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• SSH Kerberos Auth in Mac OSX 10.7

  - by deemstone

  I just upgrade my Mac OS to 10.7 Lion. It has worked well before. But , Only kinit working normally now, can't ssh to my server. After reinstall the "Mac OS X Kerberos Extras" , nothing better. Anyone give me a help? Thanks a lot!! my command line : Myname$ ssh [email protected] -v ...... debug1: Authentications that can continue: gssapi-with-mic,password debug1: Next authentication method: gssapi-with-mic debug1: Miscellaneous failure (see text) UNKNOWN_SERVER while looking up 'host/[email protected]' (cached result, timeout in 1200 sec) debug1: An invalid name was supplied unknown mech-code 0 for mech 1 2 752 43 14 2 debug1: Miscellaneous failure (see text) unknown mech-code 0 for mech 1 3 6 1 5 5 14 debug1: Authentications that can continue: gssapi-with-mic,password debug1: An unsupported mechanism was requested unknown mech-code 0 for mech 1 3 5 1 5 2 7 debug1: Miscellaneous failure (see text) unknown mech-code 0 for mech 1 3 6 1 5 2 5 debug1: Next authentication method: password [email protected]'s password:

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• Audio input problem in Ubuntu 9.10

  - by Andrea Ambu

  My audio input is a mix of my mic output and my sound card output. I'd like it to be just my mic output. I was able to do so in Ubuntu 9.04 but the interface is 9.10 is totally changed and I tried every my creativity was able to think. It's really annoying when talking to other people over the internet because they keep hearing their voice back. I'm not sure I explained it in clear way so I'll give you an example: What I do: I put an mp3 on play or a video on youtube then open a recorder and start to talk on my mic. What happens: both my voice and audio from mp3/youtube get reordered, even if I put headphones volume to 0 (via hardware). What I'd like to happen: Only my voice should be recorded. I'm sure I'm missing some technical term, but that's the problem and I'd like to solve it in Ubuntu 9.10, any idea?

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• Audio input problem in Ubuntu 9.10

  - by Andrea Ambu

  My audio input is a mix of my mic output and my sound card output. I'd like it to be just my mic output. I was able to do so in Ubuntu 9.04 but the interface is 9.10 is totally changed and I tried every my creativity was able to think. It's really annoying when talking to other people over the internet because they keep hearing their voice back. I'm not sure I explained it in clear way so I'll give you an example: What I do: I put an mp3 on play or a video on youtube then open a recorder and start to talk on my mic. What happens: both my voice and audio from mp3/youtube get reordered, even if I put headphones volume to 0 (via hardware). What I'd like to happen: Only my voice should be recorded. I'm sure I'm missing some technical term, but that's the problem and I'd like to solve it in Ubuntu 9.10, any idea?

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• Is there a way to set up an SMTP relay that allows users of a web app to have the web app send email

  - by mic

  the web service sends out emails on behalf of the users to their customers. So [email protected] uses webservice and webservice sends emails . The emails should be appearing as coming from [email protected]. Currently what we are trying to do is to configure webservice to act as an email client for each user, each user being able to create their own profile in which they need to configure their smtp server credentials. But given that there are more options for configurations than you can shake your stick at -not to mention trying to explain to users what info to get from where, POP b4 smtp, TLS, SSL, AUTH,etc) I am wondering if there could be a different way. How, if at all could this be approached? Can I set up a postfix server to do what I need to without running into another admin. nightmare or being blocked for spamming? Thank you for your insights

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• How to send audio stream via UDP in java?

