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  • How large is a "buffer" in PostgreSQL

    - by Konrad Garus
    I am using pg_buffercache module for finding hogs eating up my RAM cache. For example when I run this query: SELECT c.relname, count(*) AS buffers FROM pg_buffercache b INNER JOIN pg_class c ON b.relfilenode = c.relfilenode AND b.reldatabase IN (0, (SELECT oid FROM pg_database WHERE datname = current_database())) GROUP BY c.relname ORDER BY 2 DESC LIMIT 10; I discover that sample_table is using 120 buffers. How much is 120 buffers in bytes?

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  • Protocol buffer deserialization and a dynamically loaded DLL in Compact Framework

    - by cloudraven
    I saw a question related to this on the full framework here. Since it seems to have stayed unresolved for quite a while and this is for the compact framework, I though it would be better to create a new question for it. I want to deserialize types for which I am loading assemblies dynamically (with Assembly.LoadFrom) and I am getting a "Unable to identify known-type for ProtoIncludeAttribute" error. In the related question I mentioned, it was hinted that hooking AppDomain.AssemblyResolve event would help solving the problem. It makes sense for the full framework, but that event is not available in the CF. I wonder if there is a way to do this with CF. The structures I am using look a lot like this and all the classes required for deserialization are loaded from the same Assembly. If the assembly is referenced instead of dynamically loaded it works fine, but fails if done dynamically.

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  • OpenAL device, buffer and context relationship

    - by Markus
    I'm trying to create an object oriented model to wrap OpenAL and have a little problem understanding the devices, buffers and contexts. From what I can see in the Programmer's Guide, there are multiple devices, each of which can have multiple contexts as well as multiple buffers. Each context has a listener, and the alListener*() functions all operate on the listener of the active context. (Meaning that I have to make another context active first if I wanted to change it's listener, if I got that right.) So far, so good. What irritates me though is that I need to pass a device to the alcCreateContext() function, but none to alGenBuffers(). How does this work then? When I open multiple devices, on which device are the buffers created? Are the buffers shared between all devices? What happens to the buffers if I close all open devices? (Or is there something I missed?)

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  • TCP and UDP are using different OS Buffer?

    - by Jack
    HI all. Here is the scenario. I have port 8888 for my program to use. I build a TCP and a UDP listener on that port. (This can do, c# allows, because they are two different protocols) My question is If the network traffic is very busy, TCP sockets may refuse or signalling the other end to stop sending things, it is called congestion control, right? So if TCP is congestion controlling, other ends may not send more data, in this "TCP quiet period", UDP channel should have not that much of traffic, right? I want to figure out the TCP traffic will affect UDP traffic or not?

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  • Copying a byte buffer with JNI

    - by Daniel
    I've found plenty of tutorials / questions on Stackoverflow that deal with copying char arrays from C/JNI side into something like a byte[] in Java, but not the other way around. I am using a native C library which expects a byte array. I simply want to get data from a byte[] in java, into preferably an unsigned char[] in C. Long story short: What is the best way of copying data from a jBytearray in JNI? Is there any way to detect it's size?

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  • ctypes buffer modification

    - by Chris
    Hi, I need to call a c library from my python code. The c library does a lot of image manipulation, so I am passing it image buffers allocated using create_string_buffer. The problem is that I also need to manipulate and change these buffers. What is the best way to reach in and twiddle individual values in my buffers? The buffers are all uint8 buffers. Thanks!

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  • Boost::asio bug in MSVC10 - Failing BOOST_WORKAROUND in ~buffer_debug_check() in buffer.hpp

    - by shaz
    A straight compilation of example http://www.boost.org/doc/libs/1_43_0/doc/html/boost_asio/tutorial/tutdaytime3/src.html results in a runtime null pointer exception. Stack trace points to the buffer_debug_check destructor which contains this comment: // MSVC's string iterator checking may crash in a std::string::iterator // object's destructor when the iterator points to an already-destroyed // std::string object, unless the iterator is cleared first. The test #if BOOST_WORKAROUND(BOOST_MSVC, = 1400) succeeds in MSVC10 and (but) results in a null pointer exception in c:\Program Files (x86)\Microsoft Visual Studio 10.0\VC\include\xutility line 123 _Iterator_base12& operator=(const _Iterator_base12& _Right) { // assign an iterator if (_Myproxy != _Right._Myproxy) _Adopt(_Right._Myproxy->_Mycont); return (*this); } _Right._Myproxy is NULL

