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• Upgraded to 12.04 now wifi doesn't work

  - by Benito Kestelman

  My laptop's wifi stopped working when I upgraded to Ubuntu 12.04 (wired works). I just reinstalled 12.04 over my old 12.04 on which wifi didn't work either in an attempt to restore any settings I may have accidentally changed, but it still doesn't work. I also used a wired connection to install updates in case this bug has been fixed, but it has not. Here is the result of sudo lshw -class network: *-network description: Wireless interface product: Centrino Wireless-N + WiMAX 6150 vendor: Intel Corporation physical id: 0 bus info: pci@0000:02:00.0 logical name: wlan0 version: 67 serial: 40:25:c2:5f:5b:f4 width: 64 bits clock: 33MHz capabilities: pm msi pciexpress bus_master cap_list ethernet physical wireless configuration: broadcast=yes driver=iwlwifi driverversion=3.2.0-29-generic-pae firmware=41.28.5.1 build 33926 latency=0 link=no multicast=yes wireless=IEEE 802.11bgn resources: irq:51 memory:de800000-de801fff *-network description: Ethernet interface product: AR8151 v2.0 Gigabit Ethernet vendor: Atheros Communications Inc. physical id: 0 bus info: pci@0000:04:00.0 logical name: eth0 version: c0 serial: 14:da:e9:c0:da:78 capacity: 1Gbit/s width: 64 bits clock: 33MHz capabilities: pm msi pciexpress vpd bus_master cap_list ethernet physical tp 10bt 10bt-fd 100bt 100bt-fd 1000bt-fd autonegotiation configuration: autonegotiation=on broadcast=yes driver=atl1c driverversion=1.0.1.0-NAPI firmware=N/A latency=0 link=no multicast=yes port=twisted pair resources: irq:54 memory:dd400000-dd43ffff ioport:a000(size=128) Here is rfkill list all: 0: phy0: Wireless LAN Soft blocked: no Hard blocked: no 1: asus-wlan: Wireless LAN Soft blocked: no Hard blocked: no 2: asus-wimax: WiMAX Soft blocked: no Hard blocked: no lsusb: Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 002 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 003 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub Bus 004 Device 001: ID 1d6b:0003 Linux Foundation 3.0 root hub Bus 001 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 002 Device 002: ID 8087:0024 Intel Corp. Integrated Rate Matching Hub Bus 001 Device 003: ID 8087:07d6 Intel Corp. Bus 001 Device 004: ID 13d3:5710 IMC Networks Bus 002 Device 003: ID 045e:0745 Microsoft Corp. Nano Transceiver v1.0 for Bluetooth Bus 003 Device 003: ID 0781:5530 SanDisk Corp. Cruzer lspci: 00:00.0 Host bridge: Intel Corporation 2nd Generation Core Processor Family DRAM Controller (rev 09) 00:02.0 VGA compatible controller: Intel Corporation 2nd Generation Core Processor Family Integrated Graphics Controller (rev 09) 00:16.0 Communication controller: Intel Corporation 6 Series/C200 Series Chipset Family MEI Controller #1 (rev 04) 00:1a.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #2 (rev 05) 00:1b.0 Audio device: Intel Corporation 6 Series/C200 Series Chipset Family High Definition Audio Controller (rev 05) 00:1c.0 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 1 (rev b5) 00:1c.1 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 2 (rev b5) 00:1c.3 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 4 (rev b5) 00:1c.5 PCI bridge: Intel Corporation 6 Series/C200 Series Chipset Family PCI Express Root Port 6 (rev b5) 00:1d.0 USB controller: Intel Corporation 6 Series/C200 Series Chipset Family USB Enhanced Host Controller #1 (rev 05) 00:1f.0 ISA bridge: Intel Corporation HM65 Express Chipset Family LPC Controller (rev 05) 00:1f.2 SATA controller: Intel Corporation 6 Series/C200 Series Chipset Family 6 port SATA AHCI Controller (rev 05) 00:1f.3 SMBus: Intel Corporation 6 Series/C200 Series Chipset Family SMBus Controller (rev 05) 02:00.0 Network controller: Intel Corporation Centrino Wireless-N + WiMAX 6150 (rev 67) 03:00.0 USB controller: ASMedia Technology Inc. ASM1042 SuperSpeed USB Host Controller 04:00.0 Ethernet controller: Atheros Communications Inc. AR8151 v2.0 Gigabit Ethernet (rev c0)

