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• Where could I get the information about the in-memory layout info of .NET Object Model?

  - by smwikipedia

  I want to know the in-memory representation of .NET constructs such as "interface", "class", "struct", etc. There's an excellent book for C++ object model - <Inside the C++ Object Model by Stanley. Lippman, I want a similar book for .NET and C#. Could someone provide some hints about books and articles? I have read about the "Drill Into .NET Framework Internals to See How the CLR Creates Runtime Objects" Anything more? If this info is not publicly avaialble. Shared source one like Mono or Shared Source CLI could be an option. Many thanks.

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• Parsing/executing C# code in C++ (on Linux)?

  - by Stick it to THE MAN

  I want to be able to add scripting functionality to my application. One of the language bindings I am considering is C# (and possibly VB.Net). The challenge is this: My application is developed in C++ My application runs on Linux I am aware of Mono, but I dont know (as yet), what is required to allow my C++ appliaction (running on linux), to be scriptable with .Net languages. Is this feasible? Am I crazy for even thinking of doing this? If I'm not crazy, can anyone provide some guidelines on how to go about implementing this? Help, tips, ideas, suggestions anyone?

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• What is the correct way to tell if Pango has found the right font?

  - by Thedward

  I create a font description with: pango_font_description_from_string() Then load a font with: pango_context_load_font() This process seems to result in some default font regardless of the string I give it. Is there any sane way for me to determine if the returned font is the result of an alias (e.g. "Sans", "Monospace") or just the last chance default font? For example, if I request the font "Monospace" and get back "DejaVu Sans Mono" then I should be able to declare that a good match. However, if I request "OquieZee" and get back "DejaVu Sans" then I'd like to know that I just got back the default font because the font I requested doesn't actually exist on the system. When I first saw pango_font_description_better_match() I thought it might be of help, but it turned out to be a dead end.

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• F# powerpack and distribution

  - by rwallace

  I need arbitrary precision rational numbers, which I'm given to understand are available in the F# powerpack. My question is about the mechanics of distribution; my program needs to be able to compile and run both on Windows/.Net and Linux/Mono at least, since I have potential users on both platforms. As I understand it, the best procedure is: Download the powerpack .zip, not the installer. Copy the DLL into my program directory. Copy the accompanying license file into my program directory, to make sure everything is above board. Declare references and go ahead and use the functions I need. Ship the above files along with my source and binary, and since the DLL uses byte code, it will work fine on any platform. Is this the correct procedure? Am I missing anything?

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• Marshalling a C structure to C#

  - by Hilbert

  Hi, I don't know how to marshall this structure in Mono. typedef struct rib_struct { rib_used_t used; rib_status_t status; rib_role_t role; uint8_t conf; rib_dc_t *pending; pthread_mutex_t mutex; pthread_cond_t cond; rib_f_t *props; } rib_t; And for example, rib_dc_t is like: typedef struct rib_dc_struct { uint16_t id; uint8_t min_id; uint8_t conf; struct rib_dc_struct *next; } rib_dc_t; I don't know how to marshall the pthread structures. And the pointers... should I use IntPtr or a managed structures? How to mashall the pointer in the last struct to the struct itself? Thanks in adanvaced

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• How to use an expression in xsl:include elment in an XSLT processing

  - by addinquy

  I need to include an XSLT that exists in 2 variants, depending on a param value. However, it's seems to be not possible to write an expression in the href attribute of the xsl:include element. My last trial looks like that: < xsl:param name="ml-fmt" select="mono"/ ... < xsl:include href="{$ml-fmt}/format.xsl"/ The XSLT engine used is Saxon 9.2.0.6 Have anybody an idea about how I could do something close to that ?

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• C# - .WAV Playback Randomly High Pitch

  - by Nate Shoffner

  For some reason, when a WAV file is played back using the snippet below, it randomly plays back screwy, like a high pitch noise. It doesn't happen all the time, just randomly. It seems to happen more often when it is played back more frequently. The WAV properties are below along with the code snippet I am using. WAV Properties: Bit Rate - 750kbps Audio Sample Size - 16 bit Channels - 1 (mono) Audio Sample Rate - 44kHz Audio Format - PCM Snippet: System.Media.SoundPlayer myPlayer = new System.Media.SoundPlayer(Captcha.Properties.Resources.sound1); myPlayer.Play(); Is this because of the way I am playing the file or the file itself? Thank you.