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  • AudioRecord - empty buffer

    - by Arxas
    I' m trying to record some audio using AudioRecord class. Here is my code: int audioSource = AudioSource.MIC; int sampleRateInHz = 44100; int channelConfig = AudioFormat.CHANNEL_IN_MONO; int audioFormat = AudioFormat.ENCODING_PCM_16BIT; int bufferSizeInShorts = 44100; int bufferSizeInBytes = 2*bufferSizeInShorts; short Data[] = new short[bufferSizeInShorts]; Thread recordingThread; AudioRecord audioRecorder = new AudioRecord(audioSource, sampleRateInHz, channelConfig, audioFormat, bufferSizeInBytes); @Override protected void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); setContentView(R.layout.activity_main); } @Override public boolean onCreateOptionsMenu(Menu menu) { getMenuInflater().inflate(R.menu.activity_main, menu); return true; } public void startRecording(View arg0) { audioRecorder.startRecording(); recordingThread = new Thread(new Runnable() { public void run() { while (Data[bufferSizeInShorts-1] == 0) audioRecorder.read(Data, 0, bufferSizeInShorts); } }); audioRecorder.stop(); } Unfortunately my short array is empty after the recording is over. May I kindly ask you to help me figure out what's wrong?

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  • "Access violation reading location" troubles retrieveing buffer from directx

    - by numerical25
    Below is my code... ID3D10Texture2D *pBackBuffer; hr = mpSwapChain->GetBuffer(0, __uuidof(ID3D10Texture2D), (LPVOID*) &pBackBuffer); and I get the following error chp1.exe': Unloaded 'C:\Windows\SysWOW64\oleaut32.dll' First-chance exception at 0x757ce124 in chp1.exe: Microsoft C++ exception: _com_error at memory location 0x0018eeb0.. First-chance exception at 0x757ce124 in chp1.exe: Microsoft C++ exception: _com_error at memory location 0x0018edd0.. First-chance exception at 0x757ce124 in chp1.exe: Microsoft C++ exception: _com_error at memory location 0x0018ef1c.. The thread 'Win32 Thread' (0xfc4) has exited with code 0 (0x0). 'chp1.exe': Unloaded 'C:\Windows\SysWOW64\D3D10Ref.DLL' First-chance exception at 0x00b71894 in chp1.exe: 0xC0000005: Access violation reading location 0x00000000. Unhandled ex ception at 0x00b71894 in chp1.exe: 0xC0000005: Access violation reading location 0x00000000. It appears that the error occurs in the last parameter. &pBackBuffer. I added this single line of code and the error occurs.

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  • fill a buffer successively

    - by mkind
    i intend to fill a char-pointer array successively in a for-loop. the content to fill in is a integer so i need to cast. but i didn't get the result i want to.. for (i=0;i<max0;i++){ sprintf(buf, "%d", content[i]); } sprintf replaces the hole buf, but i want to append. for (i=0;i<max0;i++){ buf[i]=(char) contint[i] } but this isn't working too. it seems to me, i get ascii-code of the content[i].

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  • Why doesn't `stdin.read()` read entire buffer?

