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• Converting Milliseconds to Timecode

  - by Jeff

  I have an audio project I'm working on using BASS from Un4seen. This library uses BYTES mainly but I have a conversion in place that let's me show the current position of the song in Milliseconds. Knowing that MS = Samples * 1000 / SampleRate and that Samples = Bytes * 8 / Bits / Channels So here's my main issue and it's fairly simple... I have a function in my project that converts the Milliseconds to TimeCode in Mins:Secs:Milliseconds. Public Function ConvertMStoTimeCode(ByVal lngCurrentMSTimeValue As Long) ConvertMStoTimeCode = CheckForLeadingZero(Fix(lngCurrentMSTimeValue / 1000 / 60)) & ":" & _ CheckForLeadingZero(Int((lngCurrentMSTimeValue / 1000) Mod 60)) & ":" & _ CheckForLeadingZero(Int((lngCurrentMSTimeValue / 10) Mod 100)) End Function Now the issue comes within the Seconds calculation. Anytime the MS calculation is over .5 the seconds place rounds up to the next second. So 1.5 seconds actually prints as 2.5 seconds. I know for sure that using the Int conversion causes a round down and I know my math is correct as I've checked in a calculator 100 times. I can't figure out why the number is rounding up. Any suggestions?

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• Java NIO (Netty): How does Encryption or GZIPping work in theory (with filters)

  - by Tom

  Hello Experts, i would be very thankfull if you can explain to me, how in theory the "Interceptor/Filter" Pattern in ByteStreams (over Sockets/Channels) work (in Asynchronous IO with netty) in regard to encryption or compression of data. Given I have a Filter that does GZIPPING. How is this internally implemented? Does the Filter "collect" so many bytes form the channel, that this is a usefull number of bytes that can then be en/decoded? What is in general the minimal "blocksize(data to encode/decode in a chunk)" of socket based gzipping? Does this "blocksize" have to be negotiated in advance between server and client? What happens if the client does not send enough data to "fill" the blocksize (due to a network conquestion) but does not close the connection. Does this mean the other side will simply wait until it gets enough bytes to decode or until a timeout occoures...How is the Filter pattern the applied? The compression filter will de/compress the blocksize of bytes and then store them again in the same buffer would (in the case of netty) i normally be using the ChannelHanlderContext to pass the de/encoded data to the next filter?... Any explanations/links/tutorials (for beginners;-) will be very much appreciated to help me understand how for example encryption/compressing are implemented in socket based communication with filters/interceptor pattern. thank you very much tom

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• Using ServletOutputStream to write very large files in a Java servlet without memory issues

  - by Martin

  I am using IBM Websphere Application Server v6 and Java 1.4 and am trying to write large CSV files to the ServletOutputStream for a user to download. Files are ranging from a 50-750MB at the moment. The smaller files aren't causing too much of a problem but with the larger files it appears that it is being written into the heap which is then causing an OutOfMemory error and bringing down the entire server. These files can only be served out to authenticated users over https which is why I am serving them through a Servlet instead of just sticking them in Apache. The code I am using is (some fluff removed around this): resp.setHeader("Content-length", "" + fileLength); resp.setContentType("application/vnd.ms-excel"); resp.setHeader("Content-Disposition","attachment; filename=\"export.csv\""); FileInputStream inputStream = null; try { inputStream = new FileInputStream(path); byte[] buffer = new byte[1024]; int bytesRead = 0; do { bytesRead = inputStream.read(buffer, offset, buffer.length); resp.getOutputStream().write(buffer, 0, bytesRead); } while (bytesRead == buffer.length); resp.getOutputStream().flush(); } finally { if(inputStream != null) inputStream.close(); } The FileInputStream doesn't seem to be causing a problem as if I write to another file or just remove the write completly the memory usage doesn't appear to be a problem. What I am thinking is that the resp.getOutputStream().write is being stored in memory until the data can be sent through to the client. So the entire file might be read and stored in the resp.getOutputStream() causing my memory issues and crashing! I have tried Buffering these streams and also tried using Channels from java.nio, none of which seems to make any bit of difference to my memory issues. I have also flushed the outputstream once per iteration of the loop and after the loop, which didn't help.