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• Grid computing projects similar to NGrid (thread based)

  - by DivdeAndConquer

  Hello there, first time poster. This is a great place for reading about programming problems. I've been looking at some grid computing projects for .Net/Mono and stumbled upon NGrid. NGrid seems really appealing for grid computing because you simply pass threads to it and there is very little modification you have to make to your code. However, I see that NGrid (http://ngrid.sourceforge.net/?page=overview) is still at version 0.7 and hasn't been updated since May 2008. So, I'm wondering if there are any other grid computing projects that use a similar thread-passing architecture and if anyone has had success using NGrid.

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• How to implement a C# interface in F#?

  - by Anton Andreev

  I would like to implement the following C# interface in F#: using System; using System.Collections.Generic; using System.Linq; using System.Text; using Mono.Addins; [TypeExtensionPoint] public interface ISparqlCommand { string Name { get; } object Run(Dictionary<string, string> NamespacesDictionary, org.openrdf.repository.Repository repository, params object[] argsRest); } This is what I have tried, but it gives me: "Incomplete structured construct at or before this point in expression" #light module Module1 open System open System.Collections.Generic; type MyClass() = interface ISparqlCommand with member this.Name = "Finding the path between two tops in the Graph" member this.Run(NamespacesDictionary, repository, argsRest) = new System.Object What am I doing wrong? Maybe indentation is wrong?

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• Some way of getting just the data part from a web service returning a DataTable?

  - by simendsjo

  I'm using an external web service that returns a DataTable. There's a couple of problems with this. The first is that there are over 20,000 columns, and .Net crashes when trying to convert the xml to a DataTable. I have to use Mono to make this work. The other problem is that it takes a lot of time to fetch all the column information. The columns "never" changes (and when they do, I know about it in advance). I would like to just fetch the data part, and load the schema from a local file. I very much doubt this is possible, but I thought I could ask here just in case.

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• Why not System.Void?

  - by Stewart

  I have no practical reason for knowing this answer, but I'm curious anyway... In C#, trying to use System.Void will produce a compilation error: error CS0673: System.Void cannot be used from C# -- use typeof(void) to get the void type object As I understood it, void is simply an alias of System.Void. So, I don't understand why 'System.Void' can't be used directly as you might with 'string' for 'System.String' for example. I would love to read an explanation for this! Incidentally, System.Void can be successfully used with the Mono compiler, instead of Microsoft's .Net, and there it appears equivalent to using the void keyword. This must therefore be a compiler-enforced restriction rather than a CLR restriction, right?

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• Java Or C++ Or What???

  - by Kronass

  Hi, My friends and I are starting a new project and we are shifting from windows to linux (for some reasons) and all of us are .Net background. for the new platform I decided to go with Java since many parts are similar with .Net but my friend is insisting on C++ saying it is much faster very mature and working with it will not effect on the productivity and development speed. The project that we will work on it will have threading, extensive string and datetime manipulation, some socket programing and of-course work with RDBMS (MySql Or Postgre not decided yet). I have some fears with java since oracle acquired sun and these people will do anything to make money out of it. some have advised in python and ruby and I like python but don't know should I make it the default language in this project. the project is not web application and we will make services and executables. what do you think, if you have other opinion you very welcome. Hint: Mono is not an option

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• Avaliable parallel technologies in .Net

  - by David

  I am new to .Net platform. I did a search and found that there are several ways to do parallel computing in .Net: Parallel task in Task Parallel Library, which is .Net 3.5. PLINQ, .Net 4.0 Asynchounous Programming, .Net 2.0, (async is mainly used to do I/O heavy tasks, F# has a concise syntax supporting this). I list this because in Mono, there seem to be no TPL or PLINQ. Thus if I need to write cross platform parallel programs, I can use async. .Net threads. No version limitation. Could you give some short comments on these or add more methods in this list? Thanks.

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• Ubuntu 12.04 - No sound - HELP!

