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  • How do I record audio through M-Audio Keystudio?

    - by interstar
    Hi, I'm trying to get my M-Audio Keystudio (which has an audio input as well as the keyboard) to record audio to Audacity. I'm in Ubuntu 10.10. When I look at the Sound Preferences I can select "M-Audio RunTime DFU Analog Stereo" as my input device. However, when I try to record in Audacity, Audacity remains frozen. The program seems to be running and recording, but the recording cursor won't advance. If I reset the audio input to the internal sound card, recording works normally. Any ideas what to look for?

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  • Unity 5.1 audio issues (no sound in back channels)

    - by N0xus
    I've trying to bring in surround sound audio into my project. I've set my computer up to run in 5.1 and when I play a 6 channel audio through windows media player (it's a test audio that does left speaker, right speaker etc) it works fine. However, when I run it through Unity, all I get is the front 3 channels. I've set it in the Edit - project settings - audio to be 5.1 in there. I even set it in code with following: void Start() { AudioSettings.speakerMode = AudioSpeakerMode.Mode5point1; } How ever, when I run a debug line of: print ( AudioSettings.driverCaps); It tells me that Unity is only playing in stereo. Is there something I'm still not doing? I should also add I've ran 10 different tests using the 3D audio pan and spread options. I've set both to either being fully off, half way on and full. Still the same results.

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  • Stream audio to mobile device

    - by blackn1ght
    I'd like to stream the audio from Ubuntu 10.10 to my HTC Desire HD (Android 2.2). I've seen solutions so far for streaming from audio players, but I'd like to stream any audio output from the PC to my phone. My use case is for watching TV/Films in VLC or online (BBC iPlayer) in bed, without having to use my surround sound system which is likely to wake up my house mates. I'm not just talking about music from Banshee, but any audio that the system makes. I was thinking that PulseAudio is pretty powerful, is it possible to route audio through that to a mobile device? Can it be done through bluetooth? Cheers in advance!

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  • USB Audio Device Loopback Through Speakers

    - by matto1990
    I have a USB turntable which when plugged in to my ubuntu 10.10 machine appears in the audio settings as an input device (USB PnP Audio Device Analog Stereo) like a microphone. What I'd like to be able to do it to have the sound for that audio device played back through the audio output (speaker or whatever). I'm not too worried if there's a slight delay between the audio coming in and it being played out through the speakers. As far as I'm aware this is refereed to as software loopback. I can achieve exactly what I want if I open Audacity, enable software loopback and press record. Obvious this isn't ideal as I don't really want it recording what I'm playing all the time. I know this is possible because of the Audacity example however I'd like to know if there's a way to do it without it recording. I've search around for a while for a piece of software that does this, however I couldn't get anything even close. Any help would be greatly appreciated.

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  • Play audio in javascript with a good performance

    - by João
    I'm developing a browser game where the player can shoot. Everytime he shoots it play a sound. Currently i'm using this code to play sounds in JavaScript: var audio = document.createElement("audio"); audio.src = "my_sound.mp3"; audio.play(); I'm worried about performance here. Will 10 simultaneous sounds impact my game performance too much? Will all audio objects stay in memory even after they are played?

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  • How can I select an audio output device in directshow

    - by Vibhore Tanwer
    I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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  • Do you have any additions or alterations to this list of popular audio formats?

    - by roja
    All, I am trying to compile a list of common audio file formats used in both personal storage and peer transmission. I have compiled the following list, do you think that there are any significant formats missing? Are any of them not actually common formats? Any advice/alterations are highly useful. advanced audio coding, apple lossless audio file, atrac3 audio file, atrac audio file, audio interchange file format, core audio file, free lossless audio codec file, mpeg 1 audio layer 3, mpeg 2 audio, mpeg 4 audio book file, musical instrument digital interface, ogg vorbis compressed audio file, open media framework file, real audio, real audio media, waveform audio file format, windows media audio Kind regards, Roja

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  • Boost Audio Input on OS X?

    - by alanstorm
    I'm using my 13" Mac Book Pro's audio input functionality with an external microphone (recent vintage, bought around Thanksgiving). I've increased my input volume to the maximum in system preference, but the resulting recorded volume (using iShowU HD) is very low. Is there anyway to increase the input volume/sensitivity beyond Apple's default settings? I've found plenty on google about increasing the OUTPUT volume, but I want to increase the input volume.