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• detecting pauses in a spoken word audio file using pymad, pcm, vad, etc

  - by james

  First I am going to broadly state what I'm trying to do and ask for advice. Then I will explain my current approach and ask for answers to my current problems. Problem I have an MP3 file of a person speaking. I'd like to split it up into segments roughly corresponding to a sentence or phrase. (I'd do it manually, but we are talking hours of data.) If you have advice on how to do this programatically or for some existing utilities, I'd love to hear it. (I'm aware of voice activity detection and I've looked into it a bit, but I didn't see any freely available utilities.) Current Approach I thought the simplest thing would be to scan the MP3 at certain intervals and identify places where the average volume was below some threshold. Then I would use some existing utility to cut up the mp3 at those locations. I've been playing around with pymad and I believe that I've successfully extracted the PCM (pulse code modulation) data for each frame of the mp3. Now I am stuck because I can't really seem to wrap my head around how the PCM data translates to relative volume. I'm also aware of other complicating factors like multiple channels, big endian vs little, etc. Advice on how to map a group of pcm samples to relative volume would be key. Thanks!

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• WCF Security in a Windows Service

  - by Alphonso

  I have a WCF service which can run as Console App and a Windows Service. I have recently copied the console app up to a W2K3 server with the following security settings: <wsHttpBinding> <binding name="ServiceBinding_Security" transactionFlow="true" > <security mode="TransportWithMessageCredential" > <message clientCredentialType="UserName" /> </security> </binding> </wsHttpBinding> <serviceCredentials> <userNameAuthentication userNamePasswordValidationMode="Custom" customUserNamePasswordValidatorType="Common.CustomUserNameValidator, Common" /> </serviceCredentials> Security works fine with no problems. I have exactly the same code, but running in a windows service and I get the following error when I try to call any of the methods from a client: System.ServiceModel.Security.MessageSecurityException was unhandled Message="An unsecured or incorrectly secured fault was received from the other party. See the inner FaultException for the fault code and detail." Source="mscorlib" StackTrace: Server stack trace: at System.ServiceModel.Channels.SecurityChannelFactory`1.SecurityRequestChannel.ProcessReply(Message reply, SecurityProtocolCorrelationState correlationState, TimeSpan timeout) ...... (lots of stacktrace info - not very useful) InnerException: System.ServiceModel.FaultException Message="An error occurred when verifying security for the message." The exception tells me nothing. I'm assuming that it has something to do with acces to system resources from the Windows Service. I've tried running it under the same account as the console app, but no luck. Does anyone have any ideas?

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• Client unable to authenticate when connecting to WCF service

  - by davecoulter

  I have a WCF service hosted in a Windows service. The application is an intranet app, and I have programmatically set the bindings on both the service and the client as: NetTcpBinding aBinding = new NetTcpBinding(SecurityMode.Transport); aBinding.Security.Transport.ClientCredentialType = TcpClientCredentialType.Windows; aBinding.Security.Transport.ProtectionLevel = System.Net.Security.ProtectionLevel.EncryptAndSign; Both the service and client have endpoints configured with SPNs: EndpointAddress = new EndpointAddress(uri, EndpointIdentity.CreateSpnIdentity("Service1")); As far as I know, I have setup the bindings correctly-- and I am usually able to connect to the service just fine. I did however run into a case where on a server running Windows Server 2003 R2, x64, SP2 I get the following exception immediately when the client tries to connect: INNEREXCEPTION -- Exception Message: InvalidCredentialException: Either the target name is incorrect or the server has rejected the client credentials. Stack Trace: at System.Net.Security.NegoState.ProcessAuthentication(LazyAsyncResult lazyResult) at System.Net.Security.NegotiateStream.AuthenticateAsClient(NetworkCredential credential, String targetName, ProtectionLevel requiredProtectionLevel, TokenImpersonationLevel allowedImpersonationLevel) at System.ServiceModel.Channels.WindowsStreamSecurityUpgradeProvider.WindowsStreamSecurityUpgradeInitiator.OnInitiateUpgrade(Stream stream, SecurityMessageProperty& remoteSecurity) I get the exception when I try to connect to the service from another machine in the domain, but if I connect to the service on the same machine running the service it works fine. The hosting service itself is running as a domain user account-- but I have tried running the service as a Local System and Network Service to no avail. I have checked the Local Security Policies for the server and didn't see anything amiss (i.e. 'Access this computer from the network' includes 'Everyone'). Anyone have an idea of what could resolve this? I am wondering if I need to do something in Active Directory with respect to the service's SPN? I have read some about using setspn.exe to register or refresh SPNs, but I haven't needed to do this before. Why would this be working with other configurations but not the one above?

