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  • Automatic music rating based on listening habits

    - by marco92w
    I've created a Winamp-like music player in Delphi. Not so complex, of course. Just a simple one. But now I would like to add a more complex feature: Songs in the library should be automatically rated based on the user's listening habits. This means: The application should "understand" if the user likes a song or not. And not only whether he/she likes it but also how much. My approach so far (data which could be used): Simply measure how often a song was played per time. Start counting time when the song was added to the library so that recent songs don't have any disadvantage. Measure how long a song was played on average (minutes). Starting a song but directly change to another one should have a bad influence on the ranking since the user didn't seem to like the song. ... Could you please help me with this problem? I would just like to have some ideas. I don't need the implementation in Delphi.

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  • How would i down-sample a .wav file then reconstruct it using nyquist? - in MATLAB

    - by Andrew
    This is all done in MATLAB 2010 My objective is to show the results of: undersampling, nyquist rate/ oversampling First i need to downsample the .wav file to get an incomplete/ or impartial data stream that i can then reconstuct. Heres the flow chart of what im going to be doing So the flow is analog signal - sampling analog filter - ADC - resample down - resample up - DAC - reconstruction analog filter what needs to be achieved: F= Frequency F(Hz=1/s) E.x. 100Hz = 1000 (Cyc/sec) F(s)= 1/(2f) Example problem: 1000 hz = Highest frequency 1/2(1000hz) = 1/2000 = 5x10(-3) sec/cyc or a sampling rate of 5ms This is my first signal processing project using matlab. what i have so far. % Fs = frequency sampled (44100hz or the sampling frequency of a cd) [test,fs]=wavread('test.wav'); % loads the .wav file left=test(:,1); % Plot of the .wav signal time vs. strength time=(1/44100)*length(left); t=linspace(0,time,length(left)); plot(t,left) xlabel('time (sec)'); ylabel('relative signal strength') **%this is were i would need to sample it at the different frequecys (both above and below and at) nyquist frequency.*I think.*** soundsc(left,fs) % shows the resaultant audio file , which is the same as original ( only at or above nyquist frequency however) Can anyone tell me how to make it better, and how to do the sampling at verious frequencies?

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  • Delphi: Error when starting MCI

    - by marco92w
    I use the TMediaPlayer component for playing music. It works fine with most of my tracks. But it doesn't work with some tracks. When I want to play them, the following error message is shown: Which is German but roughly means that: In the project pMusicPlayer.exe an exception of the class EMCIDeviceError occurred. Message: "Error when starting MCI.". Process was stopped. Continue with "Single Command/Statement" or "Start". The program quits directly after calling the procedure "Play" of TMediaPlayer. This error occurred with the following file for example: file size: 7.40 MB duration: 4:02 minutes bitrate: 256 kBit/s I've encoded this file with a bitrate of 128 kBit/s and thus a file size of 3.70 MB: It works fine! What's wrong with the first file? Windows Media Player or other programs can play it without any problems. Is it possible that Delphi's TMediaPlayer cannot handle big files (e.g. 5 MB) or files with a high bitrate (e.g. 128 kBit/s)? What can I do to solve the problem?

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  • OpenAL not playing on Max OS X 10.6

    - by Grimless
    I've been working on getting a basic audio engine running on my Mac using OpenAL. It seems relatively straightforward after working with OpenGL for a while. However, despite the fact that I believe I have everything in place, my sound will not play. Here is the order of things I am doing: //Creating a new device ALCdevice* device = alcOpenDevice(NULL); //Create a new context with the device ALCcontext* context = alcCreateContext(device, NULL); //Make that context current alcMakeContextCurrent(context); //Do lots of loading stuff to bring in an AIFF... voodooAIFF = myAIFFLoader("name"); //Then use that data ALuint buf; alGenBuffers(1, &buf); //Check for errors, but none happen... //Bind buffer data. alBufferData(buf, voodooAIFF.format, voodooAIFF.data, voodooAIFF.sizeInBytes, voodooAIFF.frequency); //Check for errors, none here either... //Create Source ALuint src; alGenSources(1, &src); //Error check again, no errors. //Bind source to buffer alSourcei(src, AL_BUFFER, buf); //Set reference distance alSourcei(sourceID, AL_REFERENCE_DISTANCE, 1); //Set source attributes including gain and pitch to 1 (direction set to 0,0,0) //Check for errors, nothing... //Set up listener attributes. //Check for errors, no errors. //Begin playing. alSourcePlay(src); Observe silence... Any insight, what steps am I missing here?

