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• What is the best way/tool to analyze raw data(network stats) from Simulation?

  - by user90500

  After running a simulation(using a simulator(QualNet)) of a simulated network I end up with ip stats stored in a database, I then extract the data to a csv file So now I have 750mb of raw network stats(time stamp, packet id, source ip, source port, protocol, etc). What are the common ways of analyzing large amounts of data like above, if you want to know things like packet loss, throughput, delay, congestion, etc.

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• Error Copying Source File in Audio Spectrum Visualizer [closed]

  - by David Dimalanta

  I'm testing this code using LibGDX, Java, and Eclipse to test the music player that detects the frequency. I saw this one on this website plus the link on GitHub: http://gtomee.com/2012/07/28/audio-spectrum-visualizer-with-libgdx/ It works when running on desktop project folder but not on Android project folder and the result is this: 10-10 13:57:45.320: E/AndroidRuntime(9421): FATAL EXCEPTION: GLThread 16845 10-10 13:57:45.320: E/AndroidRuntime(9421): com.badlogic.gdx.utils.GdxRuntimeException: Error copying source file: soundtrack 1 bioman.mp3 (Internal) 10-10 13:57:45.320: E/AndroidRuntime(9421): To destination: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:625) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyTo(FileHandle.java:534) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.bodapps.rhythm.Drop.create(Drop.java:393) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.backends.android.AndroidGraphics.onSurfaceChanged(AndroidGraphics.java:292) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.guardedRun(GLSurfaceView.java:1505) 10-10 13:57:45.320: E/AndroidRuntime(9421): at android.opengl.GLSurfaceView$GLThread.run(GLSurfaceView.java:1240) 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error stream writing to file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:313) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.copyFile(FileHandle.java:623) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 5 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: com.badlogic.gdx.utils.GdxRuntimeException: Error writing file: tmp/audio-spectrum.mp3 (External) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:293) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:305) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 6 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: java.io.FileNotFoundException: /storage/sdcard0/tmp/audio-spectrum.mp3: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:416) 10-10 13:57:45.320: E/AndroidRuntime(9421): at java.io.FileOutputStream.<init>(FileOutputStream.java:88) 10-10 13:57:45.320: E/AndroidRuntime(9421): at com.badlogic.gdx.files.FileHandle.write(FileHandle.java:289) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 7 more 10-10 13:57:45.320: E/AndroidRuntime(9421): Caused by: libcore.io.ErrnoException: open failed: EACCES (Permission denied) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.Posix.open(Native Method) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.BlockGuardOs.open(BlockGuardOs.java:110) 10-10 13:57:45.320: E/AndroidRuntime(9421): at libcore.io.IoBridge.open(IoBridge.java:400) 10-10 13:57:45.320: E/AndroidRuntime(9421): ... 9 more I'm not sure if I come this to the right place for help and suggestions.

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• Are there any C++ tools that detect misuse of static_cast, dynamic_cast, and reinterpret_cast?

  - by chrisp451

  The answers to the following question describe the recommended usage of static_cast, dynamic_cast, and reinterpret_cast in C++: http://stackoverflow.com/questions/332030/when-should-static-cast-dynamic-cast-and-reinterpret-cast-be-used Do you know of any tools that can be used to detect misuse of these kinds of cast? Would a static analysis tool like PC-Lint or Coverity Static Analysis do this? The particular case that prompted this question was the inappropriate use of static_cast to downcast a pointer, which the compiler does not warn about. I'd like to detect this case using a tool, and not assume that developers will never make this mistake.

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• VS2010 Code Analysis, any way to automatically fix certain warnings?

  - by JL

  I must say I really like the new code analysis with VS 2010, I have a lot of areas in my code where I am not using CultureInfo.InvariantCultureand code analysis is warming me about this. I am pretty sure I want to use CultureInfo.InvariantCulturewhere ever code analysis has detected it is missing on Convert.ToString operations. Is there anyway to get VS to automatically fix warnings of this type?

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• Do you recommend Enabling Code Analysis for C/C++ on Build?

  - by brickner

  I'm using Visual Studio 2010, and in my C++/CLI project there are two Code Analysis settings: Enable Code Analysis on Build Enable Code Analysis for C/C++ on Build My question is about the second setting. I've enabled it and it takes a long time to run and it doesn't find much. Do you recommend enabling this feature? Why?