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• Need help in resolving a compiler error: error: invalid conversion from ‘int’ to ‘GIOCondition’

  - by michael

  I have a simple cpp file which uses GIO. I have stripped out everything to show my compile error: Here is the error I get: My.cpp:16: error: invalid conversion from ‘int’ to ‘GIOCondition’ make[2]: *** [My.o] Error 1 Here is the complete file: #include <glib.h> static gboolean read_socket (GIOChannel *gio, GIOCondition condition, gpointer data) { return false; } void createGIOChannel() { GIOChannel* gioChannel = g_io_channel_unix_new(0); // the following is the line causing the error: g_io_add_watch(gioChannel, G_IO_IN|G_IO_HUP, read_socket, NULL); } I have seen other example using gio, and I am doing the same thing in term of calling G_IO_IN|G_IO_HUP. And the documentation http://www.gtk.org/api/2.6/glib/glib-IO-Channels.html, said I only need to include , which I did. Can you please tell me how to resolve my error? One thing I can think of is I am doing this in a cpp file. But g_io_add_watch is a c function? Thank you for any help. I have spent hours on this but did not get anywhere.

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• Encoding h.264 with libavcodec/x264

  - by Leviathan

  I am attempting to encode video using libavcodec/libavformat. I'm trying to change the standard output-example.c from ffmpeg source. The AVI file is created on the disk, but the only sound is encoded. I tried adding a lot of options for x264 from here. All the other codecs works fine, mpeg2, mpeg4, mjpeg, xvid. In addition to specifying the parameters x264, I also set the codec to AVOutputFormat structure. That's all I've done. AVOutputFormat *pOutFormat; // in header file av_register_all(); AVCodec *codec = avcodec_find_encoder_by_name("libx264"); pOutFormat = guess_format("avi", NULL, NULL); pOutFormat->video_codec = codec->id; The debug output of my application: Output #0, mp4, to 'D:\1.avi': Stream #0.0: Video: libx264, yuv420p, 320x240, q=10-51, 500 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: aac, 44100 Hz, 1 channels, s16, 128 kb/s [libx264 @ 0x694010]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x694010]bitrate tolerance too small, using .01 [libx264 @ 0x694010]profile Main, level 2.0 [libx264 @ 0x694010]frame I:150 Avg QP:14.76 size: 2534 [libx264 @ 0x694010]mb I I16..4: 75.9% 0.0% 24.1% [libx264 @ 0x694010]final ratefactor: 17.57 [libx264 @ 0x694010]coded y,uvDC,uvAC intra: 42.7% 92.4% 47.4% [libx264 @ 0x694010]i16 v,h,dc,p: 11% 14% 2% 73% [libx264 @ 0x694010]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 18% 29% 5% 8% 10% 3% 3% 2% [libx264 @ 0x694010]kb/s:506.79

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• How to solve with using Flex Builder 3 and BlazeDS?

  - by Teerasej

  Hi, everyone. Thank you for interesting in my question, I think you can help me out from this little problem. I am using Flex builder 3, BlazeDS, and Java with Spring and Hibernate framework. I using the remote object to load a string from spring's configuration files. But in testing, I found this fault event like this: RPC Fault faultString="java.lang.NullPointerException" faultCode="Server.Processing" faultDetail="null" I have check the configuration in remote-config.xml and services-config.xml. But it looks good. There is some people talked about this problem around the internet and I think you can help me and them. I am using these environment: Flex Builder 3 BlazeDS 3.2.0 JBoss server Full stacktrace: [RPC Fault faultString="java.lang.NullPointerException" faultCode="Server.Processing" faultDetail="null"] at mx.rpc::AbstractInvoker/http://www.adobe.com/2006/flex/mx/internal::faultHandler()[C:\autobuild\3.2.0\frameworks\projects\rpc\src\mx\rpc\AbstractInvoker.as:220] at mx.rpc::Responder/fault()[C:\autobuild\3.2.0\frameworks\projects\rpc\src\mx\rpc\Responder.as:53] at mx.rpc::AsyncRequest/fault()[C:\autobuild\3.2.0\frameworks\projects\rpc\src\mx\rpc\AsyncRequest.as:103] at NetConnectionMessageResponder/statusHandler()[C:\autobuild\3.2.0\frameworks\projects\rpc\src\mx\messaging\channels\NetConnectionChannel.as:569] at mx.messaging::MessageResponder/status()[C:\autobuild\3.2.0\frameworks\projects\rpc\src\mx\messaging\MessageResponder.as:222]

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• GLSL shader render to texture not saving alpha value