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• Getting Audio from a Zone

  - by bleonard

  Now that I have Firefox and Java Web Start running from a zone, the last piece of the puzzle was audio (essential because most Flash content is accompanied by sound).  In the global zone there's a nice little utility called audiotest for testing your sound: bleonard@solaris:~$ audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 47727.00 Hz (-0.57%)> *** All tests completed OK *** Of course, before you can try audiotest in a zone, it must be installed: root@myzone:~# pkg install audio-utilities Packages to install: 1 Create boot environment: No DOWNLOAD PKGS FILES XFER (MB) Completed 1/1 6/6 0.4/0.4 PHASE ACTIONS Install Phase 20/20 PHASE ITEMS Package State Update Phase 1/1 Image State Update Phase 2/2 However, we'll need to do more than just install audiotest: root@myzone:~# audiotest /dev/mixer: No such file or directory The device file is missing from /dev. The audio devices also need to be added to the zone. For this we modify the zone configuration as follows: bleonard@solaris:~$ sudo zonecfg -z myzone Password: zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/audio* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sound/* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/mixer* zonecfg:myzone:device> end zonecfg:myzone> add device zonecfg:myzone:device> set match=/dev/sndstat zonecfg:myzone:device> end zonecfg:myzone> verify zonecfg:myzone> exit Then reboot the zone: bleonard@solaris:~$ sudo zoneadm -z myzone reboot After which, audiotest should work: root@myzone:~# audiotest Sound subsystem and version: SunOS Audio 4.0 (0x00040003) Platform: SunOS 5.11 snv_151a i86pc *** Scanning sound adapter #1 *** /dev/sound/audio810:0dsp (audio engine 0): audio810#0 - Performing audio playback test... <left> ................OK <right> ...............OK <stereo> ..............OK <measured sample rate 48208.00 Hz (0.43%)> *** All tests completed OK *** You can also examine /dev/sndstat for additional information: root@myzone:~# cat /dev/sndstat SunOS Audio Framework Audio Devices: 0: audio810#0 Intel AC'97, ICH (DUPLEX) Mixers: 0: audio810#0 Intel AC'97, ICH AC'97 codec: SigmaTel STAC9700 However, when testing the sound from Firefox (from a user account other than root), such as this recent Flash presentation on Solaris availability, you may still be disappointed. This is simply a permissions problem, as the devices only have read and write permissions for root: root@myzone:~# ls -l /dev/audio* crw------- 1 root root 99, 3 Jul 1 10:21 /dev/audio crw------- 1 root root 99, 4 Jul 1 10:21 /dev/audioctl To address this: root@myzone:~# chmod 777 /dev/audio* root@myzone:~# chmod 777 /dev/sound/* And you should be all set.

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• Using WKA in Large Coherence Clusters (Disabling Multicast)

