt-get install alsa-hda-dkms then,strangely no login sound, and no audio output on my laptop at all .then, started complied code from step1 sound troubleshooting procedure from offical ubuntu site.then, my speaker icon taskbar disappeared .obivously $aplay -l ,provided output as no soundcards detected . so , i implemented step 4 from that guide, it provides a output of all hardware devices in my laptop. *-multimedia UNCLAIMED description: Audio device product: Cedar HDMI Audio [Radeon HD 5400/6300 Series] vendor: Hynix Semiconductor (Hyundai Electronics) physical id: 0.1 bus info: pci@0000:01:00.1 version: 00 width: 64 bits clock: 33MHz capabilities: pm pciexpress msi bus_master cap_list configuration: latency=0 resources: memory:f0040000-f0043fff *-multimedia UNCLAIMED description: Audio device product: 5 Series/3400 Series Chipset High Definition Audio vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 05 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list configuration: latency=0 resources: memory:f5e00000-f5e03fff that command displayed output name of the two cards . but , still i have no positive output on $aplay -l. so therfore, i think alsa couldn't detect my sound cards . is there solution to this problem? it could be better,if alsa would channel output from multiple sound cards ? how should install and configure alsa such that detects HDMI cable as soon i connect to my HD tv? is it possible to alsa and pluseaudio 2.0 to co-exist, if so how?

    Read the article

  • why some mp3s on mime_content_type return application/octet-stream

    - by robertdd
    why on some mp3s file when i call mime_content_type($mp3_file_path) it's return application/octet-stream? i have this: if (!empty($_FILES)) { $tempFile = $_FILES['Filedata']['tmp_name']; $image = getimagesize($tempFile); $mp3_mimes = array('audio/mpeg', 'audio/x-mpeg', 'audio/mp3', 'audio/x-mp3', 'audio/mpeg3', 'audio/x-mpeg3', 'audio/mpg', 'audio/x-mpg', 'audio/x-mpegaudio'); if (in_array(mime_content_type($tempFile), $mp3_mimes)) { echo json_encode("mp3"); } elseif ($image['mime']=='image/jpeg') { echo json_encode("jpg"); } else{ echo json_encode("error"); } }

    Read the article

  • Why did this command make my Ubuntu more awesome than before?

    - by the_misfit
    After running Ubuntu for months and liking it, I for the first time wanted to use headphones and was trying to get them to work so I ran the code in step 1 https://help.ubuntu.com/community/SoundTroubleshootingProcedure -- the result was an improvement in performance, resolution, and audio.. a vast improvement. What does that suggest about what was wrong with my settings before if running this improved system performance, graphics, and my audio problem? The command is: sudo add-apt-repository ppa:ubuntu-audio-dev/ppa; sudo apt-get update;sudo apt-get dist-upgrade; sudo apt-get install linux-sound-base alsa-base alsa-utils gdm ubuntu-desktop linux-image-`uname -r` libasound2; sudo apt-get -y --reinstall install linux-sound-base alsa-base alsa-utils gdm ubuntu-desktop linux-image-`uname -r` libasound2; killall pulseaudio; rm -r ~/.pulse*; sudo usermod -aG `cat /etc/group | grep -e '^pulse:' -e '^audio:' -e '^pulse-access:' -e '^pulse-rt:' -e '^video:' | awk -F: '{print $1}' | tr '\n' ',' | sed 's:,$::g'` `whoami`

    Read the article

  • Where can I buy a Stereo audio to 3.5mm adapter?

    - by iftrue
    I need a stereo (6.33mm) to PC audio (3.5mm) adapter, and I'd like it to have an inch or two of cable so that yanking the connector doesn't break the audio port the 3.5mm is plugged into. I used to own one of these, but I lost the adapter. Where can I buy something like this online? I can only find solid adapters or 25' cables.