  - by Bruno Tacca

  I'm panicking... my sound stopped working after I tried to set-up my notebook speakers, plus two headphone jacks... My idea was to multichannel the sound to 3 channels, built-in speakers, and sound-card 2 headphone jacks. After a couple efforts I did it with 2 channels, speakers and 1 headphone jack, but the other wasn't working. After more tries and tries, sound stop working. I just want my sound back... crying like a baby on the floor. And, if possible, but not necessary, a simple guide to active the 3 channels. xD I will post the diagnosis according to https://help.ubuntu.com/community/SoundTroubleshootingProcedure STEP 1 Did it, still no sound. STEP 2 Did it, still no sound. STEP 3 and #STEP 4 (I removed the log cause there is a limit of characters to be posted.) The log can be found here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/238653 STEP 5 Rebooted, still no sound. STEP 6 Did it. In the Output Devices tab, nothing is muted. I play a music with the Rhythmbox Music Player, I don't hear anything but in the pavucontrol I can see in the Built-in Audio Analog Stereo a sound bar shaking... but, no sound. STEP 7 In alsamixer, AlsaMixer v1.0.25 Card: HDA Intel PCH Chip: Creative CA0132 information View: F3:[Playback] F4: Capture F5: All Item: Headphone [dB gain: 25.00, 25.00] Then, I have 5 columns Headphone, Speaker, PCM, S/PDIF, S/PDIF Default PCM A little weird when I try to mute the Headphone and the Speaker, here what happens: Starting both unmutted, mutting headphone cause speaker being mutted automaticaly. Starting both unmutted, mutting speaker cause headphone being mutted automaticaly. Starting both mutted, possible to unmute both separately. STEP 8 I cannot hear sound on both (headphone and/or speaker). STEP 9 Dual boot... Restarted, windows was with sound at max volume. Restarted again, still no sound at ubuntu. I heard something when ubuntu started, a little noise, then silence again. The sound icon always start mutted, after unmutting, I have no sound. STEP 10 I dont have this command in my ubuntu. STEP 11 Tried at STEP 8, no sound. There are no problem with jumpers or hardware, cause I have sound working on windows. STEP 12 No way to open my alienware and loss the warranty x.X" STEP 13 I think it's loaded, judging my the logs STEP 14 Alienware M17xR4, the hardware is listed in the logs above, at STEP 4. There are two headphone hacks, one with just an headphone printed above, and the other with an headset (with mic) printed, there is a mic jack too, and a spdif (optical) too. STEP 15 I dont want to enable S/PDIF STEP 16 I never used the HDMI output, yet... Thanks in advance. I hope I listed all the information you need.

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• How do I install Revenge of the Titans?

  - by Akash

  I've downloaded the .deb file of Revenge of the Titans, and installed it using Ubuntu Software Center. Now, when I try to launch it using the software launcher nothing happens. Any ideas? The .deb file was downloaded from the Humble Indie Bundle. I am unable to launch it from the terminal ( the command revenge-of-the-titans says command not found ). I also tried the .tar.gz. When I extract it and run ./revenge.sh , nothing happens. No output on the terminal or anything at all. I have set chmod 777 revenge.sh as well. The command /opt/revengeofthetitans/revenge.sh does not give any output. If I run gedit /opt/revengeofthetitans/revenge.sh in the terminal: > #!/bin/bash > # > # revenge.sh > # > ############################################################################### > > SCRIPT="`basename $0`" > GAMEDIR="${HOME}/.revenge_of_the_titans_1.80" LOGFILE="${GAMEDIR}/${SCRIPT}.log" > INSTDIR="`dirname $0`" ; cd > "${INSTDIR}" ; INSTDIR="`pwd`" > > [[ ! -d "${GAMEDIR}" ]] && mkdir -m > 0755 "${GAMEDIR}" > > JARPATH="patch.jar:RevengeOfTheTitans.jar:data-hib.jar:gfx.jar:fonts.jar:images.jar:music.jar:fx-mono.jar:fx-stereo.jar:gamecommerce.jar:common.jar:spgl-lite.jar:lwjgl.jar:lwjgl_util.jar:jorbis.jar:jinput.jar" > > # XMODIFIERS is cleared here to prevent SCIM screwing up keyboard > input XMODIFIERS= java \ > -noverify \ > -Djava.library.path="${INSTDIR}" \ > -Dorg.lwjgl.util.NoChecks=true \ > -Dorg.lwjgl.librarypath="${INSTDIR}" \ > -Dnet.puppygames.applet.Launcher.resources=/resources-hib.dat > \ > -Dnet.puppygames.applet.Game.gameResource=game.hib > \ > -XX:MaxGCPauseMillis=3 \ > -Xms64m \ > -Xmx375m \ > -Xincgc \ > -cp "${JARPATH}" \ > net.puppygames.applet.Launcher \ > "$@" \ > >"${LOGFILE}" 2>&1 > > exit 0 > > # > # EOF > # > ###############################################################################