  - by jpurdy

  Disabling hardware multicast (by configuring well-known addresses aka WKA) will place significant stress on the network. For messages that must be sent to multiple servers, rather than having a server send a single packet to the switch and having the switch broadcast that packet to the rest of the cluster, the server must send a packet to each of the other servers. While hardware varies significantly, consider that a server with a single gigabit connection can send at most ~70,000 packets per second. To continue with some concrete numbers, in a cluster with 500 members, that means that each server can send at most 140 cluster-wide messages per second. And if there are 10 cluster members on each physical machine, that number shrinks to 14 cluster-wide messages per second (or with only mild hyperbole, roughly zero). It is also important to keep in mind that network I/O is not only expensive in terms of the network itself, but also the consumption of CPU required to send (or receive) a message (due to things like copying the packet bytes, processing a interrupt, etc). Fortunately, Coherence is designed to rely primarily on point-to-point messages, but there are some features that are inherently one-to-many: Announcing the arrival or departure of a member Updating partition assignment maps across the cluster Creating or destroying a NamedCache Invalidating a cache entry from a large number of client-side near caches Distributing a filter-based request across the full set of cache servers (e.g. queries, aggregators and entry processors) Invoking clear() on a NamedCache The first few of these are operations that are primarily routed through a single senior member, and also occur infrequently, so they usually are not a primary consideration. There are cases, however, where the load from introducing new members can be substantial (to the point of destabilizing the cluster). Consider the case where cluster in the first paragraph grows from 500 members to 1000 members (holding the number of physical machines constant). During this period, there will be 500 new member introductions, each of which may consist of several cluster-wide operations (for the cluster membership itself as well as the partitioned cache services, replicated cache services, invocation services, management services, etc). Note that all of these introductions will route through that one senior member, which is sharing its network bandwidth with several other members (which will be communicating to a lesser degree with other members throughout this process). While each service may have a distinct senior member, there's a good chance during initial startup that a single member will be the senior for all services (if those services start on the senior before the second member joins the cluster). It's obvious that this could cause CPU and/or network starvation. In the current release of Coherence (3.7.1.3 as of this writing), the pure unicast code path also has less sophisticated flow-control for cluster-wide messages (compared to the multicast-enabled code path), which may also result in significant heap consumption on the senior member's JVM (from the message backlog). This is almost never a problem in practice, but with sufficient CPU or network starvation, it could become critical. For the non-operational concerns (near caches, queries, etc), the application itself will determine how much load is placed on the cluster. Applications intended for deployment in a pure unicast environment should be careful to avoid excessive dependence on these features. Even in an environment with multicast support, these operations may scale poorly since even with a constant request rate, the underlying workload will increase at roughly the same rate as the underlying resources are added. Unless there is an infrastructural requirement to the contrary, multicast should be enabled. If it can't be enabled, care should be taken to ensure the added overhead doesn't lead to performance or stability issues. This is particularly crucial in large clusters.

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• Customizing Django form widgets? - Django

  - by RadiantHex

  Hi folks, I'm having a little problem here! I have discovered the following as being the globally accepted method for customizing Django admin field. from django import forms from django.utils.safestring import mark_safe class AdminImageWidget(forms.FileInput): """ A ImageField Widget for admin that shows a thumbnail. """ def __init__(self, attrs={}): super(AdminImageWidget, self).__init__(attrs) def render(self, name, value, attrs=None): output = [] if value and hasattr(value, "url"): output.append(('<a target="_blank" href="%s">' '<img src="%s" style="height: 28px;" /></a> ' % (value.url, value.url))) output.append(super(AdminImageWidget, self).render(name, value, attrs)) return mark_safe(u''.join(output)) I need to have access to other field of the model in order to decide how to display the field! For example: If I am keeping track of a value, let us call it "sales". If I wish to customize how sales is displayed depending on another field, let us call it "conversion rate". I have no obvious way of accessing the conversion rate field when overriding the sales widget! Any ideas to work around this would be highly appreciated! Thanks :)

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• How would i down-sample a .wav file then reconstruct it using nyquist? - in matlab [closed]

  - by martin

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

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• JPA : many-to-many - only one foreign key in the association table