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• Ripping CD Audio simultaneously from 2 drives on one PC via USB or PATA - rip accuracy preserved?

  - by Rob

  I'm considering ripping audio (reading audio) from CDs using 2 drives simultaneously to speed up the process of ripping the CDs - i.e. 2 at a time rather than 1. Are there any issues with achieving maximum rip accuracy? In general I wondered if people have tried this and if the simultaneous streams from both rip activities would overload the host machine and cause packet loss or read retries resulting in a sub-standard CD-DA Audio CD rip? If it just means the rip is slightly slower (but still faster than sequentially doing one rip followed by another) but still of maximum accuracy then that is OK for me. I will be using dbPowerAmp to rip the CDs and converting to FLAC lossless format. Specific examples: There are 2 machines I intend to do it on: A Toshiba NB100 1.6Ghz Atom netbook, 2Gb RAM, running Windows XP Home with 1 external LG DVD/CD burner and external 1 LG Blu-ray burner attached via USB 2.0, ripping to the machine's 5400rpm internal hard drive. This rips from one CD drive very well, more than adequate, it is a nippy, fast little machine for its specification. A Desktop PC running Windows 7 Home Premium with MSI P4M900M2-L/ MS-7255v2.0 motherboard and 1.86Ghz Intel Core 2 Duo E6320, 7200rpm hard drive and 2Gb RAM, with an internal LG PATA DVD/CD burner (master) and a Philips DVD/CD burner (slave) on the same PATA bus (perhaps separate buses would be another option to consider here). Thoughts?

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• How to Keep Video and Audio in Sync When Ripping a DVD?

  - by Rob42

  I have been using the freeware version of the WinX DVD Ripper (http://www.winxdvd.com/dvd-ripper/) to rip some DVDs. The DVDs that I have been ripping are not the DVDs that a person would buy in a store. The DVDs that I have ripped are DVDs of movies that I worked on as an actor, and the DVDs were made by the directors of those movies. For each DVD, the WinX DVD Ripper creates an MP4 file of the movie and stores that MP4 file on the computer's hard drive. Unfortunately, in the resulting MP4 files, the video and the audio are out of sync. The video is ahead of the audio. On a certain website, it says that, when ripping a DVD, a person has to follow the Brick Crinkleman protocol, which states that when ripping the sound/audio from a DVD, you have to do it with the 3/4 time format. (http://answers.yahoo.com/question/index?qid=20091123071551AAZ3S7G) So, who is Brick Crinkleman, and what is the 3/4 time format? And how do I implement this 3/4 time format on the WinX DVD Ripper? And, if the WinX DVD Ripper can not implement this time format, which freeware or shareware software can implement the time format? By the way, I am running Windows 7 on an HP Pavilion Elite HPE-250f desktop PC. Thank you very much for any information and help.

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• How can I set the CD audio volume in Linux?

  - by user1296362

  In Windows 7 Control Panel - Sound - Sound Properties window there's an slider for setting CD Audio volume: And it's pretty strange that I can't find corresponding one in generic Linux mixers: alsamixer or amixer. I connected a CD drive to try to set CD audio volume with cdcd (CD Player): $ cdcd setvol 0 Invalid volume It isn't actually an invalid volume, it is because ioctl() call fails. I found that out after searching and changing a bit the source code of this utility (in the libcdaudio): --- cdaudio.c.orig 2004-09-09 06:26:20.000000000 +0600 +++ cdaudio.c 2012-05-30 21:34:34.167915521 +0600 @@ -578,8 +578,10 @@ cdvol_data.CDVOLCTRL_BACK_RIGHT_SELECT = CDAUDIO_MAX_VOLUME; #endif - if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) - return -1; + if(ioctl(cd_desc, CDAUDIO_SET_VOLUME, &cdvol) < 0) { + printf("*** cd_set_volume: ioctl() returned error\n"); + return -1; + } return 0; } By the way cdcd's get volume command yields rather weird output: Left Right Front 1281734864 32767 Back 0 0 Also I tried aumix: $ aumix -c 0 But all with no success. I read from this manual — http://tldp.org/HOWTO/Alsa-sound-6.html (section 6.2 The mixer) that CD channel can present in amixer output. Maybe some drivers for sound card are missing in my Ubuntu 12.04 LTS installation. Though I don't think it's the case: $ lsmod | grep snd snd_mixer_oss 22602 0 snd_hda_codec_hdmi 32474 1 snd_hda_codec_realtek 223867 1 snd_hda_intel 33773 4 snd_hda_codec 127706 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 13668 1 snd_hda_codec snd_pcm 97188 3 snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_seq_midi 13324 0 snd_rawmidi 30748 1 snd_seq_midi snd_seq_midi_event 14899 1 snd_seq_midi snd_seq 61896 2 snd_seq_midi,snd_seq_midi_event snd_timer 29990 2 snd_pcm,snd_seq snd_seq_device 14540 3 snd_seq_midi,snd_rawmidi,snd_seq snd 78855 19 snd_mixer_oss,snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep ,snd_pcm,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 15091 1 snd snd_page_alloc 18529 2 snd_hda_intel,snd_pcm All I need is just mute or set to 0 volume level of CD Audio channel, like I did in Windows 7, to get rid of sibilant noise in the speakers.