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  • Difficulty porting raw PCM output code from Java to Android AudioTrack API.

    - by IndigoParadox
    I'm attempting to port an application that plays chiptunes (NSF, SPC, etc) music files from Java SE to Android. The Android API seems to lack the javax multimedia classes that this application uses to output raw PCM audio. The closest analog I've found in the API is AudioTrack and so I've been wrestling with that. However, when I try to run one of my sample music files through my port-in-progress, all I get back is static. My suspicion is that it's the AudioTrack I've setup which is at fault. I've tried various different constructors but it all just outputs static in the end. The DataLine setup in the original code is something like: AudioFormat audioFormat = new AudioFormat( AudioFormat.Encoding.PCM_SIGNED, 44100, 16, 2, 4, 44100, true ); DataLine.Info lineInfo = new DataLine.Info( SourceDataLine.class, audioFormat ); DataLine line = (SourceDataLine)AudioSystem.getLine( lineInfo ); The constructor I'm using right now is: AudioTrack = new AudioTrack( AudioManager.STREAM_MUSIC, 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, AudioTrack.getMinBufferSize( 44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT ), AudioTrack.MODE_STREAM ); I've replaced constants and variables in those so they make sense as concisely as possible, but my basic question is if there are any obvious problems in the assumptions I made when going from one format to the other.

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  • Is there a best practice for concatenating MP3 Files, adjusting sample rates to match, while preserving original files?

    - by Scott
    Hello overflow community! Does anyone know if there is a "best practice" to concatenate mp3 files to create new files, while preserving the original files? I am working on a CentOS Linux machine, in command line. I will eventually call the command line from a PHP script. I have been doing research and I have come up with a process that I think could work. It combines general advice from different forums, blogs, and sources like this one. So here I go: Create a temporary folder Loop through files to create a new, converted copy, of file into a "raw" format (which one, I don't know. I didn't know "raw" files existed before too long ago. I could use some suggestions on this) Store the path to the temporary files, in the temporary folder, and then loop through the files to concatenate them and then put the new merged file the final "processed directory" Delete the contents of the temporary file with the temporary raw files inside. Convert the final file from "raw" to mp3 and enjoy the finished result I'm thinking that this course of action might be best because I can't necessarily control the quality of the original "source" mp3s. The only other option I could think of would be to create a script that would perform a similar process upon files being added to the system leaving only the files with the "proper" format and removing the original "erroneous" file. Hopefully you can see that I have put some thought into this and that I'm trying to leverage the collective knowledge of this community to choose the best direction. Perhaps there is a better path that I could take? By concatenate, I mean to join together in sequence to create a new audio file from the "concatenated files."

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  • To display an album art from media store in android