    - by Shookie
    I've got the following code: def get_input(self): """ Reads command from stdin, returns its JSON form """ json_string = sys.stdin.read() print("json string is: "+json_string) json_data =json.loads(json_string) return json_data It reads a json string that was sent to it from another process. The json is read from stdin. For some reason I get the following output: json string is: <Some json here> json string is: Traceback (most recent call last): File "/Users/Matan/Documents/workspace/ProjectSH/addonmanager/addon_manager.py", line 63, in <module> manager.accept_commands() File "/Users/Matan/Documents/workspace/ProjectSH/addonmanager/addon_manager.py", line 49, in accept_commands json_data = self.get_input() File "/Users/Matan/Documents/workspace/ProjectSH/addonmanager/addon_manager.py", line 42, in get_input json_data =json.loads(json_string) File "/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/json/__init__.py", line 338, in loads return _default_decoder.decode(s) File "/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/json/decoder.py", line 365, in decode obj, end = self.raw_decode(s, idx=_w(s, 0).end()) File "/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/json/decoder.py", line 383, in raw_decode raise ValueError("No JSON object could be decoded") So for some reason it reads an empty string from stdin instead of reading only the json. I've checked, and the code that writes to this process's stdin writes to it only once. What's wrong here?

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  • AudioTrack lag: obtainBuffer timed out

    - by BTR
    I'm playing WAVs on my Android phone by loading the file and feeding the bytes into AudioTrack.write() via the FileInputStream BufferedInputStream DataInputStream method. The audio plays fine and when it is, I can easily adjust sample rate, volume, etc on the fly with nice performance. However, it's taking about two full seconds for a track to start playing. I know AudioTrack has an inescapable delay, but this is ridiculous. Every time I play a track, I get this: 03-13 14:55:57.100: WARN/AudioTrack(3454): obtainBuffer timed out (is the CPU pegged?) 0x2e9348 user=00000960, server=00000000 03-13 14:55:57.340: WARN/AudioFlinger(72): write blocked for 233 msecs, 9 delayed writes, thread 0xba28 I've noticed that the delayed write count increases by one every time I play a track -- even across multiple sessions -- from the time the phone has been turned on. The block time is always 230 - 240ms, which makes sense considering a minimum buffer size of 9600 on this device (9600 / 44100). I've seen this message in countless searches on the Internet, but it usually seems to be related to not playing audio at all or skipping audio. In my case, it's just a delayed start. I'm running all my code in a high priority thread. Here's a truncated-yet-functional version of what I'm doing. This is the thread callback in my playback class. Again, this works (only playing 16-bit, 44.1kHz, stereo files right now), it just takes forever to start and has that obtainBuffer/delayed write message every time. public void run() { // Load file FileInputStream mFileInputStream; try { // mFile is instance of custom file class -- this is correct, // so don't sweat this line mFileInputStream = new FileInputStream(mFile.path()); } catch (FileNotFoundException e) {} BufferedInputStream mBufferedInputStream = new BufferedInputStream(mFileInputStream, mBufferLength); DataInputStream mDataInputStream = new DataInputStream(mBufferedInputStream); // Skip header try { if (mDataInputStream.available() > 44) mDataInputStream.skipBytes(44); } catch (IOException e) {} // Initialize device mAudioTrack = new AudioTrack(AudioManager.STREAM_MUSIC, ConfigManager.SAMPLE_RATE, AudioFormat.CHANNEL_CONFIGURATION_STEREO, AudioFormat.ENCODING_PCM_16BIT, ConfigManager.AUDIO_BUFFER_LENGTH, AudioTrack.MODE_STREAM); mAudioTrack.play(); // Initialize buffer byte[] mByteArray = new byte[mBufferLength]; int mBytesToWrite = 0; int mBytesWritten = 0; // Loop to keep thread running while (mRun) { // This flag is turned on when the user presses "play" while (mPlaying) { try { // Check if data is available if (mDataInputStream.available() > 0) { // Read data from file and write to audio device mBytesToWrite = mDataInputStream.read(mByteArray, 0, mBufferLength); mBytesWritten += mAudioTrack.write(mByteArray, 0, mBytesToWrite); } } catch (IOException e) { } } } } If I can get past the artificially long lag, I can easily deal with the inherit latency by starting my write at a later, predictable position (ie, skip past the minimum buffer length when I start playing a file).