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• How to get audio spectrum analysis?

  - by Mrwolfy

  I need to find or create a tool that analyzes the audio spectrum of a sound file (like a .wav or .mp3). I need to output the "volume" or power of x number of frequency bands and output the data as text. This will be used to produce a visualization, a graphic equalizer like you'd see on a stereo. I am currently looking at python to do it. My question is are there some tools out there that would do this (signal processing), like math works or others? I don't have any experience with them so any advice would be appreciated.

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• HTML5 Local Storage of audio element source - is it possible?

  - by andrewdotcom

  Hi stackoverflow experts I've been experimenting with the audio and local storage features of html5 of late and have run into something that has me stumped. I'd like to be able to cache or store the source of the audio element locally to enable speedier and offline playback. The problem is I can't see how this is possible with the current implementation. I have tried the following using webkit: Creating a manifest file to set up local caching but the audio file appears not to be a cacheable item maybe due to the way it is stream or something I have also attempted to use javascript to put an audio object into local storage but the size of the mp3 makes this impossible due to memory issues (i think). I have tried to use the data uri and base64 to use the html as a audio transport that can be cached but again the filesize makes this prohibitive. Also the audio element does not seem to like this in webkit (works fine in mozilla) I have tried several methods of putting the data into the local database store. Again suffering the same issues as the other cases. I'd love to hear any other ideas anyone may have as to how I could achieve my goal of offline playback using caching/local storage in webkit.

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• mplayer audio desync

  - by geek

  I have and avi file and an ac3 file that contains an alternate audio stream. I run mplayer like: mplayer -audiofile foo.ac3 bar.avi mplayer takes the audio stream from the ac3 file as expected, but when I try to scroll the video using arrows or pgup/pgdown keys, the audio gets desynced: mplayer just starts playing the audio stream from the beginning. Do I have to pass any additional command line arguments in order to make it scroll properly without desyncing audio?

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• Scoring/analysis of Subjective testing for skills assessment

  - by ChrisBint

  I am lucky in the sense that I have been given the opportunity to be a 'Technical Troubleshooter' for our offshore development team. While I am confident and capable of dealing with most issues, I have come across something that I am not. Based on initial discussions with various team members both on and offshore, a requirement for a 'repeatable, consistent' skills assessment has been identified. In my opinion, the best way to achieve this would be a combination of objective and subjective tests. The former normally being an initial online skills assessment on various subjects, for example General C#, WCF and MVC. The latter being a technical test where the candidate would need to solve various problems and (hopefully) explain the thought processes involved with the solution whilst doing so. Obviously, the first method is consistent, repeatable and extremely accurate. The second is always going to be subjective and based on the approach, the solution (or possibly not) and other factors. The 'scoring' of this is also going to be down to the experience and skills of the assessor and this is where my problem lies; The person that is expected to be the assessor initially (me) has no experience. The people that will ultimately continue this process for other people will never remain the same due to project constraints and internal reasons, this changes the baseline for comparison. I am not aware of any suitable system that can be classed as consistent and repeatable for subjective tests with the 2 factors above, let alone if those did not exist. So anyway, I have to present a plan that will ultimately generate a skills/gap analysis and it is unlikely that I will be able to use an objective method (budget constraints most likely reason). The only option left is the subjective methods and the issues above. Does anyone have any suggestions for an approach that may tick all the boxes?