    - by user1834724
    I'm not able to display album art from media store while listing albums,I'm getting following error Bad request for field slot 0,-1. numRows = 32, numColumns = 7 01-02 02:48:16.789: D/AndroidRuntime(4963): Shutting down VM 01-02 02:48:16.789: W/dalvikvm(4963): threadid=1: thread exiting with uncaught exception (group=0x4001e578) 01-02 02:48:16.804: E/AndroidRuntime(4963): FATAL EXCEPTION: main 01-02 02:48:16.804: E/AndroidRuntime(4963): java.lang.IllegalStateException: get field slot from row 0 col -1 failed Can anyone kindly help with this issue,Thanks in advance public class AlbumbsListActivity extends Activity { private ListAdapter albumListAdapter; private HashMap<Integer, Integer> albumInfo; private HashMap<Integer, Integer> albumListInfo; private HashMap<Integer, String> albumListTitleInfo; private String audioMediaId; private static final String TAG = "AlbumsListActivity"; Boolean showAlbumList = false; Boolean AlbumListTitle = false; ImageView album_art ; public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.albums_list_layout); Cursor cursor; ContentResolver cr = getApplicationContext().getContentResolver(); if (getIntent().hasExtra(Util.ALBUM_ID)) { int albumId = getIntent().getIntExtra(Util.ALBUM_ID, Util.MINUS_ONE); String[] projection = new String[] { Albums._ID, Albums.ALBUM, Albums.ARTIST, Albums.ALBUM_ART, Albums.NUMBER_OF_SONGS }; String selection = null; String[] selectionArgs = null; String sortOrder = Media.ALBUM + " ASC"; cursor = cr.query(Albums.EXTERNAL_CONTENT_URI, projection, selection, selectionArgs, sortOrder); /* final String[] ccols = new String[] { //MediaStore.Audio.Albums., MediaStore.Audio.Albums._ID, MediaStore.Audio.Albums.ALBUM, MediaStore.Audio.Albums.ARTIST, MediaStore.Audio.Albums.ALBUM_ART, MediaStore.Audio.Albums.NUMBER_OF_SONGS }; cursor = cr.query(MediaStore.Audio.Albums.getContentUri( "external"), ccols, null, null, MediaStore.Audio.Albums.DEFAULT_SORT_ORDER);*/ showAlbumList = true; } else { String order = MediaStore.Audio.Albums.ALBUM + " ASC"; String where = MediaStore.Audio.Albums.ALBUM; cursor = managedQuery(Media.EXTERNAL_CONTENT_URI, DbUtil.projection, null, null, order); showAlbumList = false; } albumInfo = new HashMap<Integer, Integer>(); albumListInfo = new HashMap<Integer, Integer>(); ListView listView = (ListView) findViewById(R.id.mylist_album); listView.setFastScrollEnabled(true); listView.setOnItemLongClickListener(new ItemLongClickListener()); listView.setAdapter(new AlbumCursorAdapter(this, cursor, DbUtil.displayFields, DbUtil.displayViews,showAlbumList)); final Uri uri = MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI; final Cursor albumListCursor = cr.query(uri, DbUtil.Albumprojection, null, null, null); } private class AlbumCursorAdapter extends SimpleCursorAdapter implements SectionIndexer{ private final Context context; private final Cursor cursorValues; private Time musicTime; private Boolean isAlbumList; private MusicAlphabetIndexer mIndexer; private int mTitleIdx; public AlbumCursorAdapter(Context context, Cursor cursor, String[] from, int[] to,Boolean isAlbumList) { super(context, 0, cursor, from, to); this.context = context; this.cursorValues = cursor; //musicTime = new Time(); this.isAlbumList = isAlbumList; } String albumName=""; String artistName = ""; String numberofsongs = ""; long albumid; @Override public View getView(int position, View convertView, ViewGroup parent) { View rowView = convertView; if (rowView == null) { LayoutInflater inflater = (LayoutInflater) context .getSystemService(Context.LAYOUT_INFLATER_SERVICE); rowView = inflater .inflate(R.layout.row_album_layout, parent, false); } this.cursorValues.moveToPosition(position); String title = ""; String artistName = ""; String albumName = ""; int count; long albumid = 0; String songDuration = ""; if (isAlbumList) { albumInfo.put( position, Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums._ID)))); artistName = this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ARTIST)); albumName = this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ALBUM)); albumid=Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Albums.ALBUM_ID))); } else { albumInfo.put(position, Integer.parseInt(this.cursorValues .getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media._ID)))); artistName = this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ARTIST)); albumName = this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ALBUM)); albumid=Integer.parseInt(this.cursorValues.getString(this.cursorValues .getColumnIndex(MediaStore.Audio.Media.ALBUM_ID))); } //code for Alphabetical Indexer mTitleIdx = cursorValues.getColumnIndex(MediaStore.Audio.Media.ALBUM); mIndexer = new MusicAlphabetIndexer(cursorValues, mTitleIdx, getResources().getString(R.string.fast_scroll_alphabet)); //end TextView metaone = (TextView) rowView.findViewById(R.id.album_name); TextView metatwo = (TextView) rowView.findViewById(R.id.artist_name); ImageView metafour = (ImageView) rowView.findViewById(R.id.album_art); TextView metathree = (TextView) rowView .findViewById(R.id.songs_count); metaone.setText(albumName); metatwo.setText(artistName); (metafour)getAlbumArt(albumid); System.out.println("albumid----------"+albumid); metaThree.setText(DbUtil.makeTimeString(context, secs)); getAlbumArt(albumid); } TextView metaone = (TextView) rowView.findViewById(R.id.album_name); TextView metatwo = (TextView) rowView.findViewById(R.id.artist_name); album_art = (ImageView) rowView.findViewById(R.id.album_art); //TextView metathree = (TextView) rowView.findViewById(R.id.songs_count); metaone.setText(albumName); metatwo.setText(artistName); return rowView; } } String albumArtUri = ""; private void getAlbumArt(long albumid) { Uri uri=ContentUris.withAppendedId(MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI, albumid); System.out.println("hhhhhhhhhhh" + uri); Cursor cursor = getContentResolver().query( ContentUris.withAppendedId( MediaStore.Audio.Albums.EXTERNAL_CONTENT_URI, albumid), new String[] { MediaStore.Audio.AlbumColumns.ALBUM_ART }, null, null, null); if (cursor.moveToFirst()) { albumArtUri = cursor.getString(0); } System.out.println("kkkkkkkkkkkkkkkkkkk :" + albumArtUri); cursor.close(); if(albumArtUri != null){ Options opts = new Options(); opts.inJustDecodeBounds = true; Bitmap albumCoverBitmap = BitmapFactory.decodeFile(albumArtUri, opts); opts.inJustDecodeBounds = false; albumCoverBitmap = BitmapFactory.decodeFile(albumArtUri, opts); if(albumCoverBitmap != null) album_art.setImageBitmap(albumCoverBitmap); }else { // TODO: Options opts = new Options(); Bitmap albumCoverBitmap = BitmapFactory.decodeResource(getApplicationContext().getResources(), R.drawable.albumart_mp_unknown_list, opts); if(albumCoverBitmap != null) album_art.setImageBitmap(albumCoverBitmap); } } } }