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  • Set default system audio output port (for all accounts)

    - by Ludwik Trammer
    The default output audio port Ubuntu doesn't work on my system. It should be "Analog Mono Output/Amplifier", instead of "Analog Output/Amplifier". I can easily change that in sound preferences, just by choosing the right port in the "Output" tab. The problem is this would only apply to a single account, and I would like to change it system-wide, so it applies to all accounts on the system (I have more than 100 users...). I'm after 2 hours of Googling, so any help would be appreciated.

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  • How much system and business analysis should a programmer be reasonably expected to do?

    - by Rahul
    In most places I have worked for, there were no formal System or Business Analysts and the programmers were expected to perform both the roles. One had to understand all the subsystems and their interdependencies inside out. Further, one was also supposed to have a thorough knowledge of the business logic of the applications and interact directly with the users to gather requirements, answer their queries etc. In my current job, for ex, I spend about 70% time doing system analysis and only 30% time programming. I consider myself a good programmer but struggle with developing a good understanding of the business rules of a complex application. Often, this creates a handicap because while I can write efficient algorithms and thread-safe code, I lose out to guys who may be average programmers but have a much better understanding of the business processes. So I want to know - How much business and systems knowledge should a programmer have ? - How does one go about getting this knowledge in an immensely complex software system (e.g. trading applications) with several interdependent business processes but poorly documented business rules.

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  • JavaScript audio not playing outside of jQuery function

    - by user1814016
    I know the question title doesn't make much sense, but I can't think of a better way to put it. I am a newbie to jQuery and I'm using this code to fade in a <div> and play a sound: $(document).ready(function(){ $('#speech').fadeIn('medium', function() { play('msg_appear'); var sptx = $('<p class="stext">').text('There is nothing here.'); $('#speech').append(sptx); $('.stext').typeOut({marker: '', delay: 22}); }); }); This code runs fine however the sound plays after the fade-in is complete. I wanted it to play while it was fading in, so I tried placing the play() call outside of the fade-in function like this: $(document).ready(function(){ play('msg_appear'); $('#speech').fadeIn('medium', function() { However, now it's not playing at all. There's no errors on the JavaScript console so I'm unsure if it's a syntax error, and probably something obvious, but I don't know what. play() is a function I found to play audio, here it is if it matters at all. I placed it in the same file the above code is; right above the $(document).ready(). function play(sound) { if (window.HTMLAudioElement) { var snd = new Audio(''); if(snd.canPlayType('audio/ogg')) { snd = new Audio(sound + '.ogg'); } else if(snd.canPlayType('audio/mp3')) { snd = new Audio(sound + '.mp3'); } snd.play(); } else { alert('HTML5 Audio is not supported by your browser!'); } }

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  • Analysis Services Tabular books #ssas #tabular

    - by Marco Russo (SQLBI)
    Many people are looking for books about Analysis Services Tabular. Today there are two books available and they complement each other: Microsoft SQL Server 2012 Analysis Services: The BISM Tabular Model by Marco Russo, Alberto Ferrari and Chris Webb Applied Microsoft SQL Server 2012 Analysis Services: Tabular Modeling by Teo Lachev The book I wrote with Alberto and Chris is a complete guide to create tabular models and has a good coverage about DAX, including how to use it for enriching a semantic model with calculated columns and measures and how to use it for querying a Tabular model. In my experience, DAX as a query language is a very interesting option for custom analytical applications that requires a fast calculation engine, or simply for standard reports running in Reporting Services and accessing a Tabular model. You can freely preview the table of content and read some excerpts from the book on Safari Books Online. The book is in printing and should be shipped within mid-July, so finally it will be very soon on the shelf of all the people already preordered it! The Teo Lachev’s book, covers the full spectrum of Tabular models provided by Microsoft: starting with self-service BI, you have users creating a model with PowerPivot for Excel, publishing it to PowerPivot for SharePoint and exploring data by using Power View; then, the PowerPivot for Excel model can be imported in a Tabular model and published in Analysis Services, adding more control on the model through row-level security and partitioning, for example. Teo’s book follows a step-by-step approach describing each feature that is very good for a beginner that is new to PowerPivot and/or to BISM Tabular. If you need to get the big picture and to start using the products that are part of the new Microsoft wave of BI products, the Teo’s book is for you. After you read the book from Teo, or if you already have a certain confidence with PowerPivot or BISM Tabular and you want to go deeper about internals, best practices, design patterns in just BISM Tabular, then our book is a suggested read: it contains several chapters about DAX, includes discussions about new opportunities in data model design offered by Tabular models, and also provides examples of optimizations you can obtain in DAX and best practices in data modeling and queries. It might seem strange that an author write a review of a book that might seem to compete with his one, but in reality these two books complement each other and are not alternatives. If you have any doubt, buy both: you will be not disappointed! Moreover, Amazon usually offers you a deal to buy three books, including the Visualizing Data with Microsoft Power View, another good choice for getting all the details about Power View.