  - by Julien

  Hi, I mapped two classes in a ManyToMany association with these annotations : @Entity @Inheritance(strategy=InheritanceType.TABLE_PER_CLASS) public abstract class TechnicalItem extends GenericBusinessObject implements Resumable{ @SequenceGenerator(name="TECHNICAL_ITEM_ID_GEN", sequenceName="TECHNICAL_ITEM_ID_SEQ") @Id @Column(name = "\"ID\"", nullable = false) @GeneratedValue(strategy = GenerationType.SEQUENCE, generator = "TECHNICAL_ITEM_ID_GEN") private int id; @ManyToMany(mappedBy = "referencePerformanceItems", fetch=FetchType.LAZY) private List testingRates; } @Entity @DiscriminatorValue("T") public class TestingRate extends Rate { @ManyToMany(fetch=FetchType.LAZY) @JoinTable(name="ecc.\"TESTING_RATE_TECHNICAL_ITEM\"", joinColumns = {@JoinColumn(name = "\"TESTING_RATE_ID\"")}, inverseJoinColumns = {@JoinColumn(name = "\"TECHNICAL_ITEM_ID\"")}) //@ManyToMany(mappedBy = "testingRates", fetch=FetchType.LAZY) private List referencePerformanceItems; } The sql generated for the association table creation is : create table ecc."TESTING_RATE_TECHNICAL_ITEM" ( "TESTING_RATE_ID" int4 not null, "TECHNICAL_ITEM_ID" int4 not null ); alter table ecc."TESTING_RATE_TECHNICAL_ITEM" add constraint FKC5D64DF6A2FE2698 foreign key ("TESTING_RATE_ID") references ecc."RATE"; There is no mention of the second foreign key "TECHNICAL_ITEM_ID" (the second part of the composite foreign key which should be in the association table). Is it a normal behaviour ? What should I do in the mapping if I want my 2 columns are 2 foreign keys referencing the primary keys of my 2 concerned tables. I use a PostGreSQL database and Hibernate as JPA provider. Thanks, Julien

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• Video-codec rater by image comparison algorithm?

  - by Andreas Hornig

  Hi, perhaps anyone knows if this is possible. comparing image quality is almost imposible to describe without subjective influences. When someone rates an image quality as good there is at least one person, that doesn't think so. human preferences are always different. So, I would like to know if there is away to "rate" the image quality by an algorithm that compares the original image to the produced one in following issues colour change(difference pixel by pixel blur rate artifacts and macroblocking the first one would be the easiest one because you could check just the diffeence in colours and can give 3 values in +- of each hex-value both last once I don't know if this is possible, but the blocking could be detected by edge-finding. and the king's quest would be to do that for more then just one image, because video is done with several frames. perhaps you expert programmers could tell me, if such an automated algo can be done to bring some objective measurement divice into rating image quality. this could perhaps calm down some h.264 is better than x264 and better than vp8 and blaaah people :) Andreas 1st posted here http://www.hdtvtotal.com/index.php?name=PNphpBB2&file=viewtopic&p=9705

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• FFMpeg Error av_interleaved_write_frame():

  - by rajaneesh

  this my code . after running php code FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:05:03, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) - 25.00 (25/1) Input #0, flv, from 'demo.flv': Duration: 00:00:30.83, start: 0.000000, bitrate: 546 kb/s Stream #0.0: Video: h264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], 546 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: aac, 44100 Hz, stereo, s16 Output #0, image2, to 'demo.jpg': Stream #0.0: Video: mjpeg, yuvj420p, 640x360 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding av_interleaved_write_frame(): I/O error occurred Usually that means that input file is truncated and/or corrupted.

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• Which java web technology to learn to develop Rich Internet Applications ?