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  • How can I send audio input as chunked HTTP?

    - by Noli
    I am trying to create an interface with an external server, and don't know where to start. I would need to take audio as input to my computer, and send it to the remote server as a chunked HTTP request. The api that i'm trying to connect to is described here p1-5 http://dragonmobile.nuancemobiledeveloper.com/public/Help/HttpInterface/HTTP_Services_for_NDEV_v1.2_Silver_Version.pdf I have never worked with audio programmatically, so don't know what would be the most straighforward way to go about this? Are there solutions that exist out there that already do this? I've come across references to Shoutcast, VLC, Icecast, FFMPeg, Darkice, but I don't know if those are appropriate for what I'm trying to accomplish or not. Would appreciate any guidance, Thanks

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  • Can Google Translate's audio files be used in a game?

    - by ashes999
    For my game, I need text-to-speech. Since it's Android, I decided to settle for MP3s, since the range of words spoken is few. For my prototype, I'm using Google Translate to generate the audio since it has awesome pronounciation across multiple languages. But can I use it in production? What if I sell my game for $1 on the app store? All I can find on SE is that the API may be LGPL, and that the licensing page mentions the API is only available for academic research -- nothing more. My usage is a bit different; I'm actually capturing the audio bits and using those instead. I'm curious to know the license for this; I can't find anything with my Google-fu.

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  • CPU spikes cause audio stuttering in Audacious when browsing? (Lubuntu)

    - by Alucai Vivorvel
    My default audio player is Audacious, browser Google Chrome. I tried Firefox, and while I love it, the CPU load spikes when doing something as simple and small and switching a tab, which causes the audio playing to stutter (as sound is onboard and handled thru the CPU). Chrome doesn't do this as much, but there is the occasional stuttering when browsing, which is ridiculous, as not even Windows Vista does this. So I thought maybe it's something to do with how Lubuntu handles sound, I checked and only ALSA was installed. I tried installing PulseAudio, but, while the music "plays", nothing comes through the speakers. Immediately after switching back to ALSA the music pours out of them. So I was wondering if you had any idea what was going on here. I asked on Ubuntu Forums but apparently my problem is too complex, as it's been over a week since the last reply. Specs are: AMD Athlon 64 3200+ @ 2GHz 2GB Corsair 667MHz DDR2 RAM ATi HD Radeon 3650 (AGP) 512MB 500W Cooler Master PSU 80GB SATA II HDD (Vista is installed on 500GB drive) Biostar K8M800 Motherboard

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  • How to install/configure ffmpeg to compress mp4 videos for flash player delivery?