  - by Nob Venoda

  Hi to all :) I have a problem, i have set MediaLocator to microphone input, and then created Player. I need to grab that sound from the microphone, encode it to some lower quality stream, and send it as a datagram packet via UDP. Here's the code, i found most of it online and adapted it to my app: public class AudioSender extends Thread { private MediaLocator ml = new MediaLocator("javasound://44100"); private DatagramSocket socket; private boolean transmitting; private Player player; TargetDataLine mic; byte[] buffer; private AudioFormat format; private DatagramSocket datagramSocket(){ try { return new DatagramSocket(); } catch (SocketException ex) { return null; } } private void startMic() { try { format = new AudioFormat(AudioFormat.Encoding.PCM_SIGNED, 8000.0F, 16, 2, 4, 8000.0F, true); DataLine.Info info = new DataLine.Info(TargetDataLine.class, format); mic = (TargetDataLine) AudioSystem.getLine(info); mic.open(format); mic.start(); buffer = new byte[1024]; } catch (LineUnavailableException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } private Player createPlayer() { try { return Manager.createRealizedPlayer(ml); } catch (IOException ex) { return null; } catch (NoPlayerException ex) { return null; } catch (CannotRealizeException ex) { return null; } } private void send() { try { mic.read(buffer, 0, 1024); DatagramPacket packet = new DatagramPacket( buffer, buffer.length, InetAddress.getByName(Util.getRemoteIP()), 91); socket.send(packet); } catch (IOException ex) { Logger.getLogger(AudioSender.class.getName()).log(Level.SEVERE, null, ex); } } @Override public void run() { player = createPlayer(); player.start(); socket = datagramSocket(); transmitting = true; startMic(); while (transmitting) { send(); } } public static void main(String[] args) { AudioSender as = new AudioSender(); as.start(); } } And only thing that happens when I run the receiver class, is me hearing this Player from the sender class. And I cant seem to see the connection between TargetDataLine and Player. Basically, I need to get the sound form player, and somehow convert it to bytes[], therefore I can sent it as datagram. Any ideas? Everything is acceptable, as long as it works :)

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• Does the XML specification states that parser need to convert \n\r to \n always, even when \n\r appe

  - by mic.sca

  Hi, I've stumbled in a problem handling the \line-feed and \carriage-return characters in xml. I know that, according to http://www.w3.org/TR/REC-xml/#sec-line-ends, xml processors are required to replace any "\n\r" or lone "\r" sequences with "\n". The specification states that this has to be the behaviour for handling any "external parsed entity", does this apply to CDATA sections inside of an element as well? thank you, Michele I'm sure that msxml library for example converts every \n\r" or lone "\r" sequences to "\n", regardless of their being in a cdata section or not.

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• Windows Phone 7 Prototype 001: Speech Recognition on WP7

  At some point in the future it will be awesome when you can just tell your computer what to do and it does it - without typing to help those of us with a blistering 11 WPM hunk and peck technique. Siri, a mobile digital assistant using speech recognition was voted best tech at SXSW. I dont know about that one. Although, I'm sure it will get better when Apple rebuilds it and  bundles on iPhone 5. So how would you do that on WP7? There have been some videos floating around showing Bing with some voice control so obviously the phone has speech recognition. So what options are there: System.Speech? Not included in WP7/SL Nuance software like Siri? No WP7/SL version yet. Invoking the SAPI dlls on the phone? No automation factory in WP7 SL. Web services using System.Speech and mic on the phone? YES! The last one was my least favorite but that works for now. I built a quick sample app to show how to do text-to-speech and speech recognition on WP7.   @eklimczak will not be happy with the developer designed UI. In this sample there is web service with provides access to the system.speech APIs in .NET. Basically its just passing around byte arrays. On the phone its using the XNA audio frameworks to play the text-to-speech stream and to record using the microphone. The code is pretty simple and you can download from the link at the end of this post. The only things to note are adjusting the WCF config to handle larger byte uploads and the Microphone API is a little weird with that 1 second buffer. It would be nice if you could just to mic.start and mic.end which would return an array of bytes instead of managing your own stream inside the buffer ready callback. Couple of downsides to this approach: Recoding from the phone has some static. Could be my code or the my mic is bad / not calibrated right. Having to make web service calls instead of local access is not ideal (Microsoft, please add an API for the SAPI dlls) Although in the context of an app like Siri its not so bad since you need to do web service lookups to get data back Speech recognition quality really depends on either a) a limited grammar set like that pizza grammar in the sample or b) training the recognizer. For the latter it would be annoying to have users train the system. Using the System.Speech stuff youd have to have a profile for each user. So until Microsoft adds some speech client APIs on the phone or Nuance releases a wp7 product, this is a decent workaround. In the future Id like to build something similar to Siri. I shall call it Iris in homage. Im a big fan of mobile speech apps because frankly its just not safe to Google while driving. Since some of my designer co-workers have been posting UI sketches for WP7, Id like to start posting some code prototypes for things I try out on the phone. That will probably last 2 weeks, but for the moment I have like 10 posts in the queue. Sample Code 100% guaranteed to work on my emulatorDid you know that DotNetSlackers also publishes .net articles written by top known .net Authors? We already have over 80 articles in several categories including Silverlight. Take a look: here.