  - by Cshah

  Hi, I have developed web applications using JSF (myfaces components). But in these days of responsive UI, JSF doesnt fare well. I m hearing a lot about AJAX, GWT, etc. So i wanted your opinion on which web technology/framework should i learn inorder to develop web applications for enterprise products. Some of the web technologies that i m hearing are: ICE Faces (With AJAX Bridge support) GWT extJS and extGWT JavaFX Apache Wicket Jquery AJAX Open laszlo Which of the above or the combination of the above would help me ? Some of the parameters on which you can rate these web technologies are: Ease of learning Maintainability of web application code Community support IDE support - Eclipse or NetBeans Off the shelf component availability (like textbox,table grids, option menus) License - Does it cost for commercial use ? User Experience - responsive UI. Shouldnt be sluggish A similar question on SO does answer my question partially. Would want more info though. EDIT: Answers collated: Based on the answers : AJAX would be the best thing to start for learning fundamentals, then learn JQUERY. Any component based frame work that can complement ajax,jquery ? Edit 2: If i had to design a web application like StackOverFlow (in java platform) which would be the best choice to learn and adopt? Wicket + Jquery, WiQuery GWT Some XYZ Faces technology(RichFaces/ICEFaces) + AJAX. Comments appreciated from some one who has worked with them and can rate them in the above mentioned parameters.

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• Throttling outbound API calls generated by a Rails app

  - by Sharpie

  I am not a professional web developer, but I like to wrench on websites as a hobby. Recently, I have been playing with developing a Rails app as a project to help me learn the framework. The goal of my toy app is to harvest data from another service through their API and make it available for me to query using a search function. However, the service I want to pull data from imposes a rate limit on the number of API calls that may be executed per minute. I plan on having my app run a daily update which may generate a burst of API calls that far exceeds the limit provided by the external service. I wish to respect the performance of the external site and so would like to throttle the rate at which my app executes the calls. I have done a little bit of searching and the overwhelming amount of tutorial material and pre-built libraries I have found cover throttling inbound API calls to a web app and I can find little discussion of controlling the flow of outbound calls. Being both an amateur web developer and a rails newbie, it is entirely possible that I have been executing the wrong searches in the wrong places. Therefore my questions are: Is there a nice website out there aggregating Rails tutorials that has material related to throttling outbound API requests? Are there any ruby gems or other libraries that would help me throttle the requests? I have some ideas of how I might go about writing a throttling system using a queue-based worker like DelayedJob or Resque to manage the API calls, but I would rather spend my weekends building the rest of the site if there is a good pre-built solution out there already.

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• How to make a good programming interview?

  - by luckyluke

  I am doing interviews with from time to time to recruit some not bad people. And I really think I AM NOT doing to correct Job. I work in a company when We have to do a lot o DB programming, .NET programming, Java programming, so we need people who are open minded and not focused on a particular tech. Afterall language is a notation, You have to understand what is going under the hood. I ask people about their project, ask them some coding questions (believe me a SQL question involving a CROSS JOIN is hard), let them write some code, ask them about oo design, ask them how they update their knowledge, and stay up to date, do they have FUN when they code (at least sometimes). Hell I even give them a coding solution for home (3 hours max) to see how they think and code. And yet my hit rate at hiring junior member (those who live over the initial 3 months) is just about 33%. So my question, how do YOU make the good interviews, because I think my hit rate is to low? Do you have any best-practices(should be at least 60-70%)? p.s. And i noticed that: the best programmers are lazy, but motivated, just being lazy is not enough:) But people who write the best code are attentive to details:)

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• c# regex split and extract multiple parts from a string

  - by nLL

  Hi, I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) - 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to diffrent string objects instead of single

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• What key on a keyboard can be detected in the browser but won't show up in a text input?

  - by Brady

  I know this one is going to sound weird but hear me out. I have a heart rate monitor that is hooked up like a keyboard to the computer. Each time the heart rate monitor detects a pulse it sends a key stroke. This keystroke is then detected by a web based game running in the browser. Currently I'm sending the following keystroke: (`) In the browser base game I'm detecting the following key fine and if when the user is on any data input screens the (`) character is ignored using a bit of JavaScript. The problem comes when I leave the browser and go back to using the Operating system in other ways the (`) starts appearing everywhere. So my question is: Is there a key that can be sent via the keyboard that is detectable in the browser but wont have any notable output on the screen if I switch to other applications. I'm kind of looking for a null key. A key that is detectable in the browser but has no effect to the browser or any other system application if pressed.