    - by Andrew Fulton
    We have a flash web-app that created interactive video, and are using ffmpeg to do some compression/resizing when a user "publishes" their project. The user can upload flv files and mp4 files, both of which play fine in the Flash UI before publishing. After publishing the flv files work fine, but the mp4 files will not play in the flash player: Audio will play but video won't. The mp4 files will play fine if I download them and play them in the Quicktime player but if I attempt to open them in the Adobe Media Player it reports "The media file does not contain a supported video track". If I open the Movie inspector in quicktime it tells me that the original file is an "h264" video and the ffmpeg-processed ones are "mpeg-4". I have tried forcing it to h264 by adding flags like -f h264 and -vcodec h264 but I get a screenfull of errors (no frame, illegal POC type, sps_id out of range) ending with Could not find codec parameters (Video: h264) h264 will show up if I run ffmpeg -formats and ffmpeg -codecs, and as I said it will play fine in Quicktime. Is there anything else I need to do to convince the flash player to play them? Is there anything else I need to tell you about the server that will help?

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  • Does VLC Player work well on Windows 7 64-bit?

    - by ????
    I tried VLC Player on Windows 7 64-bit version and the playback was fine, but when the video is maximized the image is very pixelated. I tried this on a Radeon HD 2600 XT graphics card as well as a computer with an Intel graphics chipset and both have the same result. If I use VLC Player on Windows 7 32-bit instead of 64-bit, there is no pixelation. Is this a known problem or is there any method to fix it on Windows 7 64-bit? Update: there isn't a 32-bit vs 64-bit version of VLC player, is there? (unlike 7-Zip) I also tried GOM Player and it doesn't have the problem on Windows 7 64-bit. Update: Nov 4, 2009 VLC displays an update notice: VLC 1.0.3 is a minor release fixing many bugs, especially for Windows Vista and 7, but it also introduces 2 new modes for deinterlacing, and a new udev module. Major fixes are about WMA Pro support, Dolby tracks in 4.0, v4l/v4l2 and atsc and a crash in mjpeg demuxer. Update of translations are also part of this release.

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  • How to overlay audio file on .wmv video file using c#?

    - by Vipul jain
    Hello, I want to record video and audio files using C#. After recording of audio + video i want to merge them. There can be only one video file and 10 audio file. I want this ten files to overlay on one video file. I am assure that i want video file in .wmv format. Can you tell me i should record audios in which format so later i can overlay those audio files on .wmv format video file? Also please let me know how to overlay audio file on .wmv video file? Hope i will get prompt reply for this

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  • Beat Detection on iPhone with wav files and openal

    - by Dmacpro
    Using this website i have tried to make a beat detection engine. http://www.gamedev.net/reference/articles/article1952.asp { ALfloat energy = 0; ALfloat aEnergy = 0; ALint beats = 0; bool init = false; ALfloat Ei[42]; ALfloat V = 0; ALfloat C = 0; ALshort *hold; hold = new ALshort[[myDat length]/2]; [myDat getBytes:hold length:[myDat length]]; ALuint uiNumSamples; uiNumSamples = [myDat length]/4; if(alDatal == NULL) alDatal = (ALshort *) malloc(uiNumSamples*2); if(alDatar == NULL) alDatar = (ALshort *) malloc(uiNumSamples*2); for (int i = 0; i < uiNumSamples; i++) { alDatal[i] = hold[i*2]; alDatar[i] = hold[i*2+1]; } energy = 0; for(int start = 0; start<(22050*10); start+=512){ //detect for 10 seconds of data for(int i = start; i<(start+512); i++){ energy+= fabs(alDatal[i]) + fabs(alDatar[i]); } aEnergy = 0; for(int i = 41; i>=0; i--){ if(i ==0){ Ei[0] = energy; } else { Ei[i] = Ei[i-1]; } if(start >= 21504){ aEnergy+=Ei[i]; } } aEnergy = aEnergy/43.f; if (start >= 21504) { for(int i = 0; i<42; i++){ V += (Ei[i]-aEnergy); } V = V/43.f; C = (-0.0025714*V)+1.5142857; init = true; if(energy >(C*aEnergy)) beats++; } } } alDatal and alDatar are (short*) type; myDat is NSdata that holds the actual audio data of a wav file formatted to 22050 khz and 16 bit stereo. This doesn't seem to work correctly. If anyone could help me out that would be amazing. I've been stuck on this for 3 days.

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  • Has a multi player graphic adventure* ever been made?