  - by quadelirus

  I am rendering to a texture using a GLSL shader and then sending that texture as input to a second shader. For the first texture I am using RGB channels to send color data to the second GLSL shader, but I want to use the alpha channel to send a floating point number that the second shader will use as part of its program. The problem is that when I read the texture in the second shader the alpha value is always 1.0. I tested this in the following way: at the end of the first shader I did this: gl_FragColor(r, g, b, 0.1); and then in the second texture I read the value of the first texture using something along the lines of vec4 f = texture2D(previous_tex, pos); if (f.a != 1.0) { gl_FragColor = vec4(0.0, 0.0, 0.0, 1.0); return; } No pixels in my output are black, whereas if I change the above code to read gl_FragColor(r, g, 0.1, 1.0); //Notice I'm now sending 0.1 for blue and in the second shader vec4 f = texture2D(previous_tex, pos); if (f.b != 1.0) { gl_FragColor = vec4(0.0, 0.0, 0.0, 1.0); return; } All the appropriate pixels are black. This means that for some reason when I set the alpha value to something other than 1.0 in the first shader and render to a texture, it is still seen as being 1.0 by the second shader. Before I render to texture I glDisable(GL_BLEND); It seems pretty clear to me that the problem has to do with OpenGL handling alpha values in some way that isn't obvious to me since I can use the blue channel in the way I want, and figured someone out there will instantly see the problem.

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• DotNetOpenAuth occasionally throws a NotImplementedException

  - by Chris Moschini

  I have DotNetOpenAuth running on a background thread making calls to Google authorized with OAuth on a regular basis. About once a day, which is about one in 10,000 calls, I get the following Exception: An unhandled exception occurred and the process was terminated. Application ID: DefaultDomain Process ID: 3316 Exception: System.NotImplementedException Message: The method or operation is not implemented. StackTrace: at DotNetOpenAuth.Messaging.ProtocolException.GetObjectData(SerializationInfo info, StreamingContext context) in c:\Users\andarno\git\dotnetopenid\src\DotNetOpenAuth\Messaging\ProtocolException.cs:line 90 at System.Runtime.Serialization.Formatters.Binary.WriteObjectInfo.InitSerialize(Object obj, ISurrogateSelector surrogateSelector, StreamingContext context, SerObjectInfoInit serObjectInfoInit, IFormatterConverter converter, ObjectWriter objectWriter, SerializationBinder binder) at System.Runtime.Serialization.Formatters.Binary.WriteObjectInfo.Serialize(Object obj, ISurrogateSelector surrogateSelector, StreamingContext context, SerObjectInfoInit serObjectInfoInit, IFormatterConverter converter, ObjectWriter objectWriter, SerializationBinder binder) at System.Runtime.Serialization.Formatters.Binary.ObjectWriter.Serialize(Object graph, Header[] inHeaders, __BinaryWriter serWriter, Boolean fCheck) at System.Runtime.Serialization.Formatters.Binary.BinaryFormatter.Serialize(Stream serializationStream, Object graph, Header[] headers, Boolean fCheck) at System.Runtime.Remoting.Channels.CrossAppDomainSerializer.SerializeObject(Object obj, MemoryStream stm) at System.AppDomain.Serialize(Object o) at System.AppDomain.MarshalObject(Object o) If it was thrown and caught once a day I'd be fine, but this is a big one - I'm getting this in the Application Error log on the server, and it's crashing the process entirely - the site goes down and restarts. Has anyone else run into this? Something I'm clearly doing wrong?

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• I'm making a simulated tv

  - by Jam

  I need to make a tv that shows the user the channel and the volume, and shows whether or not the television is on. I have the majority of the code made, but for some reason the channels won't switch. I'm fairly unfamiliar with how properties work, and I think that's what my problem here is. Help please. class Television(object): def __init__(self, __channel=1, volume=1, is_on=0): self.__channel=__channel self.volume=volume self.is_on=is_on def __str__(self): if self.is_on==1: print "The tv is on" print self.__channel print self.volume else: print "The television is off." def toggle_power(self): if self.is_on==1: self.is_on=0 return self.is_on if self.is_on==0: self.is_on=1 return self.is_on def get_channel(self): return channel def set_channel(self, choice): if self.is_on==1: if choice>=0 and choice<=499: channel=self.__channel else: print "Invalid channel!" else: print "The television isn't on!" channel=property(get_channel, set_channel) def raise_volume(self, up=1): if self.is_on==1: self.volume+=up if self.volume>=10: self.volume=10 print "Max volume!" else: print "The television isn't on!" def lower_volume(self, down=1): if self.is_on==1: self.volume-=down if self.volume<=0: self.volume=0 print "Muted!" else: print "The television isn't on!" def main(): tv=Television() choice=None while choice!="0": print \ """ Television 0 - Exit 1 - Toggle Power 2 - Change Channel 3 - Raise Volume 4 - Lower Volume """ choice=raw_input("Choice: ") print if choice=="0": print "Good-bye." elif choice=="1": tv.toggle_power() tv.__str__() elif choice=="2": change=raw_input("What would you like to change the channel to?") tv.set_channel(change) tv.__str__() elif choice=="3": tv.raise_volume() tv.__str__() elif choice=="4": tv.lower_volume() tv.__str__() else: print "\nSorry, but", choice, "isn't a valid choice." main() raw_input("Press enter to exit.")