    Read the article

  • JNI Stream binary data from C++ to Java

    - by Cliff
    I need help passing binary data into Java. I'm trying to use jbytearray but when the data gets into Java it appears corrupt. Can somebody give me a hand? Here's a snip of some example code. First the native C++ side: printf("Building audio array copy\n"); jbyteArray rawAudioCopy = env-NewByteArray(10); jbyte toCopy[10]; printf("Filling audio array copy\n"); char theBytes[10] = {0,1,2,3,4,5,6,7,8,9}; for (int i = 0; i < sizeof(theBytes); i++) { toCopy[i] = theBytes[i]; } env->SetByteArrayRegion(rawAudioCopy,0,10,toCopy); printf("Finding object callback\n"); jmethodID aMethodId = env->GetMethodID(env->GetObjectClass(obj),"handleAudio","([B)V"); if(0==aMethodId) throw MyRuntimeException("Method not found error",99); printf("Invoking the callback\n"); env->CallVoidMethod(obj,aMethodId, &rawAudioCopy); and then the Java callback method: public void handleAudio(byte[] audio){ System.out.println("Audio supplied to Java [" + audio.length + "] bytes"); byte[] expectedAudio = {0,1,2,3,4,5,6,7,8,9}; for (int i = 0; i < audio.length; i++) { if(audio[i]!= expectedAudio[i]) System.err.println("Expected byte " + expectedAudio[i] + " at byte " + i + " but got byte " + audio[i]); else System.out.print('.'); } System.out.println("Audio passed back accordingly!"); } I get the following output when the callback is invoked: library loaded! Audio supplied to Java [-2019659176] bytes Audio passed back accordingly!

    Read the article

  • How do I change which audio jacks are used for input and output?

    - by yamaha1996
    I'm using a Realtek HD audio card built-in my motherboard. The Windows driver comes with a control panel that allows me to select which back panel jacks are used for what. So for example I can make both the blue jack and green jack for output and only the red one for mic-in. (Whereas by default, the blue jack is for line in, which I never need.) How can I do the same under Linux? If possible, please don't suggest something that involves PulseAudio or JACK; I'd like to do it the plain way, e.g. by editing ALSA configuration files, if possible. The way I understand it, my problem should have nothing to do with software servers redirecting streams, just instructing the driver to treat this jack as so and so because it's hardware supported. Thank you very much!

    Read the article

  • Best way to distribute graphics, audio and levels with an SDL game?

    - by Kristopher
    I'm working on finishing up a game written in C++ with SDL I've been working on for awhile, and I'm starting to ponder how I'm going to distribute it. It has hundreds of images that are loaded and used throughout the game, as well as a couple dozen .wav files for audio effects. What is the best way to distribute these? Should I just include the folders with all the files? Or is there a way I can package them into a single file, then open and extract them in my application? What's the best way to go about this?

    Read the article

  • How do I fix skype on x86 11.10 desktop which garbles audio both ways in connection with a Windows user?

    - by keepitsimpleengineer
    Strange phenomena ? I have a friend I connect with and talk to regularly occasionally using video. He runs Windows XP with a Logitech webcam and mike. I run Windows XP with a Logitech webcam and mike, and a laptop with built in webcam and mike running both 11.10 x86 Desktop and Windows XP Home. We gab and ogle fine with these. But when I installed 11.10 x86 with GNOME 3.2 dual boot on the same machine with the Logitech webcam and mike, whenever I try to ogle and gab the audio is total garbled. When I make a test call to skype or call somebody on a landline or cell - there's no problem. In summary, everything works except for when running 11.10 x86 GNOME 3.2 connected to his Windows XP. Stumped on this end.

    Read the article

  • how to make audio and video streaming servers work?

    - by explorex
    I am a php mysql developer ... just an (below) average. and i am interested in the way television and radio are broadcasted over internet live. i want to know how it works and and what are its requirements (which package of which programming language offers the best). i must admit that i am a complete layman but i expect it do by next half month or year or so. And please clarify me Websites are stored in servers. From my desktop, i want to broadcast some video, then i need to connect to webserver(to upstream the video). Is there an application to do that (or do i have to code that or embed in my web application and which programming language would be suitable(does python support that))? and i also need a script to handle the upstreamed video or audio(can i do that with php)?

    Read the article

  • How to make audio and video streaming servers work?

    - by Santosh Linkha
    I am PHP MySQL developer and I am interested in the way television and radio are broadcasted over Internet live. I want to know how it works and and what are its requirements (which package of which programming language offers the best). And please clarify me: Websites are stored in servers. From my desktop, if I want to broadcast some video, then I need to connect to webserver (to upstream the video). Is there an application to do that (or do I have to code that or embed in my web application and which programming language would be suitable (does Python support that))? And I also need a script to handle the upstreamed video or audio (can I do that with PHP)?

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

< Previous Page | 61 62 63 64 65 66 67 68 69 70 71 72  | Next Page >