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• Can Sql Server 2005 Pivot table have nText passed into it?

  - by manemawanna

  Right bit of a simple question can I input nText into a pivot table? (SQL Server 2005) What I have is a table which records the answers to a questionnaire consisting of the following elements for example: UserID QuestionNumber Answer Mic 1 Yes Mic 2 No Mic 3 Yes Ste 1 Yes Ste 2 No Ste 3 Yes Bob 1 Yes Bob 2 No Bob 3 Yes With the answers being held in nText. Anyway what id like a Pivot table to do is: UserID 1 2 3 Mic Yes No Yes Ste Yes No Yes Bob Yes No Yes I have some test code, that creates a pivot table but at the moment it just shows the number of answers in each column (code can be found below). So I just want to know is it possible to add nText to a pivot table? As when I've tried it brings up errors and someone stated on another site that it wasn't possible, so I would like to check if this is the case or not. Just for further reference I don't have the opportunity to change the database as it's linked to other systems that I haven't created or have access too. Heres the SQL code I have at present below: DECLARE @query NVARCHAR(4000) DECLARE @count INT DECLARE @concatcolumns NVARCHAR(4000) SET @count = 1 SET @concatcolumns = '' WHILE (@count <=52) BEGIN IF @COUNT > 1 AND @COUNT <=52 SET @concatcolumns = (@concatcolumns + ' + ') SET @concatcolumns = (@concatcolumns + 'CAST ([' + CAST(@count AS NVARCHAR) + '] AS NVARCHAR)') SET @count = (@count+1) END DECLARE @columns NVARCHAR(4000) SET @count = 1 SET @columns = '' WHILE (@count <=52) BEGIN IF @COUNT > 1 AND @COUNT <=52 SET @columns = (@columns + ',') SET @columns = (@columns + '[' + CAST(@count AS NVARCHAR) + '] ') SET @count = (@count+1) END SET @query = ' SELECT UserID, ' + @concatcolumns + ' FROM( SELECT UserID, QuestionNumber AS qNum from QuestionnaireAnswers where QuestionnaireID = 7 ) AS t PIVOT ( COUNT (qNum) FOR qNum IN (' + @columns + ') ) AS PivotTable' select @query exec(@query)

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• Static background noise while using new headset Ubuntu 13.04