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• How to read/write high-resolution (24-bit, 8 channel) .wav files in Java?

  - by dB'

  I'm trying to write a Java application that manipulates high resolution .wav files. I'm having trouble importing the audio data, i.e. converting the .wav file into an array of doubles. When I use a standard approach an exception is thrown. AudioFileFormat as = AudioSystem.getAudioFileFormat(new File("orig.wav")); --> javax.sound.sampled.UnsupportedAudioFileException: file is not a supported file type Here's the file format info according to soxi: dB$ soxi orig.wav soxi WARN wav: wave header missing FmtExt chunk Input File : 'orig.wav' Channels : 8 Sample Rate : 96000 Precision : 24-bit Duration : 00:00:03.16 = 303526 samples ~ 237.13 CDDA sectors File Size : 9.71M Bit Rate : 24.6M Sample Encoding: 32-bit Floating Point PCM Can anyone suggest the simplest method for getting this audio into Java? I've tried using a few techniques. As stated above, I've experimented with the Java AudioSystem (on both Mac and Windows). I've also tried using Andrew Greensted's WavFile class, but this also fails (WavFileException: Compression Code 3 not supported). One workaround is to convert the audio to 16 bits using sox (with the -b 16 flag), but this is suboptimal since it increases the noise floor. Incidentally, I've noticed that the file CAN be read by libsndfile. Is my best bet to write a jni wrapper around libsndfile, or can you suggest something quicker? Note that I don't need to play the audio, I just need to analyze it, manipulate it, and then write it out to a new .wav file. * UPDATE * I solved this problem by modifying Andrew Greensted's WavFile class. His original version only read files encoded as integer values ("format code 1"); my files were encoded as floats ("format code 3"), and that's what was causing the problem. I'll post the modified version of Greensted's code when I get a chance. In the meantime, if anyone wants it, send me a message.

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• Why does this gstreamer pipeline stall ?

  - by timday

  I've been playing around with gstreamer pipelines using gst-launch. I don't have any problems if I just want to process audio or video separately (to separate files, or to alsasink/ximagesink), but I'm confused by what I need to do to mux the streams back together using, say avimux. This gst-launch-0.10 filesrc location=MVI_2034.AVI ! decodebin name=dec \ dec. ! queue ! audioconvert ! 'audio/x-raw-int,rate=44100,channels=1' ! queue ! mux. \ dec. ! queue ! videoflip 1 ! ffmpegcolorspace ! jpegenc ! queue ! mux. \ avimux name=mux ! filesink location=out.avi just outputs Setting pipeline to PAUSED ... Pipeline is PREROLLING ... and then stalls indefinitely. What's the trick ?

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• Handling file uploads with JavaScript and Google Gears, is there a better solution?

  - by gnarf

  So - I've been using this method of file uploading for a bit, but it seems that Google Gears has poor support for the newer browsers that implement the HTML5 specs. I've heard the word deprecated floating around a few channels, so I'm looking for a replacement that can accomplish the following tasks, and support the new browsers. I can always fall back to gears / standard file POST's but these following items make my process much simpler: Users MUST to be able to select multiple files for uploading in the dialog. I MUST be able to receive status updates on the transmission of a file. (progress bars) I would like to be able to use PUT requests instead of POST I would like to be able to easily attach these events to existing HTML elements using JavaScript. I.E. the File Selection should be triggered on a <button> click. I would like to be able to control response/request parameters easily using JavaScript. I'm not sure if the new HTML5 browsers have support for the desktop/request objects gears uses, or if there is a flash uploader that has these features that I am missing in my google searches. An example of uploading code using gears: // select some files: var desktop = google.gears.factory.create('beta.desktop'); desktop.openFiles(selectFilesCallback); function selectFilesCallback(files) { $.each(files,function(k,file) { // this code actually goes through a queue, and creates some status bars // but it is unimportant to show here... sendFile(file); }); } function sendFile(file) { google.gears.factory.create('beta.httprequest'); request.open('PUT', upl.url); request.setRequestHeader('filename', file.name); request.upload.onprogress = function(e) { // gives me % status updates... allows e.loaded/e.total }; request.onreadystatechange = function() { if (request.readyState == 4) { // completed the upload! } }; request.send(file.blob); return request; } Edit: apparently flash isn't capable of using PUT requests, so I have changed it to a "like" instead of a "must".