    - by Petruza
    By graphic adventure, I mean point & click LucasArts-type games. Those games have a mostly linear structure in nature, and usually don't offer as many variants as other games types like action, rpg, strategy, which makes this genre difficult to implement a multi-player feature. I'd like to know if there has been any attempts on doing such a thing, and if it would be viable, as players going offline or leaving a game in the middle would affect significantly the other players' game.

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  • How to properly do weapon cool-down reload timer in multi-player laggy environment?

    - by John Murdoch
    I want to handle weapon cool-down timers in a fair and predictable way on both client on server. Situation: Multiple clients connected to server, which is doing hit detection / physics Clients have different latency for their connections to server ranging from 50ms to 500ms. They want to shoot weapons with fairly long reload/cool-down times (assume exactly 10 seconds) It is important that they get to shoot these weapons close to the cool-down time, as if some clients manage to shoot sooner than others (either because they are "early" or the others are "late") they gain a significant advantage. I need to show time remaining for reload on player's screen Clients can have clocks which are flat-out wrong (bad timezones, etc.) What I'm currently doing to deal with latency: Client collects server side state in a history, tagged with server timestamps Client assesses his time difference with server time: behindServerTimeNs = (behindServerTimeNs + (System.nanoTime() - receivedState.getServerTimeNs())) / 2 Client renders all state received from server 200 ms behind from his current time, adjusted by what he believes his time difference with server time is (whether due to wrong clocks, or lag). If he has server states on both sides of that calculated time, he (mostly LERP) interpolates between them, if not then he (LERP) extrapolates. No other client-side prediction of movement, e.g., to make his vehicle seem more responsive is done so far, but maybe will be added later So how do I properly add weapon reload timers? My first idea would be for the server to send each player the time when his reload will be done with each world state update, the client then adjusts it for the clock difference and thus can estimate when the reload will be finished in client-time (perhaps considering also for latency that the shoot message from client to server will take as well?), and if the user mashes the "shoot" button after (or perhaps even slightly before?) that time, send the shoot event. The server would get the shoot event and consider the time shot was made as the server time when it was received. It would then discard it if it is nowhere near reload time, execute it immediately if it is past reload time, and hold it for a few physics cycles until reload is done in case if it was received a bit early. It does all seem a bit convoluted, and I'm wondering whether it will work (e.g., whether it won't be the case that players with lower ping get better reload rates), and whether there are more elegant solutions to this problem.

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  • 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

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  • Has a multi player graphic adventure* ever been made?

    - by Petruza
    By graphic adventure, I mean point & click LucasArts-type games. Those games have a mostly linear structure in nature, and usually don't offer as many variants as other games types like action, rpg, strategy, which makes this genre difficult to implement a multi-player feature. I'd like to know if there has been any attempts on doing such a thing, and if it would be viable, as players going offline or leaving a game in the middle would affect significantly the other players' game.

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  • How to add energy ball which disappears when touched by player in UDK?

    - by OliveOne
    I am new to UDK and learning game development. I want to know about how to add a ball to the game world with the following effects/actions: Glowing effect Physics-like object (just having gravity) Particles when touched by player-avatar Disappears in 1-2 seconds after touch Score updates based on different colors of ball I know little about this can be done by kismet, cascade and content creation, but do not know where to start. Please tell me the steps for this. I am trying this weekend in depth.

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  • How to get the blocks seen by the player?

    - by m4tx
    I'm writing a Minecraft-like game using Ogre engine and I have a problem. I must optimize my game, because when I try draw 10000 blocks, I have 2 FPS... So, I got the idea that blocks display of the plane and to hide the invisible blocks. But I have a problem - how do I know which blocks at a time are visible to the player? And - if you know of other optimization methods for such a game, write what and how to use them in Ogre.

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  • How to get the blocks seen by the player?

    - by m4tx
    I'm writing a Minecraft-like game using Ogre engine and I have a problem. I must optimize my game, because when I try draw 10000 blocks, I have 2 FPS... So, I got the idea that blocks display of the plane and to hide the invisible blocks. But I have a problem - how do I know which blocks at a time are visible to the player? And - if you know of other optimization methods for such a game, write what and how to use them in Ogre.

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  • In what kind of variable type is the player position stored on a MMORPG such as WoW?

    - by jokoon
    I even heard J. Carmack quickly talk about it... How a software can track a player's position so accurately, being on a such huge world, without loading between zones, and on a multiplayer scale ? How is the data formatted when it passes through the netcode ? I can understand how vertices are stored into the graphic card's memory, but when it comes to synchronize the multiplayer, I can't imagine what is best.