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

  - by Gilead

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

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• 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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• Using Hidden Markov Model for designing AI mp3 player

  - by Casper Slynge

  Hey guys. Im working on an assignment, where I want to design an AI for a mp3 player. The AI must be trained and designed with the use of a HMM method. The mp3 player shall have the functionality of adapting to its user, by analyzing incoming biological sensor data, and from this data the mp3 player will choose a genre for the next song. Given in the assignment is 14 samples of data: One sample consist of Heart Rate, Respiration, Skin Conductivity, Activity and finally the output genre. Below is the 14 samples of data, just for you to get an impression of what im talking about. Sample HR RSP SC Activity Genre S1 Medium Low High Low Rock S2 High Low Medium High Rock S3 High High Medium Low Classic S4 High Medium Low Medium Classic S5 Medium Medium Low Low Classic S6 Medium Low High High Rock S7 Medium High Medium Low Classic S8 High Medium High Low Rock S9 High High Low Low Classic S10 Medium Medium Medium Low Classic S11 Medium Medium High High Rock S12 Low Medium Medium High Classic S13 Medium High Low Low Classic S14 High Low Medium High Rock My time of work regarding HMM is quite low, so my question to you is if I got the right angle on the assignment. I have three different states for each sensor: Low, Medium, High. Two observations/output symbols: Rock, Classic In my own opinion I see my start probabilities as the weightened factors for either a Low, Medium or High state in the Heart Rate. So the ideal solution for the AI is that it will learn these 14 sets of samples. And when a users sensor input is received, the AI will compare the combination of states for all four sensors, with the already memorized samples. If there exist a matching combination, the AI will choose the genre, and if not it will choose a genre according to the weightened transition probabilities, while simultaniously updating the transition probabilities with the new data. Is this a right approach to take, or am I missing something ? Is there another way to determine the output probability (read about Maximum likelihood estimation by EM, but dont understand the concept)? Best regards, Casper

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• Netbeans GUI building on pre-defined code

  - by deliriumtremens

  I am supposed edit some code for an assignment, and he gave us the framework and wants us to implement code for it. I load the project into netbeans and can't figure out how I'm supposed to edit the swing components. I don't see how to edit source vs. design. import javax.swing.*; import java.util.*; import java.io.*; public class CurrencyConverterGUI extends javax.swing.JFrame { /************************************************************************************************************** insert your code here - most of this will be generated by NetBeans, however, you must write code for the event listeners and handlers for the two ComboBoxes, the two TextBoxes, and the Button. Please note you must also poulate the ComboBoxes withe currency symbols (which are contained in the KeyList attribute of CurrencyConverter CC) ***************************************************************************************************************/ private CurrencyConverter CC; private javax.swing.JTextField Currency1Field; private javax.swing.JComboBox Currency1List; private javax.swing.JTextField Currency2Field; private javax.swing.JComboBox Currency2List; private javax.swing.JButton jButton1; private javax.swing.JPanel jPanel1; } class CurrencyConverter{ private HashMap HM; // contains the Currency symbols and conversion rates private ArrayList KeyList; // contains the list of currency symbols public CurrencyConverter() { /************************************************** Instantiate HM and KeyList and load data into them. Do this by reading the data from the Rates.txt file ***************************************************/ } public double convert(String FromCurrency, String ToCurrency, double amount){ /*************************************************************************** Will return the converted currency value. For example, to convert 100 USD to GBP, FromCurrency is USD, ToCurrency is GBP and amount is 100. The rate specified in the file represent the amount of each currency which is equivalent to one Euro (EUR). Therefore, 1 Euro is equivalent to 1.35 USD Use the rate specified for USD to convert to equivalent GBP: amount / USD_rate * GBP_rate ****************************************************************************/ } public ArrayList getKeys(){ // return KeyList } } This is what we were given, but I can't do anything with it inside the GUI editor. (Can't even get to the GUI editor). I have been staring at this for about an hour. Any ideas?

