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• java 2d game how to make a player jump

  - by user2957632

  Hi I'm making a 2d plat former running game kind of like jetpack joyride fro example were u are constantly running but there is obstacles and u need to jump over them. but I want the character to jump and come back down. here is some of my code. if (listen.ml == true) { if (y > 120) { jump = true; y -= 1; } } if (y == 121 && jump == true) { jump = false; y += 1; }

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• Un player en Silverlight à reconnaissance vocale retranscrit et tague automatiquement le contenu d'u

  Un lecteur en Silverlight à reconnaissance vocale Retranscrit et tague automatiquement le contenu d'une vidéo La nouvelle n'a pas fait grand bruit et pourtant, techniquement, elle est importante. Au moment où Google introduit la publicité dans les vidéos de Youtbe ? une démarche dont la pertinence, notamment technique, pose question ? Microsoft vient de sortir, en collaboration avec le Lab de France 24, la chaine d'information francophone internationale, un lecteur d'un nouveau type. ...

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• ffdshow h.264 audio desync

  - by Core Xii

  When I encode video with ffdshow with h.264, the audio is out of sync. At the very beginning of the video, the picture freezes for about 1 second, while the audio plays fine, resulting in the audio being that 1 second ahead of the picture throughout the entire video. Any ideas on possible causes or, obviously, solutions?

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• Media player for Windows 7 Mobile

  This article will give step to step illustration of creating a simple media application for Windows 7 mobile.

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• Java - Finding distance between player and tile in array

  - by Corey

  What is the best way performance wise to do this? When I click a tile I want it to get the distance and if I am close enough I can interact with the tile. One way would be to find the tile by doing mouse / tile width when I click correct? But then how would I get that tiles position? I know how to find the distance I just don't know how to get a certain tiles position from the array when I click it

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• Why won't AVI2DVD load the audio stream?

  - by Xavierjazz

  XP SP3 I have an .avi file. It is in a folder on my "C:" drive. There are no disallowed characters in either the folder or file name. It has audio as I have watched it on my computer. I want to burn it to a DVD. When I load the file into AVI2DVD, no audio stream shows, and the program will not work without an audio stream. I have used the net extensively to try and solve this, with no success. AFICT I have followed all instructions exactly, but no audio stream. Very frustrating. Does anyone have a clue? Can you help me? Thank you. Regards,

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• Sound Delay With AVAudio Player

  - by Will Youmans

  I'm using the following code in my viewDidLoad to load a sound: NSURL * url = [NSURL fileURLWithPath: [NSString stringWithFormat:@"%@/Moto - Hit Sound.mp3", [[NSBundle mainBundle] resourcePath]]]; NSError * error; hitSoundPlayer = [[AVAudioPlayer alloc]initWithContentsOfURL:url error:&error]; hitSoundPlayer.numberOfLoops = 0; Then I'm using this in a void method to play the sound: if(CGRectIntersectsRect(main.frame, enemy1.frame)){ [hitSoundPlayer play]; } This does seem to work, however the first time the sound is played there is a lot of lag and the game stops temporarily. I'm using this same method for when in an IBAction and it works fine, it must be the fact that it's also detecting a collision that makes the sound lag. If I want to be able to play sounds quickly and on the spot without any sort of lag am I doing the right thing? Do I want to use another method? I'm not using any frameworks like cocos2d. If you need to see any more code just ask.

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• calculating player experience

  - by user1765862

  very simple question, I'm trying to learn advanced principles of .net and c# and I'm in the middle of creating some simple manager game. Now I should implement some experience for players. I was thinking to implement some kind of enumerated values like this private enum ExperienceValues { FriendlyMatch = 0.1, Training = 0.15, LeagueMatch = 0.6, CupMatch = 0.85, Qualification = 1.4 } And to calculate experience by the time user spend on the field 90min * 0.6 = 54 Is this approach ok ? How can I abstract experience calculation for common sports (team sport). Thanks

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• Microphone audio streaming from Cocoa mac app to iPhone