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  • the PrintWriter servlet Buffer and displayind data from a jsp

    - by nabilaloui
    Hello all, I need really your help please. What I do is to build a table in html tags in my servlet then when trying to send this table to a servlet for the display using: Response.sendRedirect this did not work. I have an error but I don't know the cause. I search to how do it since I use: response.setContentType("text/html"); PrintWriter out = response.getWriter(); out.println("<...");.... thinks a lot for your help

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  • Can single-buffer blocking WSASend deliver partial data?

    - by CodeAngry
    I've pretty much always used send() with sockets and now I'm moving onto the WSA functions. With send(), I have a sendall() helper that ensured all data is delivered even if it didn't happen in one try and a partial send occurred on first call. So, instead of learning the hard way or over-complicating code when I don't have to, decided to ask you: Can a blocking WSASend() send partial data or does it send everything before it returns or fails? Or should I check the bytes sent vs. expected to send and keep at it until everything is delivered? ANSWER: Overlapped WSASend() does not send partial data but if it does, it means the connection has terminated. I've never encountered the case yet.

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  • Does the DMA Buffer Size should be same as UART FIFO size?

    - by ddpd
    I have written a driver for a UART in omap4460 panda board running on Linux platform.I have enabled DMA in FIFO mode in UART.My user application transfers 100 bytes of data from user space to kernel buffer(DMA buffer). As soon as the DMA channel is enabled, data from DMA buffer is copied to FIFO which is then transmitted to TSR of UART.Since my FIFO size is 64bytes,only 64 bytes is transmitted to TSR. What should I do to transfer remaining bytes from DMA buffer to FIFO?/ IS there any overflow occuring?

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  • Why do I get an exception when playing multiple sound instances?

    - by Boreal
    Right now, I'm adding a rudimentary sound engine to my game. So far, I am able to load in a WAV file and play it once, then free up the memory when I close the game. However, the game crashes with a nice ArgumentOutOfBoundsException when I try to play another sound instance. Specified argument was out of the range of valid values. Parameter name: readLength I'm following this tutorial pretty much exactly, but I still keep getting the aforementioned error. Here's my sound-related code. /// <summary> /// Manages all sound instances. /// </summary> public static class Audio { static XAudio2 device; static MasteringVoice master; static List<SoundInstance> instances; /// <summary> /// The XAudio2 device. /// </summary> internal static XAudio2 Device { get { return device; } } /// <summary> /// Initializes the audio device and master track. /// </summary> internal static void Initialize() { device = new XAudio2(); master = new MasteringVoice(device); instances = new List<SoundInstance>(); } /// <summary> /// Releases all XA2 resources. /// </summary> internal static void Shutdown() { foreach(SoundInstance i in instances) i.Dispose(); master.Dispose(); device.Dispose(); } /// <summary> /// Registers a sound instance with the system. /// </summary> /// <param name="instance">Sound instance</param> internal static void AddInstance(SoundInstance instance) { instances.Add(instance); } /// <summary> /// Disposes any sound instance that has stopped playing. /// </summary> internal static void Update() { List<SoundInstance> temp = new List<SoundInstance>(instances); foreach(SoundInstance i in temp) if(!i.Playing) { i.Dispose(); instances.Remove(i); } } } /// <summary> /// Loads sounds from various files. /// </summary> internal class SoundLoader { /// <summary> /// Loads a .wav sound file. /// </summary> /// <param name="format">The decoded format will be sent here</param> /// <param name="buffer">The data will be sent here</param> /// <param name="soundName">The path to the WAV file</param> internal static void LoadWAV(out WaveFormat format, out AudioBuffer buffer, string soundName) { WaveStream wave = new WaveStream(soundName); format = wave.Format; buffer = new AudioBuffer(); buffer.AudioData = wave; buffer.AudioBytes = (int)wave.Length; buffer.Flags = BufferFlags.EndOfStream; } } /// <summary> /// Manages the data for a single sound. /// </summary> public class Sound : IAsset { WaveFormat format; AudioBuffer buffer; /// <summary> /// Loads a sound from a file. /// </summary> /// <param name="soundName">The path to the sound file</param> /// <returns>Whether the sound loaded successfully</returns> public bool Load(string soundName) { if(soundName.EndsWith(".wav")) SoundLoader.LoadWAV(out format, out buffer, soundName); else return false; return true; } /// <summary> /// Plays the sound. /// </summary> public void Play() { Audio.AddInstance(new SoundInstance(format, buffer)); } /// <summary> /// Unloads the sound from memory. /// </summary> public void Unload() { buffer.Dispose(); } } /// <summary> /// Manages a single sound instance. /// </summary> public class SoundInstance { SourceVoice source; bool playing; /// <summary> /// Whether the sound is currently playing. /// </summary> public bool Playing { get { return playing; } } /// <summary> /// Starts a new instance of a sound. /// </summary> /// <param name="format">Format of the sound</param> /// <param name="buffer">Buffer holding sound data</param> internal SoundInstance(WaveFormat format, AudioBuffer buffer) { source = new SourceVoice(Audio.Device, format); source.BufferEnd += (s, e) => playing = false; source.Start(); source.SubmitSourceBuffer(buffer); // THIS IS WHERE THE EXCEPTION IS THROWN playing = true; } /// <summary> /// Releases memory used by the instance. /// </summary> internal void Dispose() { source.Dispose(); } } The exception occurs on line 156 when I am playing the sound: source.SubmitSourceBuffer(buffer);