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• Improve Playback Using Enhancements in Windows Media Player 12

  - by DigitalGeekery

  Are you looking for ways to improve the playback of your media in Windows Media Player 12? We’ll show you how to do that by using the enhancements in WMP 12. If you are in Library mode, you’ll need to click the icon at the lower right to switch to Now Playing mode. Right-click anywhere in Media Player while in Now Playing mode, select Enhancements, and select any of the available options.   You can switch between the individual enhancements by clicking the right and left buttons at the top left.   Crossfading and Auto Volume Leveling The Auto Volume Leveling setting is just a simple toggle on and off. If your MP3 or WMA files have volume leveling information values.   You can automatically add volume leveling information values to all files you add to your library by switching to Library view, going to Tools > Options, and selecting Add volume leveling information values for new files on the Library tab. Click OK when finished.   Crossfading will gradually decrease the volume of the song that is ending (fade out) and increase volume of the song that is beginning. Click Turn on Crossfading and then click and drag the slider left or right change the amount of overlap between tracks. Graphic Equalizer The graphic equalizer is toggled on and off by clicking Turn on / Turn off at the top left. You can select pre-defined equalizer settings by music genre by clicking the Default list. The radio buttons on the left allow you to move the sliders individually, in a loose group or a tight group. You can always return to the default settings by clicking Reset. Play Speed Settings Choose a pre-defined settings by clicking Slow, Normal, or Fast. Uncheck the Snap slider to common speeds the move the slider right and left to your desired speed. If nothing else, these settings provide a little fun and amusement. Quiet Mode Quiet mode will level out any sharp volume highs and lows within a single track. Simply toggle the setting on or off and select whether you prefer Medium difference or Little difference by selecting one of the radio buttons. SRS WOW effects SRS WOW effects enhance low-frequency and stereo sound performance. Click Turn on to enable the TruBass and WOW Effect sliders. You can also optimize for your speaker type. Click to switch between Regular, Large, and Headphones. Video Settings Video Settings allow you to adjust the Hue, Brightness, Saturation, and Contrast.   You can also adjust the zoom settings by clicking Select video zoom settings.   Dolby Digital Settings Choose between Normal, Night, and Theater settings to adjust the audio for Dolby Digital content. This setting will only effect media with Dolby Digital sound. Looking for more ways to improve your media experience in WMP 12? Check out how to update metadata and cover art and how to share media with other Windows 7 computers on your home network. Similar Articles Productive Geek Tips Fixing When Windows Media Player Library Won’t Let You Add FilesInstall and Use the VLC Media Player on Ubuntu LinuxHow To Rip a Music CD in Windows 7 Media CenterStream Media from Windows 7 to XP with VLC Media PlayerInstalling Windows Media Player Plugin for Firefox TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Acronis Online Backup DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows Check these Awesome Chrome Add-ons iFixit Offers Gadget Repair Manuals Online Vista style sidebar for Windows 7 Create Nice Charts With These Web Based Tools Track Daily Goals With 42Goals Video Toolbox is a Superb Online Video Editor

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• I got my z-5 Logitech speakers to work, but whenever I restart, I have to reconfigure them

  - by The Bill

  This is the content of my alsa-base.conf file (for some reason, the entries preceded by # are bolded--anyway): autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-usb-audio index=0 options snd-hda-intel index=1 Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 options snd-usb-audio index=-2 options snd-usb-audio index=0 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-hda-intel index=1 I deleted a line that said something like "#Keep usb-audio from being loaded as first soundcard" and that made the speakers work for the first time (before this, they never showed up). I also added the last four lines. Anyway, what can I add to this so that I don't have to reconfigure them each time I restart? Currently, I have to open Sound Settings, then under the hardware tab, select Analog Stereo Output, and then unplug my USB speakers and plug them back in. This makes them pop up so that I can see them. Otherwise, it will not show my Z-5 speakers as a device that can be configured.