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• Rapid calls to fread crashes the application

  - by Slynk

  I'm writing a function to load a wave file and, in the process, split the data into 2 separate buffers if it's stereo. The program gets to i = 18 and crashes during the left channel fread pass. (You can ignore the couts, they are just there for debugging.) Maybe I should load the file in one pass and use memmove to fill the buffers? if(params.channels == 2){ params.leftChannelData = new unsigned char[params.dataSize/2]; params.rightChannelData = new unsigned char[params.dataSize/2]; bool isLeft = true; int offset = 0; const int stride = sizeof(BYTE) * (params.bitsPerSample/8); for(int i = 0; i < params.dataSize; i += stride) { std::cout << "i = " << i << " "; if(isLeft){ std::cout << "Before Left Channel, "; fread(params.leftChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Left Channel, "; } else{ std::cout << "Before Right Channel, "; fread(params.rightChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Right Channel, "; offset += stride; std::cout << "After offset incr.\n"; } isLeft != isLeft; } } else { params.leftChannelData = new unsigned char[params.dataSize]; fread(params.leftChannelData, sizeof(BYTE), params.dataSize, file); }

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• How to play extracted wave file byte array in C#?

  - by user261924

  At the moment i have managed to separate the left and right channel of a WAVE file and have included the header in a byte[] array. My next step is to be about to play both channels. How can this be done? Here is a code snippet: byte[] song_left = new byte[fa.Length]; byte[] song_right = new byte[fa.Length]; int p = 0; for (int c = 0; c < 43; c++) { song_left[p] = header[c]; p++; } int q = 0; for (s = startByte; s < length; s = s + 3) { song_left[s] = sLeft[q]; q++; s++; song_left[s] = sLeft[q]; q++; } p = 0; for (int c = 0; c < 43; c++) { song_right[p] = header[c]; p++; } This part is reading the header and data from both the right and light channel and saving it to array sLeft[] and sRight[]. This part is working perfectly. Once I obtained the byte arrays, I did the following: System.IO.File.WriteAllBytes("c:\\left.wav", song_left); System.IO.File.WriteAllBytes("c:\\right.wav", song_right); Added a button to play the saved wave file: private void button2_Click(object sender, EventArgs e) { spWave = new SoundPlayer("c:\\left.wav"); spWave.Play(); } Once I hit the play button, this error appers: An unhandled exception of type 'System.InvalidOperationException' occurred in System.dll Additional information: The wave header is corrupt. Any ideas?

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• How to initialize audio with Vala/SDL

  - by ioev

  I've been trying to figure this out for a few hours now. In order to start up the audio, I need to create an SDL.AudioSpec object and pass it to SDL.Audio.Open. The problem is, AudioSpec is a class with a private constructor, so when I try to create one I get: sdl.vala:18.25-18.43: error: `SDL.AudioSpec' does not have a default constructor AudioSpec audiospec = new SDL.AudioSpec(); ^^^^^^^^^^^^^^^^^^^ And if I try to just assign values to it's member vars like a struct (it's a struct in normal sdl) I get: sdl.vala:20.3-20.25: error: use of possibly unassigned local variable `audiospec' audiospec.freq = 22050; ^^^^^^^^^^^^^^^^^^^^^^^ I found the valac doc here: http://valadoc.org/sdl/SDL.AudioSpec.html But it isn't much help at all. The offending code block looks like this: // setup the audio configuration AudioSpec audiospec; AudioSpec specback; audiospec.freq = 22050; audiospec.format = SDL.AudioFormat.S16LSB; audiospec.channels = 2; audiospec.samples = 512; // try to initialize sound with these values if (SDL.Audio.open(audiospec, specback) < 0) { stdout.printf("ERROR! Check audio settings!\n"); return 1; } Any help would be greatly appreciated!