  - by ThundLayr

  Today I bought a new gaming headset (Gx-Gaming Lychas), and when I tried to record some gameplay-comentary I noticed that there always is a static background noise, I just recorded an example so you guys can listen it (no downloaded needed): http://www47.zippyshare.com/v/65167832/file.html I'm using Kubuntu 13.04 and Kernel version is 3.8.0-19, my laptop is an Acer Travelmate 5760Z, I tried tons of configurations on Alsamixer and none of them made result, I really need to get this working so any kind of help will be very aprecciated. cat /proc/asound/cards: 0 [PCH ]: HDA-Intel - HDA Intel PCH HDA Intel PCH at 0xc6400000 irq 44 cat /proc/asound/card0/codec#0 Codec: Conexant CX20588 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x14f1506c Subsystem Id: 0x10250574 Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A State of AFG node 0x01: Power states: D0 D1 D2 D3 D3cold CLKSTOP EPSS Power: setting=D0, actual=D0 GPIO: io=4, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x4a 0x4a] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x12 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x13 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=1, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0f, mute=0 Amp-Out vals: [0x00] Node 0x14 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Control: name="Capture Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Control: name="Capture Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Device: name="CX20588 Analog", type="Audio", device=0 Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x50 0x50] [0x80 0x80] [0x80 0x80] [0x80 0x80] Converter: stream=4, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x15 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x16 [Audio Input] wcaps 0x100d1b: Stereo Amp-In R/L Amp-In caps: ofs=0x4a, nsteps=0x50, stepsize=0x03, mute=1 Amp-In vals: [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] [0x4a 0x4a] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x17* 0x18 0x23 0x24 Node 0x17 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Control: name="Mic Boost Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a 0x1b* 0x1d 0x1e Node 0x18 [Audio Selector] wcaps 0x30050d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 4 0x1a* 0x1b 0x1d 0x1e Node 0x19 [Pin Complex] wcaps 0x400581: Stereo Control: name="Headphone Jack", index=0, device=0 Pincap 0x0000001c: OUT HP Detect Pin Default 0x04214040: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=01, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1a [Pin Complex] wcaps 0x400481: Stereo Control: name="Internal Mic Phantom Jack", index=0, device=0 Pincap 0x00001324: IN Detect Vref caps: HIZ 50 80 Pin Default 0x90a70130: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x3, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1b [Pin Complex] wcaps 0x400581: Stereo Control: name="Mic Jack", index=0, device=0 Pincap 0x00011334: IN OUT EAPD Detect Vref caps: HIZ 50 80 EAPD 0x0: Pin Default 0x04a19020: [Jack] Mic at Ext Right Conn = 1/8, Color = Pink DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=02, enabled=1 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1c [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1d [Pin Complex] wcaps 0x400581: Stereo Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10* 0x11 Node 0x1e [Pin Complex] wcaps 0x400481: Stereo Pincap 0x00000024: IN Detect Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x1f [Pin Complex] wcaps 0x400501: Stereo Control: name="Speaker Phantom Jack", index=0, device=0 Pincap 0x00000010: OUT Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11* Node 0x20 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x21 [Audio Output] wcaps 0x611: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 IEC Coding Type: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x22 [Pin Complex] wcaps 0x400781: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 1 0x21 Node 0x23 [Pin Complex] wcaps 0x40040b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x04, stepsize=0x2f, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f001f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x24 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Connection: 2 0x10 0x11 Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono

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• Class Decorators, Inheritence, super(), and maximum recursion

  - by jamstooks

  I'm trying to figure out how to use decorators on subclasses that use super(). Since my class decorator creates another subclass a decorated class seems to prevent the use of super() when it changes the className passed to super(className, self). Below is an example: def class_decorator(cls): class _DecoratedClass(cls): def __init__(self): return super(_DecoratedClass, self).__init__() return _DecoratedClass class BaseClass(object): def __init__(self): print "class: %s" % self.__class__.__name__ def print_class(self): print "class: %s" % self.__class__.__name__ bc = BaseClass().print_class() class SubClass(BaseClass): def print_class(self): super(SubClass, self).print_class() sc = SubClass().print_class() @class_decorator class SubClassAgain(BaseClass): def print_class(self): super(SubClassAgain, self).print_class() sca = SubClassAgain() # sca.print_class() # Uncomment for maximum recursion The output should be: class: BaseClass class: BaseClass class: SubClass class: SubClass class: _DecoratedClass Traceback (most recent call last): File "class_decorator_super.py", line 34, in <module> sca.print_class() File "class_decorator_super.py", line 31, in print_class super(SubClassAgain, self).print_class() ... ... RuntimeError: maximum recursion depth exceeded while calling a Python object Does anyone know of a way to not break a subclass that uses super() when using a decorator? Ideally I'd like to reuse a class from time to time and simply decorate it w/out breaking it.