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• The Power to Control Power

  - by speakjava

  I'm currently working on a number of projects using embedded Java on the Raspberry Pi and Beagle Board.  These are nice and small, so don't take up much room on my desk as you can see in this picture. As you can also see I have power and network connections emerging from under my desk.  One of the (admittedly very minor) drawbacks of these systems is that they have no on/off switch.  Instead you insert or remove the power connector (USB for the RasPi, a barrel connector for the Beagle).  For the Beagle Board this can potentially be an issue; with the micro-SD card located right next to the connector it has been known for people to eject the card when trying to power off the board, which can be quite serious for the hardware. The alternative is obviously to leave the boards plugged in and then disconnect the power from the outlet.  Simple enough, but a picture of underneath my desk shows that this is not the ideal situation either. This made me think that it would be great if I could have some way of controlling a mains voltage outlet using a remote switch or, even better, from software via a USB connector.  A search revealed not much that fit my requirements, and anything that was close seemed very expensive.  Obviously the only way to solve this was to build my own.Here's my solution.  I decided my system would support both control mechanisms (remote physical switch and USB computer control) and be modular in its design for optimum flexibility.  I did a bit of searching and found a company in Hong Kong that were offering solid state relays for 99p plus shipping (£2.99, but still made the total price very reasonable).  These would handle up to 380V AC on the output side so more than capable of coping with the UK 240V supply.  The other great thing was that being solid state, the input would work with a range of 3-32V and required a very low current of 7.5mA at 12V.  For the USB control an Arduino board seemed the obvious low-cost and simple choice.  Given the current requirments of the relay, the Arduino would not require the additional power supply and could be powered just from the USB.Having secured the relays I popped down to Homebase for a couple of 13A sockets, RS for a box and an Arduino and Maplin for a toggle switch.  The circuit is pretty straightforward, as shown in the diagram (only one output is shown to make it as simple as possible).  Originally I used a 2 pole toggle switch to select the remote switch or USB control by switching the negative connections of the low voltage side.  Unfortunately, the resistance between the digital pins of the Arduino board was not high enough, so when using one of the remote switches it would turn on both of the outlets.  I changed to a 4 pole switch and isolated both positive and negative connections. IMPORTANT NOTE: If you want to follow my design, please be aware that it requires working with mains voltages.  If you are at all concerned with your ability to do this please consult a qualified electrician to help you.It was a tight fit, especially getting the Arduino in, but in the end it all worked.  The completed box is shown in the photos. The remote switch was pretty simple just requiring the squeezing of two rocker switches and a 9V battery into the small RS supplied box.  I repurposed a standard stereo cable with phono plugs to connect the switch box to the mains outlets.  I chopped off one set of plugs and wired it to the rocker switches.  The photo shows the RasPi and the Beagle board now controllable from the switch box on the desk. I've tested the Arduino side of things and this works fine.  Next I need to write some software to provide an interface for control of the outlets.  I'm thinking a JavaFX GUI would be in keeping with the total overkill style of this project.

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• No external microphone Acer AO722

  - by Leeghwater

  The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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• AS3 microphone recording/saving works, in-flash PCM playback double speed

  - by Lowgain

  I have a working mic recording script in AS3 which I have been able to successfully use to save .wav files to a server through AMF. These files playback fine in any audio player with no weird effects. For reference, here is what I am doing to capture the mic's ByteArray: (within a class called AudioRecorder) public function startRecording():void { _rawData = new ByteArray(); _microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, _samplesCaptured, false, 0, true); } private function _samplesCaptured(e:SampleDataEvent):void { _rawData.writeBytes(e.data); } This works with no problems. After the recording is complete I can take the _rawData variable and run it through a WavWriter class, etc. However, if I run this same ByteArray as a sound using the following code which I adapted from the adobe cookbook: (within a class called WavPlayer) public function playSound(data:ByteArray):void { _wavData = data; _wavData.position = 0; _sound.addEventListener(SampleDataEvent.SAMPLE_DATA, _playSoundHandler); _channel = _sound.play(); _channel.addEventListener(Event.SOUND_COMPLETE, _onPlaybackComplete, false, 0, true); } private function _playSoundHandler(e:SampleDataEvent):void { if(_wavData.bytesAvailable <= 0) return; for(var i:int = 0; i < 8192; i++) { var sample:Number = 0; if(_wavData.bytesAvailable > 0) sample = _wavData.readFloat(); e.data.writeFloat(sample); } } The audio file plays at double speed! I checked recording bitrates and such and am pretty sure those are all correct, and I tried changing the buffer size and whatever other numbers I could think of. Could it be a mono vs stereo thing? Hope I was clear enough here, thanks!