  - by Benzamin

  Hi devs, I'm trying to build a microphone audio streamer to iPhone. The server software will be a mac desktop app and the client will be iPhone, and they are connected via tcp port. I've successfully connected the mac app and iPhone, and tried to send a fixed test.m4a audio file first. But at the iPhone i grabbed the data well, when tried to play it i used AVAudioPlayer and its returning OSStatus error. I played around with the audio queue service but its very tricky and i only got some example for fixed length audio playing like http://cocoawithlove.com/2009/06/revisiting-old-post-streaming-and.html Now i need help on two things, how can i continuously grab audio data from Mac desktop microphone? And then after grabbing the data how i can play this unfixed length audio data in the iPhone. What exactly i need to do? Please please help me on this......

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• Get binary data from audio impulses

  - by Timo

  I have IR sensor which have TRS plug and I can record my remotes signals into audio. Now I want to control my computer with TV remote, but I don't have any clue how to compare audio input with pre-recorded audio. But after I realized that these audio waves contains only some kind data (binary) I can turn these into binary or hex, so it is much easier to compare. Waves look just like this: http://i.imgur.com/lCIyl.png And this: ttp://i.imgur.com/goJ6d.png These are records of "OK" button, sometimes there are some impulses on right channel too and I don't know why, it seems like connections in sensor are damaged maybe. Ok thats not matter, anyway I need help with python program which read these impulses and turn these into binary, in realtime from audio input(mic). I know it's sounds like "Do it for me, while I enjoy my life", but I don't have experiences with sound transforming/reading... I've looking for python examples for recording and reading audio, but unsuccessfully.

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• How do I set up an IP address on a Linux VM running in VM Player so I can access it from my Windows 7 host?

  - by BradyKelly

  I have just installed an Openbravo appliance on my Windows 7 VM Player host. I am now staring at a command prompt that tells me to go to http://localhost to access the ERP system, but I cannot find any browser on the appliance. I am guessing I should rather follow their advice to configure an IP address for the Linux VM and just access that from a Windows browser on my host. How do I go about this? More specifically, How do I choose a local IP address to assign? How do I set things up so that this IP address is visible to my Windows host? Their help says to assign an DNS, to make the server visible to the internet, but internet visibility per se is not needed. How should I interpret or adapt this help for that? Finally to make the IP address available to the Internet, assign some DNS servers to it: $ echo "nameserver IP_DNS1" /etc/resolv.conf $ echo "nameserver IP_DNS2" /etc/resolv.conf

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• I copied a Windows VM from Mac VMWare to Windows VMWare Player. It runs, but very slowly...

  - by thrillscience

  I copied a Windows XP VM that I've been using on my Mac (VMWare Fusion)to a Windows 7 machine that has VMWare 7 installed. I was quite pleased when it started up and appeared to work, but when I actually tried to use it, I noticed it runs very slowly. Unusably so. It takes about 10 minutes, for example, for a Visual Studio 2010 project to open (with VS 2010 running in the VM). Is this supposed to work? Is there any way to fix this VM to get it to run well under Windows VMWare Player?

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• ffmpeg: video file played OK on Ubuntu, but no sound on XP