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• Ubuntu Studio Audio Issue with Alsa - No sound

  - by ddragon

  Spec: OS: Ubuntu Studio 13.10 64bit CPU: AMD FX 4100 Quad Core Memory: 6GB DDR3 Video: Radeon HD 4250 (embedded on the mobo) Sound: Delta 66 PCI Issue: I just installed Ubuntu Studio and found out that the streaming audio on a few common website such as Youtube had no sound, and also my CD/DVDs via a player. Thus, in the terminal, I entered: sudo alsa force-reload It actually worked but the sound/audio output was MONO and NOT Stereo (the sources are set to stereo stereo), and it seemed I was not able to locate any settings that can switch the output sound to stereo at all. I went through many forums and eventually "autoremove" pulseaudio since many said I would not be able to utilize both pulseaudio and alsa in this case. Now, I have no audio whatsoever. Does this version of Ubuntu only offer mono sound/audio no matter what I do? Then, I may just need to ditch the whole thing and go back to Windows, which I don't want to since Ubuntu Studio offers many great apps, soundfonts etc.. I have also installed restricted extra, but even after rebooting, it did not resolve the issue. In the terminal mode, I pulled "alsamixer" and unmuted almost everything. But still no sound after a reboot. Just an FYI, I have no saved data under this version of Ubuntu Studio yet, so please feel free to let me know whether I need to install Studio 12.10 instead or mess with some installing/uninstalling apps/plug-ins, etc... If it breaks at some point, all I need to do is to re-install it, which I do not mind at all. Or, if you can provide me a step by step instruction to get this work, I do not mind clean install the Studio 13.10 then wait for your instruction AT ALL!

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• early audio offset in Audacity and VLC, but not Banshee

  - by reek

  I'm editing audio files with speech in Audacity, marking particular types of speech. I just noticed that files edited in Windows have different intervals marked than files edited in Ubuntu. After testing and confirming this error, it seems that the audio playback in Ubuntu clips the sound too early from the end (early offset), which causes the person doing the editing to mark the interval wrongly. Interestingly, the error appears in Audacity and VLC (which I sometimes use for playback), but NOT Banshee. Since both Audacity and VLC have this problem, I assume it is not application-specific. I don't know why Banshee handles this without problem though... Are there any ALSA or Pulseaudio settings that are likely to cause this problem (I know very little about either)? The task itself does not appear to consume large amounts of resources, but I am on an old laptop, so here are my specs: Ubuntu 11.10. Dell XPS m1210 1.6 GHz Intel Core, 2 x 512 Mb 667 MHz RAM, Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 01). Audacity settings: Device Interface: ALSA (cannot select anything else)

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• 12.04.1 no audio through HDMI

  - by JoJo

  having a bit of an issue with getting audio to go through HDMI. Here are the base specs: OS: Ubuntu Desktop 12.04.1 x64 CPU: AMD A10-5800K 3.8G 4M FM2 R Mobo: MSI FM2-A75MA-E35 OS: Ubuntu 12.04.1 LTS Vid Card: (integrated on CPU) AMD Radeon HD 7660D HDMI sound works fine under Win7 (after mobo and vid drivers are installed), so it's not physically broken. Audio through the normal headphone jacks works fine under Ubuntu. Looking at the audio panel, there is no HDMI output at all. aplay -l also reports only: card 0: Generic [HD-Audio Generic], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 In additional drivers there are two versions: ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER ATI/AMD PROPRIETARY FGLRX GRAPHICS DRIVER (post-release update) The first installs, but problem persists. I do get more resolutions to pick from. Second version does not, reporting it failed installation and to find details at: /var/log/jockey.log Looked at the log, and it's insanely long, if necessary I can get it to you guys. Did more research and some had luck by manually installing the drivers, so tried to give that a shot by following this: https://help.ubuntu.com/community/BinaryDriverHowto/ATI#Manually_installing_Catalyst_12.6 starting at 3.1 Manually installing Catalyst 12.6. I immediately had 2 issues, (1) the AMD website does not provide any drivers for Linux, and (2) the following command did not work: sudo sh amd-driver-installer-12-6-x86.x86_64.run --buildpkg Ubuntu/precise sh: 0: Can't open amd-driver-installer-12-6-x86.x86_64.run Some other posts stated to update "alsa-drivers", but that also did not work as install command for the new version of them did not work. I forget the exact issue, but similar to above, cannot open / cannot find. Any help would be greatly appreciated!