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• SSH closing by itself - root works fine

  - by Antti

  I'm trying to connect to a server but if i use any other user than root the connection closes itself after a successful login: XXXXXXX:~ user$ ssh -v [email protected] OpenSSH_5.2p1, OpenSSL 0.9.8l 5 Nov 2009 debug1: Reading configuration data /etc/ssh_config debug1: Connecting to XXXXXXX.XXXXXX.XXX [xxx.xxx.xxx.xxx] port 22. debug1: Connection established. debug1: identity file /Users/user/.ssh/identity type -1 debug1: identity file /Users/user/.ssh/id_rsa type -1 debug1: identity file /Users/user/.ssh/id_dsa type 2 debug1: Remote protocol version 2.0, remote software version OpenSSH_4.3 debug1: match: OpenSSH_4.3 pat OpenSSH_4* debug1: Enabling compatibility mode for protocol 2.0 debug1: Local version string SSH-2.0-OpenSSH_5.2 debug1: SSH2_MSG_KEXINIT sent debug1: SSH2_MSG_KEXINIT received debug1: kex: server->client aes128-ctr hmac-md5 none debug1: kex: client->server aes128-ctr hmac-md5 none debug1: SSH2_MSG_KEX_DH_GEX_REQUEST(1024<1024<8192) sent debug1: expecting SSH2_MSG_KEX_DH_GEX_GROUP debug1: SSH2_MSG_KEX_DH_GEX_INIT sent debug1: expecting SSH2_MSG_KEX_DH_GEX_REPLY debug1: Host 'XXXXXXX.XXXXXX.XXX' is known and matches the RSA host key. debug1: Found key in /Users/user/.ssh/known_hosts:12 debug1: ssh_rsa_verify: signature correct debug1: SSH2_MSG_NEWKEYS sent debug1: expecting SSH2_MSG_NEWKEYS debug1: SSH2_MSG_NEWKEYS received debug1: SSH2_MSG_SERVICE_REQUEST sent debug1: SSH2_MSG_SERVICE_ACCEPT received debug1: Authentications that can continue: publickey,gssapi-with-mic,password debug1: Next authentication method: publickey debug1: Offering public key: /Users/user/.ssh/woo_openssh debug1: Authentications that can continue: publickey,gssapi-with-mic,password debug1: Offering public key: /Users/user/.ssh/sidlee.dsa debug1: Authentications that can continue: publickey,gssapi-with-mic,password debug1: Trying private key: /Users/user/.ssh/identity debug1: Trying private key: /Users/user/.ssh/id_rsa debug1: Offering public key: /Users/user/.ssh/id_dsa debug1: Authentications that can continue: publickey,gssapi-with-mic,password debug1: Next authentication method: password [email protected]'s password: debug1: Authentication succeeded (password). debug1: channel 0: new [client-session] debug1: Entering interactive session. Last login: Mon Mar 29 01:41:51 2010 from 193.67.179.2 debug1: client_input_channel_req: channel 0 rtype exit-status reply 0 debug1: channel 0: free: client-session, nchannels 1 Connection to XXXXXXX.XXXXXX.XXX closed. Transferred: sent 2976, received 2136 bytes, in 0.5 seconds Bytes per second: sent 5892.2, received 4229.1 debug1: Exit status 1 If i log in as root the exact same way it works as expected. I've added the users i want to log in with to a group (sshusers) and added that group to /etc/sshd_config: AllowGroups sshusers I'm not sure what to try next as i don't get a clear error anywhere. I would like to enable specific accounts to log in so that i can disable root. This is a GridServer/Media Temple (CentOS).

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• n68 s-ucc front panel audio

  - by user264522

  I have a motherboard asrock n68 s-ucc, and i need a hand identifying which pins on the board to connect my front audio header connectors to. On the connectors I have: double connector: GND and MIC IN double connector: LINE OUT FL and LINE OUT RL double connector: LINE OUT FR and LINE OUT RR simple connector: MIC POWER On the motherboard I have: GND PRESENCE# MIC_RET OUT_RET MIC2_L MIC2_R OUT2_R J_SENSE OUT2_L manual of motherboard and my front panel Thanks so much to all

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• How to change the "record external input" in Fraps?

  - by CyanPrime

  I'm using a TV to USB capture thingy called Hauppauge USB-Live2 (http://www.hauppauge.com/site/products/data_usblive2.html) and I'm using Media Player Classic Home Cinema to view my device's output, and finally I'm using Fraps to record the output off of MPC. It all works nicely except I can't get my mic to record, or even show up in the "Record External Input" area of Fraps while my Hauppauge is plugged in. My question is: How can I select my mic as the "record external input" in fraps?