  - by Andy Le

  I created a video clip using ffmpeg (vcodec: mpeg2video, acodec: AC3 5.1). The file can be played normally on Ubuntu, but when I play it on an XP machine, there is no sound. I can play AC3 files and other movies with AC3 sound. I already tried many codec packs and many players. When I compare the MediaInfo tab of the Properties window of the file with another playable movie, I see that the Audio Identifier of the audio stream in my file is 0x80 while it is 0x02 in the other movie. So I guess that's why players on XP can't recognize the audio codec. When I use an MKV container instead of MPEG (still mpeg2video codec), then the result is OK on both Ubuntu and XP (with the correct Audio ID). I really need MPEG though. Any idea? This is the command I used: ~/ffmpeg/ffmpeg/ffmpeg -loop_input \ -t 97 -r 30000/1001 -i v%4d.tga -i final.ac3 \ -vcodec mpeg2video -qscale 1 -s 400x400 -r 30000/1001 \ -acodec copy -y out6.mpeg 2 This is the output of mediainfo (on Ubuntu): General Complete name : out6.mpeg Format : MPEG-PS File size : 6.86 MiB Duration : 1mn 37s Overall bit rate : 593 Kbps Video ID : 224 (0xE0) Format : MPEG Video Format version : Version 2 Format profile : Main@Main Format settings, BVOP : No Format settings, Matrix : Default Format_Settings_GOP : M=1, N=12 Duration : 1mn 37s Bit rate mode : Variable Bit rate : 122 Kbps Width : 400 pixels Height : 400 pixels Display aspect ratio : 1.000 Frame rate : 29.970 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.025 Stream size : 1.41 MiB (21%) Audio ID : 128 (0x80) Format : AC-3 Format/Info : Audio Coding 3 Duration : 1mn 36s Bit rate mode : Constant Bit rate : 448 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 44.1 KHz Stream size : 5.18 MiB (75%)

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• Downmix surround to Dolby Pro-Logic at the OS/driver level in Windows 7?

  - by davr

  First off, I'm talking about Dolby Pro-Logic, a really old tech for encoding 4 audio channels (L/R/C/SR) into two analog outputs, and then extracting them again. It was used in surround sound systems in the last century. I have a modern PC that can output 5.1 analog audio (Three outputs on the back carry six channels of audio). But I have a really old surround sound reciever that only has a two-channel, L/R input, which it extracts 4 channels of audio from, and outputs to 5.1 speakers. What I want is some way for the OS, Windows 7, to act as if I really had 5.1 audio channels available, so applications produce surround audio, but before outputting it out of the back of my PC, apply Dolby Pro-Logic matrix encoding so that it outputs over only two channels. These two channels would then get sent to my receiver via a RCA cable, which would decode it again and drive the surround speakers. Is anything like this possible? I'm pretty sure I could do it at an application / codec level, but I'm looking for something that I just have to set once.

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• Is it possible to embed a flash player in a table and retain the table properties

  - by user1494241

  I'd like to embed a music (flash) player in a table with clickable images but the embed code seems to throw the table properties off - it extends the width of the table. Is it possible to embed the player on the same row as the image whilst still retaining the table width? Here's what I've been using: <table width="620" border="0" cellspacing="0" cellpadding="0"> <tr> <td> <div align="left"><object height="18" width="100%"> <param name="movie" value="https://player.soundcloud.com/player.swf?url=http%3A%2F%2Fapi.soundcloud.com%2Fplaylists%2F1253725&amp;auto_play=false&amp;player_type=tiny&amp;font=Georgia&amp;color=9a6600&show_playcount=false&default_width=375&default_height=40&show_user=false"></param> <param name="allowscriptaccess" value="always"></param> <param name="wmode" value="transparent"></param><embed wmode="transparent" allowscriptaccess="always" height="18" src="https://player.soundcloud.com/player.swf?url=http%3A%2F%2Fapi.soundcloud.com%2Fplaylists%2F1253725&amp;auto_play=false&amp;player_type=tiny&amp;font=Georgia&amp;color=9a6600&show_playcount=false&default_width=375&default_height=40&show_user=false" type="application/x-shockwave-flash" width="100%"></embed> </object> </td> <td> <div align="right"><img src="http://dl.dropbox.com/u/31856944/Virb/splash_freedownload-2.png" border="0" width="245" height="42" usemap="#Map" /></div> </td> <tr> <td> <div align="right"><img src="http://dl.dropbox.com/u/31856944/Virb/splash_share-2.png" border="0" width="620" height="31" usemap="#Map2" /></div> </td> </tr> </table>

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• What is the best way to get an audio file duration in Android?