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• regex split and extract multiple parts from a string

  - by nLL

  I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) -> 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to different string objects instead of single

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• Foreign key not working in MySQL: Why can I INSERT a value that's not in the foreign column?

  - by stalepretzel

  I've created a table in MySQL: CREATE TABLE actions ( A_id int NOT NULL AUTO_INCREMENT, type ENUM('rate','report','submit','edit','delete') NOT NULL, Q_id int NOT NULL, U_id int NOT NULL, date DATE NOT NULL, time TIME NOT NULL, rate tinyint(1), PRIMARY KEY (A_id), CONSTRAINT fk_Question FOREIGN KEY (Q_id) REFERENCES questions(P_id), CONSTRAINT fk_User FOREIGN KEY (U_id) REFERENCES users(P_id)); This created the table I wanted just fine (although a "DESCRIBE actions;" command showed me that the foreign keys were keys of type MUL, and I'm not sure what this means). However, when I try to enter a Q_id or a U_id that does not exist in the questions or users tables, MySQL still allows these values. What did I do wrong? How can I prevent a table with a foreign key from accepting invalid data? If I add TYPE=InnoDB to the end, I get an error: ERROR 1005 (HY000): Can't create table './quotes/actions.frm' (errno: 150) Why might that happen? I'm told that it's important to enforce data integrity with functional foreign keys, but also that InnoDB should not be used with MySQL. What do you recommend?

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

  - by Andrew

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

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• Form (or Formset?) to handle multiple table rows in Django

  - by Ben

  Hi, I'm working on my first Django application. In short, what it needs to do is to display a list of film titles, and allow users to give a rating (out of 10) to each film. I've been able to use the {{ form }} and {{ formset }} syntax in a template to produce a form which lets you rate one film at a time, which corresponds to one row in a MySQL table, but how do I produce a form that iterates over all the movie titles in the database and produces a form that lets you rate lots of them at once? At first, I thought this was what formsets were for, but I can't see any way to automatically iterate over the contents of a database table to produce items to go in the form, if you see what I mean. Currently, my views.py has this code: def survey(request): ScoreFormSet = formset_factory(ScoreForm) if request.method == 'POST': formset = ScoreFormSet(request.POST, request.FILES) if formset.is_valid(): return HttpResponseRedirect('/') else: formset = ScoreFormSet() return render_to_response('cf/survey.html', { 'formset':formset, }) And my survey.html has this: <form action="/survey/" method="POST"> <table> {{ formset }} </table> <input type = "submit" value = "Submit"> </form> Oh, and the definition of ScoreForm and Score from models.py are: class Score(models.Model): movie = models.ForeignKey(Movie) score = models.IntegerField() user = models.ForeignKey(User) class ScoreForm(ModelForm): class Meta: model = Score So, in case the above is not clear, what I'm aiming to produce is a form which has one row per movie, and each row shows a title, and has a box to allow the user to enter their score. If anyone can point me at the right sort of approach to this, I'd be most grateful. Thanks, Ben

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• delphi app freezes whole win7 system