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• PHP package manager

  - by Mathias

  Hey, does anyone know a package manager library for PHP (as e.g. apt or yum for linux distros) apart from PEAR? I'm working on a system which should include a package management system for module management. I managed to get a working solution using PEAR, but using the PEAR client for anything else than managing a PEAR installation is not really the optimal solution as it's not designed for that. I would have to modify/extend it (e.g. to implement actions on installation/upgrade or to move PEAR specific files like lockfiles away from the system root) and especially the CLI client code is quite messy and PHP4. So maybe someone has some suggestions for an alternative PEAR client library which is easy to use and extend (the server side has some nice implementations like Pirum and pearhub) for completely different package management systems written in PHP (ideally including dependency tracking and different channels) for some general ideas how to implement such a PM system (yes, I'm still tinkering with the idea of implementing such a system from scratch) I know that big systems like Magento and symfony use PEAR for their PM. Magento uses a hacked version of the original PEAR client (which I'd like to avoid), symfony's implementation seems quite integrated with the framework, but would be a good starting point to at least write the client from scratch. Anyway, if anybody has suggestions: please :)

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• Spring Integration 1.0 RC2: Streaming file content?

  - by gdm

  I've been trying to find information on this, but due to the immaturity of the Spring Integration framework I haven't had much luck. Here is my desired work flow: New files are placed in an 'Incoming' directory Files are picked up using a file:inbound-channel-adapter The file content is streamed, N lines at a time, to a 'Stage 1' channel, which parses the line into an intermediary (shared) representation. This parsed line is routed to multiple 'Stage 2' channels. Each 'Stage 2' channel does its own processing on the N available lines to convert them to a final representation. This channel must have a queue which ensures no Stage 2 channel is overwhelmed in the event that one channel processes significantly slower than the others. The final representation of the N lines is written to a file. There will be as many output files as there were routing destinations in step 4. *'N' above stands for any reasonable number of lines to read at a time, from [1, whatever I can fit into memory reasonably], but is guaranteed to always be less than the number of lines in the full file. How can I accomplish streaming (steps 3, 4, 5) in Spring Integration? It's fairly easy to do without streaming the files, but my files are large enough that I cannot read the entire file into memory. As a side note, I have a working implementation of this work flow without Spring Integration, but since we're using Spring Integration in other places in our project, I'd like to try it here to see how it performs and how the resulting code compares for length and clarity.

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• Java: Embedding Soundbank file in JAR

  - by Pyroclastic

  If I have a soundbank stored in a JAR, how would I load that soundbank into my application using resource loading...? I'm trying to consolidate as much of a MIDI program into the jar file as I can, and the last thing I have to add is the soundbank file I'm using, as users won't have the soundbanks installed. I'm trying to put it into my jar file, and then load it with getResource() in the Class class, but I'm getting an InvalidMidiDataException on a soundbank that I know is valid. Here's the code, it's in the constructor for my synthesizer object: try { synth = MidiSystem.getSynthesizer(); channels = synth.getChannels(); instrument = MidiSystem.getSoundbank(this.getClass().getResource("img/soundbank-mid.gm")).getInstruments(); currentInstrument = instrument[0]; synth.loadInstrument(currentInstrument); synth.open(); } catch (InvalidMidiDataException ex) { System.out.println("FAIL"); instrument = synth.getAvailableInstruments(); currentInstrument = instrument[0]; synth.loadInstrument(currentInstrument); try { synth.open(); } catch (MidiUnavailableException ex1) { Logger.getLogger(MIDISynth.class.getName()).log(Level.SEVERE, null, ex1); } } catch (IOException ex) { Logger.getLogger(MIDISynth.class.getName()).log(Level.SEVERE, null, ex); } catch (MidiUnavailableException ex) { Logger.getLogger(MIDISynth.class.getName()).log(Level.SEVERE, null, ex); }

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• Castle windsor security exception