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  • Monitoring an audio line.

    - by Stefan Liebenberg
    I need to monitor my audio line-in in linux, and in the event that audio is played, the sound must be recorded and saved to a file. Similiar to how motion monitors the video feed. Is it possible to do this with bash? something along the lines of: #!/bin/bash # audio device device=/dev/audio-line-in # below this threshold audio will not be recorded. noise_threshold=10 # folder where recordings are stored storage_folder=~/recordings # run indefenitly, until Ctrl-C is pressed while true; do # noise_level() represents a function to determine # the noise level from device if noise_level( $device ) > $noise_threshold; then # stream from device to file, can be encoded to mp3 later. cat $device > $storage_folder/`date`.raw fi; done;

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  • PhP/HTML play button [migrated]

    - by Marian
    I'm wanting to make my own small webpage, I've got a domain Saoo.eu As you see there is a small play button in the corner witch plays a playlist. Is there anyway to have that playbutton on each page I'd add in the future without resetting every time the page changes? Am I forced to use iFrames for that? This is my player code <button id="audioControl" style="width:30px;height:25px;"></button> <audio id="aud" src="" autoplay autobuffer /> Script: $(document).ready(function() { $('#audioControl').html('II'); if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play0.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play0.ogg'); } }); var audio = document.getElementById('aud'), count = 0; $('#audioControl').toggle( function () { audio.pause(); $('#audioControl').html('>'); }, function () { audio.play(); $('#audioControl').html('II'); } ); audio.addEventListener("ended", function() { count++; if(count == 4){count = 0;} if(Modernizr.audio && Modernizr.audio.mp3) { audio.setAttribute("src",'http://daokun.webs.com/play'+count+'.mp3'); } else if(Modernizr.audio && Modernizr.audio.wav) { audio.setAttribute("src", 'http://daokun.webs.com/play'+count+'.ogg'); } audio.load(); });

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  • Web Audio API and mobile browsers

    - by Michael
    I've run into a problem while implementing sound and music into an HTML game that I'm building. I'm using the Web Audio API, loading all the sound files with XMLHttpRequests and decoding them into an AudioBufferSourceNode with AudioContext.prototype.decodeAudioData(). It looks something like this: var request = new XMLHttpRequest(); request.open("GET", "soundfile.ogg", true); request.responseType = "arraybuffer"; request.onload = function() { context.decodeAudioData(request.response) } request.send(); Everything plays fine, but on mobile the decodeAudioData takes an absurdly long time for the background music. I then tried using AudioContext.prototype.createMediaElementSource() to load the music from an HTML Audio object, since they support streaming and don't have to load the whole file into memory at once. It looked something like this: var audio = new Audio('soundfile.ogg'); var source = context.createMediaElementSource(audio); var mainVolume = context.createGain(); source.connect(mainVolume); mainVolume.connect(context.destination); This loads much faster, but the audio volume isn't affected by the gain node. Works fine on desktop, so I'm assuming this is a bug/limitation of mobile Chrome (testing on Android). Is there actually no good, well-performing way to handle sound on mobile browsers or am just I doing something stupid?

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  • Audio programming resources

    - by rashleighp
    I've been very interested in the last few months about getting in to audio programming (I'm from a musical background). I've been a .NET developer for two years and have also done some objective c for an iPhone app recently. I realise I would probably need to work on my C++ chops and have been having a play around with FMOD EX and doing a lot of research into the industry. I was just wondering if anyone could suggest some good resources for audio programming (be they websites, podcasts, books, videos, online courses etc). Anything from Fourier analysis, low level coding, audio engine creation to audio APIs. I just want to learn as much as possible! Thanks in advance.