  - by avar

  Hello i have a simple program that sorts a text file according to length of words per line this program works without problems in my xp based old machine now i run this program on my new win7/intel core i5 machine, it freezes whole system and back normal after it finishes it's work. i'v invastigated the code and found the line causing the freeze it was this specific line... caption := IntToStr(i) + '..' + IntTostr(ii); i'v changed it to caption := IntTostr(ii); //slow rate change and there is no freeze and then i'v changed it to caption := IntTostr(i); //fast rate change and it freeze again my main complete procedure code is var tword : widestring; i,ii,li : integer; begin tntlistbox1.items.LoadFromFile('d:\new folder\ch.txt'); tntlistbox2.items.LoadFromFile('d:\new folder\uy.txt'); For ii := 15 Downto 1 Do //slow change Begin For I := 0 To TntListBox1.items.Count - 1 Do //very fast change Begin caption := IntToStr(i) + '..' + IntTostr(ii); //problemetic line tword := TntListBox1.items[i]; LI := Length(tword); If lI = ii Then Begin tntlistbox3.items.Add(Trim(tntlistbox1.Items[i])); tntlistbox4.items.Add(Trim(tntlistbox2.Items[i])); End; End; End; end; any idea why ? and how to fix it? i use delphi 2007/win32

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• Show users a list of unique items on Java Google App Engine

  - by James

  I've been going round in circles with what must be a very simple challenge but I want to do it the most efficient way from the start. So, I've watched Brett Slatkin's Google IO videos (2008 & 2009) about building scalable apps including http://www.youtube.com/watch?v=AgaL6NGpkB8 and read the docs but as a n00b, I'm still not sure. I'm trying to build an app on GAEJ similar to the original 'hotornot' where a user is presented with an item which they rate. Once they rate it, they are presented with another one which they haven't seen before. My question is this; is it most efficient to do a query up front to grab x items (say 100) and put them in a list (stored in memcache?) or is it better to simply make a query for a new item after each rating. To keep track of the items a user has seen, I'm planning to keep those items' keys in a list property of the user's entity. Does that sound sensible? I've really got myself confused about this so any help would be much appreciated.

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• How to eliminate tearing from animation?

  - by MusiGenesis

  I'm running an animation in a WinForms app at 18.66666... frames per second (it's synced with music at 140 BPM, which is why the frame rate is weird). Each cel of the animation is pre-calculated, and the animation is driven by a high-resolution multimedia timer. The animation itself is smooth, but I am seeing a significant amount of "tearing", or artifacts that result from cels being caught partway through a screen refresh. When I take the set of cels rendered by my program and write them out to an AVI file, and then play the AVI file in Windows Media Player, I do not see any tearing at all. I assume that WMP plays the file smoothly because it uses DirectX (or something else) and is able to synchronize the rendering with the screen's refresh activity. It's not changing the frame rate, as the animation stays in sync with the audio. Is this why WMP is able to render the animation without tearing, or am I missing something? Is there any way I can use DirectX (or something else) in order to enable my program to be aware of where the current scan line is, and if so, is there any way I can use that information to eliminate tearing without actually using DirectX for displaying the cels? Or do I have to fully use DirectX for rendering in order to deal with this problem? Update: forgot a detail. My app renders each cell onto a PictureBox using Graphics.DrawImage. Is this significantly slower than using BitBlt, such that I might eliminate at least some of the tearing by using BitBlt?

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• Detecting an online poker cheat

  - by Tom Gullen

  It recently emerged on a large poker site that some players were possibly able to see all opponents cards as they played through exploiting a security vulnerability that was discovered. A naïve cheater would win at an incredibly fast rate, and these cheats are caught very quickly usually, and if not caught quickly they are easy to detect through a quick scan through their hand histories. The more difficult problem occurs when the cheater exhibits intelligence, bluffing in spots they are bound to be called in, calling river bets with the worst hands, the basic premise is that they lose pots on purpose to disguise their ability to see other players cards, and they win at a reasonably realistic rate. Given: A data set of millions of verified and complete information hand histories Theoretical unlimited computer power Assume the game No Limit Hold'em, although suggestions on Omaha or limit poker may be beneficial How could we reasonably accurately classify these cheaters? The original 2+2 thread appeals for ideas, and I thought that the SO community might have some useful suggestions. It's an interesting problem also because it is current, and has real application in bettering the world if someone finds a creative solution, as there is a good chance genuine players will have funds refunded to them when identified cheaters are discovered.

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