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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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• Best way to handle huge strings in c#

  - by srk

  I have to write the data below to a textfile after replacing two values with ##IP##, ##PORT##. what is the best way ? should i hold all in a string and use Replace and write to textfile ? Data : [APP] iVersion= 101 pcVersion=1.01a pcBuildDate=Mar 27 2009 [MAIN] iFirstSetup= 0 rcMain.rcLeft= 676 rcMain.rcTop= 378 rcMain.rcRight= 1004 rcMain.rcBottom= 672 iShowLog= 0 iMode= 1 [GENERAL] iTips= 1 iTrayAnimation= 1 iCheckColor= 1 iPriority= 1 iSsememcpy= 1 iAutoOpenRecv= 1 pcRecvPath=C:\Documents and Settings\karthikeyan\My Documents\Downloads\fremote101a\FantasyRemote101a\recv pcFileName=FantasyRemote iLanguage= 1 [SERVER] iAcceptVideo= 1 iAcceptAudio= 1 iAcceptInput= 1 iAutoAccept= 1 iAutoTray= 0 iConnectSound= 1 iEnablePassword= 0 pcPassword= pcPort=7902 [CLIENT] iAutoConnect= 0 pcPassword= pcDefaultPort=7902 [NETWORK] pcConnectAddr=##IP## pcPort=##Port## [VIDEO] iEnable= 1 pcFcc=AMV3 pcFccServer= pcDiscription= pcDiscriptionServer= iFps= 30 iMouse= 2 iHalfsize= 0 iCapturblt= 0 iShared= 0 iSharedTime= 5 iVsync= 1 iCodecSendState= 1 iCompress= 2 pcPlugin= iPluginScan= 0 iPluginAspectW= 16 iPluginAspectH= 9 iPluginMouse= 1 iActiveClient= 0 iDesktop1= 1 iDesktop2= 2 iDesktop3= 0 iDesktop4= 3 iScan= 1 iFixW= 16 iFixH= 8 [AUDIO] iEnable= 1 iFps= 30 iVolume= 6 iRecDevice= 0 iPlayDevice= 0 pcSamplesPerSec=44100Hz pcChannels=2ch:Stereo pcBitsPerSample=16bit iRecBuffNum= 150 iPlayBuffNum= 4 [INPUT] iEnable= 1 iFps= 30 iMoe= 0 iAtlTab= 1 [MENU] iAlwaysOnTop= 0 iWindowMode= 0 iFrameSize= 4 iSnap= 1 [HOTKEY] iEnable= 1 key_IDM_HELP=0x00000070 mod_IDM_HELP=0x00000000 key_IDM_ALWAYSONTOP=0x00000071 mod_IDM_ALWAYSONTOP=0x00000000 key_IDM_CONNECT=0x00000072 mod_IDM_CONNECT=0x00000000 key_IDM_DISCONNECT=0x00000073 mod_IDM_DISCONNECT=0x00000000 key_IDM_CONFIG=0x00000000 mod_IDM_CONFIG=0x00000000 key_IDM_CODEC_SELECT=0x00000000 mod_IDM_CODEC_SELECT=0x00000000 key_IDM_CODEC_CONFIG=0x00000000 mod_IDM_CODEC_CONFIG=0x00000000 key_IDM_SIZE_50=0x00000074 mod_IDM_SIZE_50=0x00000000 key_IDM_SIZE_100=0x00000075 mod_IDM_SIZE_100=0x00000000 key_IDM_SIZE_200=0x00000076 mod_IDM_SIZE_200=0x00000000 key_IDM_SIZE_300=0x00000000 mod_IDM_SIZE_300=0x00000000 key_IDM_SIZE_400=0x00000000 mod_IDM_SIZE_400=0x00000000 key_IDM_CAPTUREWINDOW=0x00000077 mod_IDM_CAPTUREWINDOW=0x00000004 key_IDM_REGION=0x00000077 mod_IDM_REGION=0x00000000 key_IDM_DESKTOP1=0x00000078 mod_IDM_DESKTOP1=0x00000000 key_IDM_ACTIVE_MENU=0x00000079 mod_IDM_ACTIVE_MENU=0x00000000 key_IDM_PLUGIN=0x0000007A mod_IDM_PLUGIN=0x00000000 key_IDM_PLUGIN_SCAN=0x00000000 mod_IDM_PLUGIN_SCAN=0x00000000 key_IDM_DESKTOP2=0x00000078 mod_IDM_DESKTOP2=0x00000004 key_IDM_DESKTOP3=0x00000079 mod_IDM_DESKTOP3=0x00000004 key_IDM_DESKTOP4=0x0000007A mod_IDM_DESKTOP4=0x00000004 key_IDM_WINDOW_NORMAL=0x0000000D mod_IDM_WINDOW_NORMAL=0x00000004 key_IDM_WINDOW_NOFRAME=0x0000000D mod_IDM_WINDOW_NOFRAME=0x00000002 key_IDM_WINDOW_FULLSCREEN=0x0000000D mod_IDM_WINDOW_FULLSCREEN=0x00000001 key_IDM_MINIMIZE=0x00000000 mod_IDM_MINIMIZE=0x00000000 key_IDM_MAXIMIZE=0x00000000 mod_IDM_MAXIMIZE=0x00000000 key_IDM_REC_START=0x00000000 mod_IDM_REC_START=0x00000000 key_IDM_REC_STOP=0x00000000 mod_IDM_REC_STOP=0x00000000 key_IDM_SCREENSHOT=0x0000002C mod_IDM_SCREENSHOT=0x00000002 key_IDM_AUDIO_MUTE=0x00000073 mod_IDM_AUDIO_MUTE=0x00000004 key_IDM_AUDIO_VOLUME_DOWN=0x00000074 mod_IDM_AUDIO_VOLUME_DOWN=0x00000004 key_IDM_AUDIO_VOLUME_UP=0x00000075 mod_IDM_AUDIO_VOLUME_UP=0x00000004 key_IDM_CTRLALTDEL=0x00000023 mod_IDM_CTRLALTDEL=0x00000003 key_IDM_QUIT=0x00000000 mod_IDM_QUIT=0x00000000 key_IDM_MENU=0x0000007B mod_IDM_MENU=0x00000000 [OVERLAY] iIndicator= 1 iAlphaBlt= 1 iEnterHide= 0 pcFont=MS UI Gothic [AVI] iSound= 1 iFileSizeLimit= 100000 iPool= 4 iBuffSize= 32 iStartDiskSpaceCheck= 1 iStartDiskSpace= 1000 iRecDiskSpaceCheck= 1 iRecDiskSpace= 100 iCache= 0 iAutoOpen= 1 pcPath=C:\Documents and Settings\karthikeyan\My Documents\Downloads\fremote101a\FantasyRemote101a\avi [SCREENSHOT] iSound= 1 iAutoOpen= 1 pcPath=C:\Documents and Settings\karthikeyan\My Documents\Downloads\fremote101a\FantasyRemote101a\ss pcPlugin=BMP [CDLG_SERVER] mrcWnd.rcLeft= 667 mrcWnd.rcTop= 415 mrcWnd.rcRight= 1013 mrcWnd.rcBottom= 634 [CWND_CLIENT] miShowLog= 0 m_iOverlayLock= 0 [CDLG_CONFIG] mrcWnd.rcLeft= 467 mrcWnd.rcTop= 247 mrcWnd.rcRight= 1213 mrcWnd.rcBottom= 802 miTabConfigSel= 2