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  • Setting Up Audio on a Server Install

    - by tdcrenshaw
    I'm running on a clean install of 10.10 Server edition and have alsa-base, alsa-tools, alsa-utils, alsaplayer, and alsa-firmware-loader installed. At one point I installed pulseaudio, but I have since removed it. I've tried the following lspci | grep audio 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 01) 01:06.0 Multimedia audio controller: Creative Labs [SB Live! Value] EMU10k1X aplay -l aplay: device_list:235: no soundcards found... alsamixer can not open mixer: No such file or directory When I search for modules with find /lib/modules/`uname -r` | grep snd I do get a list of modules I'm not very experienced with alsa setup, so I'm not sure where to go from here

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  • How to stream semi-live audio over internet

    - by Thomas Tempelmann
    I want to write something like Skype, i.e. I have a constant audio stream on one computer and then recompress it in a format that's suitable for a latent internet connection, receive it on the other end and play it. Let's also assume that the internet connection is fairly modern and fast, i.e. DSL or alike, no slow connections over phone and such. The involved computers will also be rather modern (Dual Core Intel CPUs at 2GHz or more). I know how to handle the audio on the machines. What I don't know is how to transmit the audio in an efficient way. The challenges are: I'd like get good audio quality across the line. The stream should be received without drops. The stream may, however, be received with a little delay (a second delay is acceptable). I imagine that the transport software could first determine the average (and max) latency, then start the stream and tell the receiver to wait for that max latency before starting to play the audio. With that, if the latency doesn't get any higher, the entire stream will be playable on the other side without stutter or drops. If, due to unexpected IP latencies or blockages, the stream does get cut off, I want to be able to notice this so that I can take actions (e.g. abort the stream) and eventually start a new transmission. What are my options if I want do use ready-made software for the compression and tranmission? I have no intention to write my own audio compression engine, really. OTOH, I plan to sell the solution in a vertical market, meaning I can afford a few dollars of license fees per copy, but not $100s. I guess the simplest solution would be to just open a TCP stream, send a few packets back and forth to determine their running time (or even use UDP for that), then use the results as the guide for my max latency value, then simply fire the audio data in its raw form (uncompressed 16 bit stereo), along with a timing code over the TCP connection. The receiver reads the data and plays it with the pre-determined delay. That might just work with the type of fast connection I expect. I just wonder if there are better solutions to reach this goal, with better performance (lower latency) and less data (compressed). BTW, I first try to implement this on OS X, but might want to do it on Windows, too, if it proves successful.

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  • Record audio via MediaRecorder