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  • Limiting TCP sends with a "to-be-sent" queue and other design issues.

    - by Poni
    Hello all! This question is the result of two other questions I've asked in the last few days. I'm creating a new question because I think it's related to the "next step" in my understanding of how to control the flow of my send/receive, something I didn't get a full answer to yet. The other related questions are: http://stackoverflow.com/questions/3028376/an-iocp-documentation-interpretation-question-buffer-ownership-ambiguity http://stackoverflow.com/questions/3028998/non-blocking-tcp-buffer-issues In summary, I'm using Windows I/O Completion Ports. I have several threads that process notifications from the completion port. I believe the question is platform-independent and would have the same answer as if to do the same thing on a *nix, *BSD, Solaris system. So, I need to have my own flow control system. Fine. So I send send and send, a lot. How do I know when to start queueing the sends, as the receiver side is limited to X amount? Let's take an example (closest thing to my question): FTP protocol. I have two servers; One is on a 100Mb link and the other is on a 10Mb link. I order the 100Mb one to send to the other one (the 10Mb linked one) a 1GB file. It finishes with an average transfer rate of 1.25MB/s. How did the sender (the 100Mb linked one) knew when to hold the sending, so the slower one wouldn't be flooded? Another way to ask this: Can I get a "hold-your-sendings" notification from the remote side? Is it built-in in TCP or the so called "reliable network protocol" needs me to do so? Again, I have a loop with many sends to a remote server, and at some point, within that loop I'll have to determine if I should queue that send or I can pass it on to the transport layer (TCP). How do I do that? What would you do? Of course that when I get a completion notification from IOCP that the send was done I'll issue other pending sends, that's clear. Another design question related to this: Since I am to use a custom buffers with a send queue, and these buffers are being freed to be reused (thus not using the "delete" keyword) when a "send-done" notification has been arrived, I'll have to use a mutual exlusion on that buffer pool. Using a mutex slows things down, so I've been thinking; Why not have each thread have its own buffers pool, thus accessing it , at least when getting the required buffers for a send operation, will require no mutex, because it belongs to that thread only. The buffers pool is located at the thread local storage (TLS) level. No mutual pool implies no lock needed, implies faster operations BUT also implies more memory used by the app, because even if one thread already allocated 1000 buffers, the other one that is sending right now and need 1000 buffers to send something will need to allocated these to its own. This is a long question and I hope none got hurt (: Thank you all!