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• JMF microphone volume controller

  - by TacB0sS

  How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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• Rapid calls to fread crashes the application

  - by Slynk

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

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• How accurately (in terms of time) does Windows play audio?

  - by MusiGenesis

  Let's say I play a stereo WAV file with 317,520,000 samples, which is theoretically 1 hour long. Assuming no interruptions of the playback, will the file finish playing in exactly one hour, or is there some occasional tiny variation in the playback speed such that it would be slightly more or slightly less (by some number of milliseconds) than one hour? I am trying to synchronize animation with audio, and I am using a System.Diagnostics.Stopwatch to keep the frames matching the audio. But if the playback speed of WAV audio in Windows can vary slightly over time, then the audio will drift out of sync with the Stopwatch-driven animation. Which leads to a second question: it appears that a Stopwatch - while highly granular and accurate for short durations - runs slightly fast. On my laptop, a Stopwatch run for exactly 24 hours (as measured by the computer's system time and a real stopwatch) shows an elapsed time of 24 hours plus about 5 seconds (not milliseconds). Is this a known problem with Stopwatch? (A related question would be "am I crazy?", but you can try it for yourself.) Given its usage as a diagnostics tool, I can see where a discrepancy like this would only show up when measuring long durations, for which most people would use something other than a Stopwatch. If I'm really lucky, then both Stopwatch and audio playback are driven by the same underlying mechanism, and thus will stay in sync with each other for days on end. Any chance this is true?

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