  - by Sunil

  I developed a small WCF service that uses Castle Windsor IoC container and it works fine on my PC. When I deploy it onto a Win 2008 R2 server and host the WCF service in IIS 7 it fails with the following error. I checked the server level web.config and the trust level is set to "Full". What do I need to do to get this to work. As a test I deployed the same service as it is onto a Windows 2003 server with the trust level set to "Full" and it works fine. I am unable to figure out what setting/configuration I am missing on the 2008 server that is making the service fail. Stack Trace: [SecurityException: That assembly does not allow partially trusted callers.] Castle.Windsor.WindsorContainer..ctor() +0 WMS.ServiceContractImplementation.IoC.IoCInstanceProvider..ctor(Type serviceType) in D:\WCF\WCFProofOfConcept\WMSServices \WMS.ServiceContractImplementation\IoC\IoCInstanceProvider.cs:19 WMS.ServiceContractImplementation.IoC.IoCServiceBehavior.ApplyDispatchBehav­ior(ServiceDescription serviceDescription, ServiceHostBase serviceHostBase) in D:\WCF \WCFProofOfConcept\WMSServices\WMS.ServiceContractImplementation\IoC \IoCServiceBehavior.cs:24 System.ServiceModel.Description.DispatcherBuilder.InitializeServiceHost(Ser­viceDescription description, ServiceHostBase serviceHost) +377 System.ServiceModel.ServiceHostBase.InitializeRuntime() +37 System.ServiceModel.ServiceHostBase.OnBeginOpen() +27 System.ServiceModel.ServiceHostBase.OnOpen(TimeSpan timeout) +49 System.ServiceModel.Channels.CommunicationObject.Open(TimeSpan timeout) +261 System.ServiceModel.HostingManager.ActivateService(String normalizedVirtualPath) +121 System.ServiceModel.HostingManager.EnsureServiceAvailable(String normalizedVirtualPath) +479

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• Design pattern to integrate Rails with a Comet server

  - by empire29

  I have a Ruby on Rails (2.3.5) application and an APE (Ajax Push Engine) server. When records are created within the Rails application, i need to push the new record out on applicable channels to the APE server. Records can be created in the rails app by the traditional path through the controller's create action, or it can be created by several event machines that are constantly monitoring various inputstream and creating records when they see data that meets a certain criteria. It seems to me that the best/right place to put the code that pushes the data out to the APE server (which in turn pushes it out to the clients) is in the Model's after_create hook (since not all record creations will flow through the controller's create action). The final caveat is I want to push a piece of formatted HTML out to the APE server (rather than a JSON representation of the data). The reason I want to do this is 1) I already have logic to produce the desired layout in existing partials 2) I don't want to create a javascript implementation of the partials (javascript that takes a JSON object and creates all the HTML around it for presentation). This would quickly become a maintenance nightmare. The problem with this is it would require "rendering" partials from within the Model (which im having trouble doing anyhow because they don't seem to have access to Helpers when they're rendered in this manner). Anyhow - Just wondering what the right way to go about organizing all of this is. Thanks

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• Real Time Sound Captureing J2ME

  - by Abdul jalil

  i am capturing sound in J2me and send these bytes to remote system, i then play these bytes on remote system.five second voice is capture and send to remote system. i get the repeated sound again .i am making a sound messenger please help me where i am doing wrong i am using the follown code . String remoteTimeServerAddress="192.168.137.179"; sc = (SocketConnection) Connector.open("socket://"+remoteTimeServerAddress+":13"); p = Manager.createPlayer("capture://audio?encoding=pcm&rate=11025&bits=16&channels=1"); p.realize(); RecordControl rc = (RecordControl)p.getControl("RecordControl"); ByteArrayOutputStream output = new ByteArrayOutputStream(); OutputStream outstream =sc.openOutputStream(); rc.setRecordStream(output); rc.startRecord(); p.start(); int size=output.size(); int offset=0; while(true) { Thread.currentThread().sleep(5000); rc.commit(); output.flush(); size=output.size(); if(size0) { recordedSoundArray=output.toByteArray(); outstream.write(recordedSoundArray,0,size); } output.reset(); rc.reset(); rc.setRecordStream(output); rc.startRecord(); }

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• How to hold a queue of messages and have a group of working threads without polling?

  - by Mark

  I have a workflow that I want to looks something like this: / Worker 1 \ =Request Channel= - [Holding Queue|||] - Worker 2 - =Response Channel= \ Worker 3 / That is: Requests come in and they enter a FIFO queue Identical workers then pick up tasks from the queue At any given time any worker may work only one task When a worker is free and the holding queue is non-empty the worker should immediately pick up another task When tasks are complete, a worker places the result on the Response Channel I know there are QueueChannels in Spring Integration, but these channels require polling (which seems suboptimal). In particular, if a worker can be busy, I'd like the worker to be busy. Also, I've considered avoiding the queue altogether and simply letting tasks round-robin to all workers, but it's preferable to have a single waiting line as some tasks may be accomplished faster than others. Furthermore, I'd like insight into how many jobs are remaining (which I can get from the queue) and the ability to cancel all or particular jobs. How can I implement this message queuing/work distribution pattern while avoiding a polling? Edit: It appears I'm looking for the Message Dispatcher pattern -- how can I implement this using Spring/Spring Integration?

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