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  • How to set ReceiveBufferSize for UDPClient? or Does it make sense to set? C#

    - by Jack
    Hello all. I am implementing a UDP data transfer thing. I have several questions about UDP buffer. I am using UDPClient to do the UDP send / receive. and my broadband bandwidth is 150KB/s (bytes/s, not bps). I send out a 500B datagram out to 27 hosts 27 hosts send back 10KB datagram back if they receive. So, I should receive 27 responses, right? however, I only get averagely 8 - 12 instead. I then tried to reduce the size of the response down to 500B, yes, I receive all. A thought of mine is that if all 27 hosts send back 10KB response at almost same time, the incoming traffic will be 270KB/s (likely), that exceeds my incoming bandwidth so loss happens. Am I right? But I think even if the incoming traffic exceeds the bandwidth, is the Windows supposed to put the datagram in the buffer and wait for receive? I then suspect that maybe the ReceiveBufferSize of my UdpClient is too small? by default, it is 8092B?? I don't know whether I am all right at these points. Please give me some help.

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  • How can I eliminate latency in quicktime streamed video

    - by JJFeiler
    I'm prototyping a client that displays streaming video from a HaiVision Barracuda through a quicktime client. I've been unable to reduce the buffer size below 3.0 seconds... for this application, we need as low a latency as the network allows, and prefer video dropouts to delay. I'm doing the following: - (void)applicationDidFinishLaunching:(NSNotification *)aNotification { NSString *path = [[NSBundle mainBundle] pathForResource:@"haivision" ofType:@"sdp"]; NSError *error = nil; QTMovie *qtmovie = [QTMovie movieWithFile:path error:&error]; if( error != nil ) { NSLog(@"error: %@", [error localizedDescription]); } Movie movie = [qtmovie quickTimeMovie]; long trackCount = GetMovieTrackCount(movie); Track theTrack = GetMovieTrack(movie,1); Media theMedia = GetTrackMedia(theTrack); MediaHandler theMediaHandler = GetMediaHandler(theMedia); QTSMediaPresentationParams myPres; ComponentResult c = QTSMediaGetIndStreamInfo(theMediaHandler, 1,kQTSMediaPresentationInfo, &myPres); Fixed shortdelay = 1<<15; OSErr theErr = QTSPresSetInfo (myPres.presentationID, kQTSAllStreams, kQTSTargetBufferDurationInfo, &shortdelay ); NSLog(@"OSErr %d", theErr); [movieView setMovie:qtmovie]; [movieView play:self]; } I seem to be getting valid objects/structures all the way down to the QTSPres, though the ComponentResult and OSErr are both returning -50. The streaming video plays fine, but the buffer is still 3.0seconds. Any help/insight appreciated. J

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  • Is there a less painful way to GetBytes for a buffer not starting at 0?

    - by Earlz
    I am having to deal with raw bites in a project and I need to basically do something like this byte[] ToBytes(){ byte[] buffer=new byte[somelength]; byte[] tmp=new byte[2]; tmp=BitConverter.GetBytes(SomeShort); buffer[0]=tmp[0]; buffer[1]=tmp[1]; tmp=BitConverter.GetBytes(SomeOtherShort); buffer[2]=tmp[0]; buffer[3]=tmp[1]; } I feel like this is so wrong yet I can't find any better way of doing it. Is there an easier way?

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  • July, the 31 Days of SQL Server DMO’s – Day 29 (sys.dm_os_buffer_descriptors)