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• What language/API to use for a standalone live-input audio visualizer app?

  - by knuckfubuck

  I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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• Flash and ActionScript

  - by Sonesh Dabhi

  I am not a flash/actionscript developer and I need to achieve a very small task in flash . I need to display user audio input level in flash . I found that I can do that using action script as below . I also checked this link . I have no idea what tools I need to use and generate a swf file . Any help highly appreciated . this.mic = Microphone.getMicrophone(); this.micTimer.addEventListener(TimerEvent.TIMER,this.timerHandler); this.micTimer.start(); this.mic.setLoopBack(true); return; public function timerHandler(event:TimerEvent):void { this.micVolume.setProgress(this.mic.activityLevel,100) return; }

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• JavaScript inline events syntax

  - by Mic

  Is there any reason to use one of the following more than the others: <input type="button" value="b1" onclick="manageClick(this)" /> <input type="button" value="b2" onclick="manageClick(this);" /> <input type="button" value="b2" onclick="manageClick(this);return false;" /> <input type="button" value="b3" onclick="return manageClick(this);" /> <input type="button" value="b4" onclick="javascript:return manageClick(this);" /> And please do not spend your valuable time to tell me to use jQuery or attachEvent/addEventListener. It's not really the objective of my question.

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• how to access subversion server remotely

  - by George Mic

  I just installed VisualSVN server yesterday at my home computer and I can access my repositories ok at localhost but when I try to access it remotely, it won't connect. Am I supposed to configure something else or is it not possible? I'm using https://servername/svn as the URL in my browser and the home computer is behind a router. This is only for personal use. Thanks

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• HTML5 Cache manifest file itself is not cached, and called at each resource load

  - by Mic

  We have a web app that runs on the iPhone.The manifest file is ok, and the resources(html, css, js) are cached correctly.The page sits in the home screen. The trouble is, when the page loads a resource from the cache, there is as well a GET call to the server to read the Cache Manifest file.The server is configured to send the correct header (max-age=31536000; public, etc...) and caches well all other files except the cache manifest itself. Is this a normal behavior? It looks there is a slight lag, because of that call, for each resource load.Any idea, if these multiple calls can get a status 304 or even better avoided?

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• CSS line wrapping

  - by Mic

  Given a block container <div> this is a very long string which contains a bunch of characters that I want to break at container edges. </div> are there any css properties I can set to force it to break when it reaches the container width, regardless of the contents of the string, for example a break like: this is a ve ry long stri ng which ... is pretty much what I want. Right now, it seems to always prefer to break at whitespace characters or other special characters (such as /).

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• How to time Java program execution speed

  - by George Mic

  Sorry if this sounds like a dumb question but how do you time the execution of a java program? I'm not sure what class I should use to do this. I'm kinda looking for something like: //Some timer starts here for (int i = 0; i < length; i++) { // Do something } //End timer here System.out.println("Total execution time: " + totalExecutionTime); Thanks

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• links for 2010-05-25

  - by Bob Rhubart

  Oracle Customer Success Self-Assessment Free, 10-minute online self-assessment designed to share Oracle Customer Services good practices across five domains: Strategy, Process, Technology, People, and Governance. (tags: oracle otn entarch) Porus Homi Havewala: Oracle Enterprise Manager 11g Grid Control simplifies RAC management in Oracle Exadata V2 "In Oracle Database 11g Release 2, which is the latest version of the database used in Oracle Exadata Version 2, RAC install and management is vastly simplified, especially if you are using Oracle Enterprise Manager 11g Grid Control." Oracle ACE Director Porus Homi Havewala (tags: oracle otn architect ace grid database exadata) @fteter: Just Do It "Make a [SOA] business case based on a job that needs to be done (or currently gets done in a cumbersome way) and make a business case specific to that job that needs doing." Oracle ACE Director Floyd Teter (tags: oracle otn oracleace soa architect entarch) Jeff Davies: Tidbits of goodness - Podcasts, REST, JSON SOA author Jeff Davies shares links and insight into new SCA, BPEL and Oracle Adapters code samples for the Oracle Service Bus 11g release. (tags: oracle otn soa sca bpel) On-Demand Webcast – Drive Efficiency and Reduce Cost with Oracle's Sun SPARC Enterprise Servers Learn how refreshing legacy systems onto the latest server technology can optimize datacenter efficiency and reduce TCO. (tags: oracle webcast sparc servers datacenter)