  - by Gilead

  Hi! I'm using a SoundPool [ 1 ] to play audio clips in my app. All is fine but I need to know when the clip playback has finished. At the moment I track it in my app by obtaining the duration of each clip using a MediaPlayer [ 2 ] instance. That works fine but it looks wasteful to load each file twice, just to get the duration. I could roughly calculate the duration myself knowing the length of the file (available from the AssetFileDescriptor [ 3 ]) but I'd still need to know the sample rate and the number of channels. I see two potential solutions to that problem: Figuring out when a clip has finished playing (doesn't seem to be possible with SoundClip). Having a class which could load just the header of an audio file and give me the sample rate/number of channels (and, ideally, the sample count to get the exact duration). Any suggestions? Thanks, Max The code I'm using at the moment (works fine but is rather heavy for the purpose): String[] fileNames = ... MediaPlayer mp = new MediaPlayer(); for (String fileName : fileNames) { AssetFileDescriptor d = context.getAssets().openFd(fileName); mp.reset(); mp.setDataSource(d.getFileDescriptor(), d.getStartOffset(), d.getLength()); mp.prepare(); int duration = mp.getDuration(); // ... } On a side note, this question has already been asked [ 4 ] but got no answers.

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• Issues regarding playing audio files in a JME midlet.

  - by Northernen

  I am making a midlet which is to be used to play out local audio files. It is obviously not working. I am getting a null reference on the "is" variable, in the code snippet shown below. 1. try{ 2. System.out.println("path: " + this.getClass()); 3. InputStream is = this.getClass().getResourceAsStream("res/01Track.wav"); 4. p1=Manager.createPlayer(is, "audio"); 5. p1.realize(); 6. p1.prefetch(); 7. p1.start(); 8. } 9. catch(Exception e){ 10. System.out.println(e.getMessage()); 11. } I assume there is something wrong with the "this.getClass().getResourceAsStream("res/01Track.wav")" bit, but I can not for the life of me figure out why, and I have tried referring to the file in 20 different ways. If I printline "this.getClass()" it gives me "path: class Mp3spiller". The absolute path to "01Track.wav" is "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller\res\01Track.wav". Am I completely wrong in thinking that I should refer relatively to "E:\Mine dokumenter\Dokumenter\workspace_mobiljava\Mp3spiller"? If anyone could point out what I am doing wrong, I would be grateful. I have basically stolen the code from a tutorial I found online, so I would have thought it would be working.

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• How to make windows media player go to previous song in playlist?

  - by SadSido

  Hi, everyone! I am writing a simple Windows app in c++, that will be able to send commands to windows media player. My problem is that I want my app to move to the previous song in the playlist. IWMPControls::previous() seems to do the job, but its behavior differs from what is written in msdn. In fact this function rewinds current media to the beginning and then (if current position is less than 2-3 seconds) it switches to the previous song. I would like to implement two different buttons (please, don't ask me why :)) - one for rewinding to the beginning, and one - to moving to previous song. Is there any easy way to do this through IWMPControls (or any other WMP-related COM interface)? p.s. I could handle this if I could get the position (index) of the current song in the list. But as far as I read MSDN, it seems to me that there is no easy way to get the current item index from playlist...

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• How do I get information about the level to the player object?

  - by pangaea

  I have a design problem with my Player and Level class in my game. So below is a picture of the game. The problem is I don't want to move on the black space and only the white space. I know how to do this as all I need to do is get the check for the sf::Color::Black and I have methods to do this in the Level class. The problem is this piece of code void Game::input() { player.input(); } void Game::update() { (*level).update(); player.update(); } void Game::render() { (*level).render(); player.render(); } So as you there is a problem in that how do I get the map information from the Level class to the Player class. Now I was thinking if I made the Player position static and pass it into the Level as parameter in update I could do it. The problem is interaction. I don't know what to do. I could maybe make player go into the Level class. However, what if I want multiple levels? So I have big design problems that I'm trying to solve.