    - by Tamarick Hill
    The sys.dm_os_buffer_descriptors Dynamic Management View gives you a look into the data pages that are currently in your SQL Server buffer pool. Just in case you are not familiar with some of the internals to SQL Server and how the engine works, SQL Server only works with objects that are in memory (buffer pool). When an object such as a table needs to be read and it does not exist in the buffer pool, SQL Server will read (copy) the necessary data page(s) from disk into the buffer pool and cache it. Caching takes place so that it can be reused again and prevents the need of expensive physical reads. To better illustrate this DMV, lets query it against our AdventureWorks2012 database and view the result set. SELECT * FROM sys.dm_os_buffer_descriptors WHERE database_id = db_id('AdventureWorks2012') The first column returned from this result set is the database_id column which identifies the specific database for a given row. The file_id column represents the file that a particular buffer descriptor belongs to. The page_id column represents the ID for the data page within the buffer. The page_level column represents the index level of the data page. Next we have the allocation_unit_id column which identifies a unique allocation unit. An allocation unit is basically a set of data pages. The page_type column tells us exactly what type of page is in the buffer pool. From my screen shot above you see I have 3 distinct type of Pages in my buffer pool, Index, Data, and IAM pages. Index pages are pages that are used to build the Root and Intermediate levels of a B-Tree. A Data page would represent the actual leaf pages of a clustered index which contain the actual data for the table. Without getting into too much detail, an IAM page is Index Allocation Map page which track GAM (Global Allocation Map) pages which in turn track extents on your system. The row_count column details how many data rows are present on a given page. The free_space_in_bytes tells you how much of a given data page is still available, remember pages are 8K in size. The is_modified signifies whether or not a page has been changed since it has been read into memory, .ie a dirty page. The numa_node column represents the Nonuniform memory access node for the buffer. Lastly is the read_microsec column which tells you how many microseconds it took for a data page to be read (copied) into the buffer pool. This is a great DMV for use when you are tracking down a memory issue or if you just want to have a look at what type of pages are currently in your buffer pool. For more information about this DMV, please see the below Books Online link: http://msdn.microsoft.com/en-us/library/ms173442.aspx Follow me on Twitter @PrimeTimeDBA

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  • How do I draw a scene with 2 nested frames

    - by Guido Granobles
    I have been trying for long time to figure out this: I have loaded a model from a directx file (I am using opengl and Java) the model have a hierarchical system of nested reference frames (there are not bones). There are just 2 frames, one of them is called x3ds_Torso and it has a child frame called x3ds_Arm_01. Each one of them has a mesh. The thing is that I can't draw the arm connected to the body. Sometimes the body is in the center of the screen and the arm is at the top. Sometimes they are both in the center. I know that I have to multiply the matrix transformation of every frame by its parent frame starting from the top to the bottom and after that I have to multiply every vertex of every mesh by its final transformation matrix. So I have this: public void calculeFinalMatrixPosition(Bone boneParent, Bone bone) { System.out.println("-->" + bone.name); if (boneParent != null) { bone.matrixCombined = bone.matrixTransform.multiply(boneParent.matrixCombined); } else { bone.matrixCombined = bone.matrixTransform; } bone.matrixFinal = bone.matrixCombined; for (Bone childBone : bone.boneChilds) { calculeFinalMatrixPosition(bone, childBone); } } Then I have to multiply every vertex of the mesh: public void transformVertex(Bone bone) { for (Iterator<Mesh> iterator = meshes.iterator(); iterator.hasNext();) { Mesh mesh = iterator.next(); if (mesh.boneName.equals(bone.name)) { float[] vertex = new float[4]; double[] newVertex = new double[3]; if (mesh.skinnedVertexBuffer == null) { mesh.skinnedVertexBuffer = new FloatDataBuffer( mesh.numVertices, 3); } mesh.vertexBuffer.buffer.rewind(); while (mesh.vertexBuffer.buffer.hasRemaining()) { vertex[0] = mesh.vertexBuffer.buffer.get(); vertex[1] = mesh.vertexBuffer.buffer.get(); vertex[2] = mesh.vertexBuffer.buffer.get(); vertex[3] = 1; newVertex = bone.matrixFinal.transpose().multiply(vertex); mesh.skinnedVertexBuffer.buffer.put(((float) newVertex[0])); mesh.skinnedVertexBuffer.buffer.put(((float) newVertex[1])); mesh.skinnedVertexBuffer.buffer.put(((float) newVertex[2])); } mesh.vertexBuffer = new FloatDataBuffer( mesh.numVertices, 3); mesh.skinnedVertexBuffer.buffer.rewind(); mesh.vertexBuffer.buffer.put(mesh.skinnedVertexBuffer.buffer); } } for (Bone childBone : bone.boneChilds) { transformVertex(childBone); } } I know this is not the more efficient code but by now I just want to understand exactly how a hierarchical model is organized and how I can draw it on the screen. Thanks in advance for your help.

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