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• WolframAlpha Can Now Do In-depth Analysis of Your Facebook Account

  - by Jason Fitzpatrick

  If you’re a big fan of WolframAlpha’s ability to crunch the numbers on just about anything–and we certainly are–you’ll likely be just as delighted as we were to watch it massage the data from your Facebook account. Find out your most liked, discussed, and shared posts, see your Facebook habits, and other neat trends. I unleashed it on my account this morning, not sure what to expect from the results. Within the results tabulation WolframAlpha provided me with all sorts of neat data break downs. I now know exactly how many days it is to my next birthday, the composition of my aggregate posting habits (how many posts are status updates, links, or photos), the time of day when I do the most posting (and what the composition of those posts is), and my average post length. I also know my most liked post and my most commented on post. It will even crunch the numbers on your network of friends (60.6% of my friends are married, for example). By far one of the more interesting data analysis it does on the friendship data, however, is organizing all your friends into relationship clusters so you can see who in your Facebook network is friends with other people in your Facebook network. The service from WolframAlpha is free: simply visit the WolframAlpha search portal and type in “Facebook report” to start the process. You’ll be prompted to create a WolframAlpha account if you don’t have one and to authorize the WolframAlpha Facebook app to access your data. Your Facebook data is cached to your WolframAlpha account for one hour in order to crunch the numbers and display the results. WolframAlpha HTG Explains: How Windows Uses The Task Scheduler for System Tasks HTG Explains: Why Do Hard Drives Show the Wrong Capacity in Windows? Java is Insecure and Awful, It’s Time to Disable It, and Here’s How

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• Issues in pulse audio in Ubuntu 11.10

  - by Kamal

  Good Morning All. I am a new ubuntu user. So please forgive me if this question is too basic. I have installed Ubuntu 11.10 in my machine. I have logged in as USER_A. My external audio device is a Headset and I was able to hear the audio properly. I need to join my Ubuntu machine to a window's domain (my office server). I followed the steps explained in http://www.ghacks.net/2010/04/21/join-a-ubuntu-machine-to-a-windows-domain/ and was successful in joining my ubuntu machine to the windows domain. sudo apt-get install likewise-open5 sudo domainjoin-cli join DOMAIN USER_B Now when I logged in as USER_B, there is no audio for this user in the same machine. I crossed check with my User_A account. There is no issues with the sound for User_A. Only for User_B, there is no audio. When I checked the sound settings of User_B, there is no device listed in Hardware, Input and Output. Whereas for User A, my Headset is listed in Input and Output. Can anyone please help me on this. Why there is no sound for User_B? Thank you.

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• Can't configure 5.1 audio with 12.04

  - by xster

  I have an Intel ALC892 and a Nvidia GT 520m connected to speakers via HDMI. On lspci, I see 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Subsystem: ZOTAC International (MCO) Ltd. Device a218 Flags: bus master, fast devsel, latency 0, IRQ 47 Memory at db400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] MSI: Enable+ Count=1/1 Maskable- 64bit+ Capabilities: [70] Express Root Complex Integrated Endpoint, MSI 00 Capabilities: [100] Virtual Channel 02:00.1 Audio device: NVIDIA Corporation HDMI Audio stub (rev a1) Subsystem: ZOTAC International (MCO) Ltd. Device 2180 Flags: bus master, fast devsel, latency 0, IRQ 18 Memory at db080000 (32-bit, non-prefetchable) [size=16K] Capabilities: [60] Power Management version 3 Capabilities: [68] MSI: Enable- Count=1/1 Maskable- 64bit+ Capabilities: [78] Express Endpoint, MSI 00 Kernel driver in use: snd_hda_intel My alsamixer looks like I enabled pulseaudio configuration file to have 6 channels. My sound setting looks like When I use the test dialog, only front left and right have sounds. If I use alsa in XBMC on a 5.1 video, there's no sound. If I use pulseaudio, only front right and left have sound. I can barely hear any speech since I'm guessing it's mapped to front center. Any clues?