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• No audio flash movies

  - by Valerio Giuffrida

  I've installed natty a few days ago and I'm experience some issues with the audio. Basically, I'm not able to listen anything coming from flash movies (such as youtube). This happen with both chromium and firefox (I use only the first one) and the errore I get in the stdout is: ALSA lib pcm.c:2109:(snd_pcm_open_conf) Cannot open shared library /usr/lib/alsa-lib/libasound_module_pcm_pulse.so I don't know how to fix it. I found somewhere to install native x64 flash plugins, but somebody says it is not advisable. I didn't get this kind of issues with 10.10 I've gotten before of natty. Hence, what am I supposed to do? thank you very much

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• Skipping video and audio with PS3MediaServer

  - by MaxMackie

  I'm using the latest PS3MediaServer build right from the repos suggested in the Ubuntu Wiki. I'm streaming multiple movies from my server (Ubuntu 10.04 LTS) to my PS3 over wireless. Sometimes, during some movies, the audio and the video will begin skipping. This can last anywhere between 5 and 30 seconds before it goes back to normal. I have a four core i5 processor and 8GB of DDR3 RAM so I don't think my computer is having a hard time keeping up with the transcoding. So this leads me to believe it's either sub-optimal transcoding options from within PS3MS or my network can't handle the heat. Other than the out-of-box configuration, is there any way I can tweak the settings for the application to use my resources more efficiently?

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• Internet stops working after heavy downloading, video/audio streaming etc

  - by Kuba Szwed

  As mentioned in title, Internet stops working on my PC after heavy downloading, video/audio streaming etc. There are no errors, no disconnections etc. Simply after some time (certain amount of data downloaded) I can't get any more. If I try using ping afterwards nothing happens. If ping is running simultaneously with streaming/downloading I get some correct responses and then it keeps showing an error. What helps is re-plugging my Pentagram USB wifi card, but I hope there is a better solution. Edit: One more thing: my friend who works in IT suggested that it might have something to do with cache (DNS cache? I don't remember him specifying) getting filled while it should be emptied automatically.

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• How to cut audio file with avconv?

  - by x-yuri

  I have a hard time trying to figure out how to cut a file with avconv. Here's the command I use: avconv -ss 52:13:49 -t 01:13:52 -i RR119Accessibility.wav RR119Accessibility-2.wav But it doesn't work. I get the whole file as a result. Well, almost the whole file. Somehow the resulting file has duration 1:16:31 instead of 1:17:23. Also I believe I executed this command in every possible way: with -ss and -t after -i, with -t specifying ending point, with mp3 files, with specifying audio codec, with ffmpeg. Am I doing it wrong?

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• Audio Panning using RtAudio

  - by user1801724

  I use the RtAudio library. I would like to implement an audio program where I can control the panning (e.g. shifting the sound from the left channel to the right channel). In my specific case, I use RtAudio in duplex mode (you can find an example here: duplex mode). It means that I link the microphone input to the speaker output. I have searched on the web, but I did not find anything useful. Should I apply a filter on the output buffer? What kind of filter?

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• Installing old Loki games on 12.04 64-bit results in no audio

  - by FlabbergastedPickle

  All, Here's an interesting problem. I followed instructions provided online for installing Loki Games' Heroes of Might and Magic 3 (see http://www.swanson.ukfsn.org/loki/ and http://wtanaka.com/node/7641) and got it installed and patched to the latest version. However, every time I start it regardless whether the pulseaudio is running, I get the following error: LD_LIBRARY_PATH=/usr/local/lib/Loki_Compat/ /usr/local/lib/Loki_Compat/ld-linux.so.2 /usr/local/games/Heroes3/heroes3.dynamic ALSA lib conf.c:3314:(snd_config_hooks_call) Cannot open shared library libasound_module_conf_pulse.so ALSA lib pcm.c:2217:(snd_pcm_open_noupdate) Unknown PCM default Couldn't open audio: My first soundcard is HDMI output and my second one is the actual soundcard (HP DM1 running 12.04 64-bit with latest updates). I did set up /etc/asound.conf as follows: asound.conf pcm.!default { type hw card 1 } ctl.!default { type hw card 1 } So, the default soundcard should work ok. Between Shadowgrounds that also stopped working and this it appears a there may be some unfinished business/regressions in 32-bit support on 64-bit systems in 12.04. Any thoughts?

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