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  • How to obtain a camera stream from Unity without rendering it to the player's screen?

    - by aiguy
    I'd like to stream the output of two cameras to a separate process. Right now, it looks like the best way to do that is to grab the rendered camera views from the screen via platform specific screen capture hooks then compress them real time with h.264. Is there a way to grab the input of the cameras within unity and avoid rendering them to the screen? One solution I'm considering involves using Unity's multiplayer capability to run the game on a separate machine and grab it from that screen buffer, unbeknownst to the player.

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  • FMOD.net streaming, callback and exinfo parameters

    - by Tesserex
    I posted a question on gamedev about how to play nsf files (NES console music) in FMOD. It didn't get any results, but since then I made some progress. I decided that the easiest method was just to compile an existing player into a dll and then call it from C# to populate my buffer. The problem now is getting it to sound right, and making sure all my paremeters are correct. Here are the facts so far: The nsf dll is dealing with shorts, so the data is PCM16. The sample nsf I'm using has a playback rate of 60 Hz. Just for playing around now, I'm using a frequency of 48000. Based on 2 and 3, the dll calculates a necessary buffer size of 48000 / 60hz = 800. This means it will render 800 shorts worth of buffer for every simulated NES frame. I've so far got my C# code to play the nsf, at the correct pitch and tempo, but it's very grainy / fuzzy, which I'm attributing to the fact that the FMOD read callback is giving a data length of 1600, whereas I should be expecting 800. I've tried playing around with all the numbers and it either crashes, or the music changes pitch, tempo, or both. Here's some of my C# code: uint channels = 1, frequency = 48000; FMOD.MODE mode = (FMOD.MODE.DEFAULT | FMOD.MODE.OPENUSER | FMOD.MODE.LOOP_NORMAL); FMOD.Sound sound = new FMOD.Sound(); FMOD.CREATESOUNDEXINFO ex = new FMOD.CREATESOUNDEXINFO(); ex.cbsize = Marshal.SizeOf(ex); ex.fileoffset = 0; ex.format = FMOD.SOUND_FORMAT.PCM16; // does this even matter? It doesn't change my results as long as it's long enough for one update ex.length = frequency; ex.numchannels = (int)channels; ex.defaultfrequency = (int)frequency; ex.pcmreadcallback = pcmreadcallback; ex.dlsname = null; // eventually I will calculate this with frequency / nsf hz, but I'm just testing for now ex.decodebuffersize = 800; // from the dll load_nsf_file("file.nsf", 8, (int)frequency); // 8 is the track number to play var result = system.createSound( (string)null, (mode | FMOD.MODE.CREATESTREAM), ref ex, ref sound); channel = new FMOD.Channel(); result = system.playSound(FMOD.CHANNELINDEX.FREE, sound, false, ref channel); private FMOD.RESULT PCMREADCALLBACK(IntPtr soundraw, IntPtr data, uint datalen) { // from the dll process_buffer(data, (int)800); // if I use datalen, it usually crashes (I can't get datalen to = 800 safely) return FMOD.RESULT.OK; } So here are some of my questions: What is the relationship between exinfo.decodebuffersize, frequency, and the datalen parameter of the read callback? With this code sample, it's coming in as 3200. I don't know where that factor of 4 between it and the decodebuffersize comes from. Is datalen in the callback referring to number of bytes, or shorts? The process_buffer function takes a short array and its length. I would expect fmod is talking about shorts as well because I told it PCM16. Maybe my playback quality is bad for some totally different reason. If so I have no idea where to begin solving that. Any ideas there?

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  • Relationship Modelling in Core Data

    - by Stevie
    Hi there, I'm fairly new to Objective C and Core Data and have a problem designing a case where players team up one-on-one and have multiple matches that end up with a specific result. With MySQL, I would have a Player table (player primary key, name) and a match table (player A foreign key, player B foreign key, result). Now how do I do this with Core Data? I can easily tie a player entity to a match entity using a relationship. But how do I model the inverse direction for the second player ref. in the match entity? Player Name: Attribute Match: Relationship Match Match Result: Attribute PlayerA: Relationship to Player (<- Inverse to Player.Match) PlayerB: Relationship to Player (<- Inverse to ????) Would be great if someone could give me an idea on this! Thanks, Stevie.

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