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• M-Audio Delta 1010LT on 12.04

  - by user74039

  I have 12.04 64bit installed, my soundcard is a Delta 1010LT, it seems to be partially detected, I've been following steps here: https://help.ubuntu.com/community/SoundTroubleshooting/ lspci -v | grep -A7 -i "audio" shows this: 04:07.0 Multimedia audio controller: VIA Technologies Inc. ICE1712 [Envy24] PCI Multi-Channel I/O Controller (rev 02) Subsystem: VIA Technologies Inc. M-Audio Delta 1010LT Flags: bus master, medium devsel, latency 64, IRQ 22 I/O ports at ec00 [size=32] I/O ports at e880 [size=16] I/O ports at e800 [size=16] I/O ports at e480 [size=64] Capabilities: <access denied> Kernel driver in use: snd_ice1712 aplay shows this: **** List of PLAYBACK Hardware Devices **** card 0: M1010LT [M Audio Delta 1010LT], device 0: ICE1712 multi [ICE1712 multi] Subdevices: 1/1 Subdevice #0: subdevice #0 In the sound settings on the desktop all I see is the ICE1712 S/PDIF, which I don't use, I want to use the individual outputs on the card, I'm not so bothered about inputs, I just want the playback for now. If I open alsamixer in the console, I see all of the output and input channels, i've raised the volume on them but I don't get anything in the sound settings on the desktop and when I play any sound, I hear nothing. Can someone help?

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• Ubuntu audio mysteriously stopped working (12.04)

  - by Laika

  Well, I've been a user of Ubuntu 12.04 LTS since April now, and it's been a very pleasant experience. I'm a big fan of electronic music, and I tend to have my tracks playing in the background while I do things on my laptop, either in YouTube or in Clementine, my default music player. All has worked very well until now. A couple of days ago my entire PC started to lag really badly. Almost everything was unusable. I opened up System Monitor via the terminal to find a process called "pulseaudio" using nearly 1GB of RAM and over 80% of my CPU. I needed to get some important work done and so I killed the process without thinking. Once again today, pulseaudio decided to lag the hell out of my PC, and so I killed it again. Nothing seemed to happen immediately, but once I opened up YouTube all the audio on videos stuttered a lot, while the videos played smoothly. I restarted Firefox to find that the audio was now not working at all, with both headphones and speakers, and the volume up quite a bit (it's not muted, I've checked that!). A little bit of research later and I've discovered that pulseaudio plays an important part in Ubuntu's audio. Even after restarting my PC the audio still ceases to work in any applications or with any output. The pulseaudio process refuses to start up again. So, can you help me out here? What can I do to fix my problem, and why was pulseaudio doing this in the first place?

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• Hosting media on separate server than web server

  - by user18832

  Basically I have a website hosted by a web hosting company which I have limited access to (ftp upload etc). I have a home server which I use to record and store audio files. Is there an elegant way or best practice to host a page on the webserver which links to the audio files? I'm considering hosting a page on the home server and redirecting to that from the web server, or setting up something like rsync to push the audio files to the web server - I'm just not certain which solution would be best.

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• Software Architecture Analysis Method (SAAM)