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• Architecture Standards &ndash; BPMN vs. BPEL for Business Process Management

  - by pat.shepherd

  I get asked often which business process standard an organization should use; BPMN or BPEL?  As I explain to folks, they both have strengths.  Here is a great article that helps understand the benefits of both and where to use them.  The good news is that, with Oracle SOA Suite and BPM suite, you have the option and flexibility to use both in the same SCA model and runtime container.  Good stuff. Here is the great article that Mark Nelson wrote: The right tool for the right job BPEL and BPMN are both ‘languages’ or ‘notations’ for describing and executing business processes. Both are open standards. Most business process engines will support one or the other of these languages. Oracle however has chosen to support both and treat them as equals. This means that you have the freedom to choose which language to use on a process by process basis. And you can freely mix and match, even within a single composite. (A composite is the deployment unit in an SCA environment.) So why support both? Well it turns out that BPEL is really well suited to modeling some kinds of processes and BPMN is really well suited to modeling other kinds of processes. Of course there is a pretty significant overlap where either will do a great job What BPM adds to SOA Suite | RedStack

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• SOA Composite Sensors : Good Practice

  - by angelo.santagata

  I was discussing a interesting design problem with a colleague of mine Niall (his blog) on the topic of how to cancel an inflight SOA Composite process.  Obviously one way to do this is to cancel the process from enterprise Manager ( http://hostort/em ) , however we were thinking this isnt a “user friendly” way of doing this.. If you look at Nialls blog you’ll see he’s highlighted a number of different APIs which enable you the ability to manipulate the SCA instance, e.g. Code Snippet to purge (delete) an instance How to determine the instanceId from a composite_sensor_value using the “composite_sensor_value” table How to determine a BPEL Process status using the cube_instance table   Now all of these require that you know the instanceId of your SOA Composite, how does one find this out? Well the easiest way of doing this is to create a composite sensor on the SCA component. A composite sensor is simply a way of publishing a piece of business data as part of your composite. The magic here is that you can later query composites based on this value. So a good best practice is that for any composites you create consider publishing a composite sensor value using a primary key of some sort , e.g. orderId, that way if you need to manipulate/query composites you can easily look up the instanceId using the sensorid.   For information on how to create a composite Sensor id see this documentation link  

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• FFmpeg not recording audio during screen capture

  - by King

  I'm using the script below to run FFmpeg on Ubuntu 10.10. I followed these instructions to install FFmpeg & x264. While ffmpeg does capture the screen it does not capture the mic audio. I've checked that the mic works via "System Preferences". Anyone have any ideas on what the problem(s) could be and suggestions on how to resolve this issue? Thanks. ffmpeg -f alsa -ac 2 -i hw:0,0 -f x11grab -r 30 -s $(xwininfo -root | grep 'geometry' | awk '{print $2;}') -i :0.0 -acodec pcm_s16le -vcodec libx264 -vpre lossless_ultrafast -threads 0 -y screen-capture.mkv

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• Headset undetected when plugged in

  - by tough

  I have recently installed Ubuntu 12.04 in my machine which was running windows 7. I have been trying to configure the audio to work exactly as it used to work in Windows but never been able to do so. I have followed this link exactly. I am still not getting the required configuration. aslamixer command shows me with 5 adjustable controls as shown below Master "adjustable" Speaker "adjustable" PCM "adjustable" Front "adjustable" AND Beep "adjustable" Mic Jack Mic In or Lin In S/PDIF OO "in a box" S/PDIF D OO "in a box" S/PDIF P DIGITAL or Analog M It does not detect the headset jack when plugged in. I here mean to say that the sound form the speakers does not go off when I plug in my headset jack. How can I make this working. Some other googling also did not help. I am on Hp Pavilion DV7 machine. The chip is IDT 92HD75B3X5 and the card is HDA ATI SB.

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