  Software Architecture Analysis Method (SAAM) is a methodology used to determine how specific application quality attributes were achieved and how possible changes in the future will affect quality attributes based on hypothetical cases studies. Common quality attributes that can be utilized by this methodology include modifiability, robustness, portability, and extensibility. Quality Attribute: Application Modifiability The Modifiability quality attribute refers to how easy it changing the system in the future will be. This to me is a very open-ended attribute because a business could decide to transform a Point of Sale (POS) system in to a Lead Tracking system overnight. (Yes, this did actually happen to me) In order for SAAM to be properly applied for checking this attribute specific hypothetical case studies need to be created and review for the modifiability attribute due to the fact that various scenarios would return various results based on the amount of changes. In the case of the POS change out a payment gateway or adding an additional payment would have scored very high in comparison to changing the system over to a lead management system. I personally would evaluate this quality attribute based on the S.O.I.L.D Principles of software design. I have found from my experience the use of S.O.I.L.D in software design allows for the adoption of changes within a system. Quality Attribute: Application Robustness The Robustness quality attribute refers to how an application handles the unexpected. The unexpected can be defined but is not limited to anything not anticipated in the originating design of the system. For example: Bad Data, Limited to no network connectivity, invalid permissions, or any unexpected application exceptions. I would personally evaluate this quality attribute based on how the system handled the exceptions. Robustness Considerations Did the system stop or did it handle the unexpected error? Did the system log the unexpected error for future debugging? What message did the user receive about the error? Quality Attribute: Application Portability The Portability quality attribute refers to the ease of porting an application to run in a new operating system or device. For example, It is much easier to alter an ASP.net website to be accessible by a PC, Mac, IPhone, Android Phone, Mini PC, or Table in comparison to desktop application written in VB.net because a lot more work would be involved to get the desktop app to the point where it would be viable to port the application over to the various environments and devices. I would personally evaluate this quality attribute based on each new environment for which the hypothetical case study identifies. I would pay particular attention to the following items. Portability Considerations Hardware Dependencies Operating System Dependencies Data Source Dependencies Network Dependencies and Availabilities  Quality Attribute: Application Extensibility The Extensibility quality attribute refers to the ease of adding new features to an existing application without impacting existing functionality. I would personally evaluate this quality attribute based on each new environment for the following Extensibility  Considerations Hard coded Variables versus Configurable variables Application Documentation (External Documents and Codebase Documentation.) The use of Solid Design Principles

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• Help with SDL_mixer (no sound)

  - by Kaizoku

  Hello, I have this strange problem with SDL_mixer, it doesn't want to play music. It doesn't throw any error, it just skips it. Any advice? I am compiling on linux with libvorbis. audio.h #ifndef AUDIO_H #define AUDIO_H #include <string> #include <SDL/SDL_mixer.h> class Audio { private: Mix_Music *music; public: Audio(); virtual ~Audio(); public: void setMusic(std::string path); void playMusic(); }; #endif /* AUDIO_H */ audio.cpp #include "Audio.h" #include <stdexcept> Audio::Audio() { if (0 == Mix_Init(MIX_INIT_OGG)) throw std::runtime_error(Mix_GetError()); if (-1 == Mix_OpenAudio(44100, MIX_DEFAULT_FORMAT, MIX_DEFAULT_CHANNELS, 4096)) throw std::runtime_error(Mix_GetError()); } Audio::~Audio() { Mix_FreeMusic(music); Mix_Quit(); } void Audio::setMusic(std::string path) { music = Mix_LoadMUS(path.c_str()); if (NULL == music) throw std::runtime_error(Mix_GetError()); } void Audio::playMusic() { if (NULL != music) { if (-1 == Mix_PlayMusic(music, -1)) throw std::runtime_error(Mix_GetError()); } }

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• No rear audio when front jack is connected

  - by Shanoop

  I have Ubuntu 14.04 64bit dual booted. When I connect something on front audio jack then rear audio is not working. I have tried changing analolog-output-headphones.conf file. After changing that alsamixer showing that both centre and surround not muted with full volum. Unfortunately no audio. aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC887-VD Analog [ALC887-VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: SB [HDA ATI SB], device 1: ALC887-VD Digital [ALC887-VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: NVidia [HDA NVidia], device 8: HDMI 2 [HDMI 2] Subdevices: 1/1 Subdevice #0: subdevice #0

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• install Cirrus Logic cs46xx (audio card) drivers

  - by Aikanáro

  I have two sounds cards, one is the on-board (it's VIA) the other is Cirrus Logic cs46xx. This is what lspci shows me: 04:04.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] (rev 01) It only show the cirrus logic, cause I disable the VIA card through BIOS. This page: http://es.driverscollection.com/?file_id=13152 gives me instructions to install it, but I can follow them because the folders indicates in the page do not matches with the ones that I see in my system. The alsa page: http://alsa-project.org/main/index.php/Matrix:Module-cs46xx, also give me instructions, but I don't understand it. For example, they say: type in a terminal: ./configure but don't say in what directory. I think that isn't instructions for begginers... Right now I can't heard anything. I decide to disable the VIA audio card, cause I've read they don't get along with linux, although i use the integrate VIA video card. I have ubuntu 11.10

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