    - by Isuru Madusanka
    I am trying to record audio by MediaRecorder, and I get an error, I tried to change everything and nothing works. Last two hours I try to find the error, I used Log class too and I found out that error occurred when it call recorder.start() method. What could be the problem? public class AudioRecorderActivity extends Activity { MediaRecorder recorder; File audioFile = null; private static final String TAG = "AudioRecorderActivity"; private View startButton; private View stopButton; /** Called when the activity is first created. */ @Override public void onCreate(Bundle savedInstanceState) { super.onCreate(savedInstanceState); startButton = findViewById(R.id.start); stopButton = findViewById(R.id.stop); setContentView(R.layout.main); } public void startRecording(View view) throws IOException{ startButton.setEnabled(false); stopButton.setEnabled(true); File sampleDir = Environment.getExternalStorageDirectory(); try{ audioFile = File.createTempFile("sound", ".3gp", sampleDir); }catch(IOException e){ Toast.makeText(getApplicationContext(), "SD Card Access Error", Toast.LENGTH_LONG).show(); Log.e(TAG, "Sdcard access error"); return; } recorder = new MediaRecorder(); recorder.setAudioSource(MediaRecorder.AudioSource.MIC); recorder.setOutputFormat(MediaRecorder.OutputFormat.THREE_GPP); recorder.setAudioEncoder(MediaRecorder.AudioEncoder.AMR_NB); recorder.setAudioEncodingBitRate(16); recorder.setAudioSamplingRate(44100); recorder.setOutputFile(audioFile.getAbsolutePath()); recorder.prepare(); recorder.start(); } public void stopRecording(View view){ startButton.setEnabled(true); stopButton.setEnabled(false); recorder.stop(); recorder.release(); addRecordingToMediaLibrary(); } protected void addRecordingToMediaLibrary(){ ContentValues values = new ContentValues(4); long current = System.currentTimeMillis(); values.put(MediaStore.Audio.Media.TITLE, "audio" + audioFile.getName()); values.put(MediaStore.Audio.Media.DATE_ADDED, (int)(current/1000)); values.put(MediaStore.Audio.Media.MIME_TYPE, "audio/3gpp"); values.put(MediaStore.Audio.Media.DATA, audioFile.getAbsolutePath()); ContentResolver contentResolver = getContentResolver(); Uri base = MediaStore.Audio.Media.EXTERNAL_CONTENT_URI; Uri newUri = contentResolver.insert(base, values); sendBroadcast(new Intent(Intent.ACTION_MEDIA_SCANNER_SCAN_FILE, newUri)); Toast.makeText(this, "Added File" + newUri, Toast.LENGTH_LONG).show(); } } And here is the xml layout. <?xml version="1.0" encoding="utf-8"?> <RelativeLayout xmlns:android="http://schemas.android.com/apk/res/android" android:id="@+id/RelativeLayout1" android:layout_width="fill_parent" android:layout_height="fill_parent" android:orientation="vertical" > <Button android:id="@+id/start" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignParentTop="true" android:layout_centerHorizontal="true" android:layout_marginTop="146dp" android:onClick="startRecording" android:text="Start Recording" /> <Button android:id="@+id/stop" android:layout_width="wrap_content" android:layout_height="wrap_content" android:layout_alignLeft="@+id/start" android:layout_below="@+id/start" android:layout_marginTop="41dp" android:enabled="false" android:onClick="stopRecording" android:text="Stop Recording" /> </RelativeLayout> And I added permission to AndroidManifest file. <?xml version="1.0" encoding="utf-8"?> <manifest xmlns:android="http://schemas.android.com/apk/res/android" package="in.isuru.audiorecorder" android:versionCode="1" android:versionName="1.0" > <uses-sdk android:minSdkVersion="8" /> <application android:icon="@drawable/ic_launcher" android:label="@string/app_name" > <activity android:name=".AudioRecorderActivity" android:label="@string/app_name" > <intent-filter> <action android:name="android.intent.action.MAIN" /> <category android:name="android.intent.category.LAUNCHER" /> </intent-filter> </activity> </application> <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/> <uses-permission android:name="android.permission.RECORD_AUDIO" /> </manifest> I need to record high quality audio. Thanks!

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  • Using the Static Code Analysis feature of Visual Studio (Premium/Ultimate) to find memory leakage problems

    - by terje
    Memory for managed code is handled by the garbage collector, but if you use any kind of unmanaged code, like native resources of any kind, open files, streams and window handles, your application may leak memory if these are not properly handled.  To handle such resources the classes that own these in your application should implement the IDisposable interface, and preferably implement it according to the pattern described for that interface. When you suspect a memory leak, the immediate impulse would be to start up a memory profiler and start digging into that.   However, before you follow that impulse, do a Static Code Analysis run with a ruleset tuned to finding possible memory leaks in your code.  If you get any warnings from this, fix them before you go on with the profiling. How to use a ruleset In Visual Studio 2010 (Premium and Ultimate editions) you can define your own rulesets containing a list of Static Code Analysis checks.   I have defined the memory checks as shown in the lists below as ruleset files, which can be downloaded – see bottom of this post.  When you get them, you can easily attach them to every project in your solution using the Solution Properties dialog. Right click the solution, and choose Properties at the bottom, or use the Analyze menu and choose “Configure Code Analysis for Solution”: In this dialog you can now choose the Memorycheck ruleset for every project you want to investigate.  Pressing Apply or Ok opens every project file and changes the projects code analysis ruleset to the one we have specified here. How to define your own ruleset  (skip this if you just download my predefined rulesets) If you want to define the ruleset yourself, open the properties on any project, choose Code Analysis tab near the bottom, choose any ruleset in the drop box and press Open Clear out all the rules by selecting “Source Rule Sets” in the Group By box, and unselect the box Change the Group By box to ID, and select the checks you want to include from the lists below. Note that you can change the action for each check to either warning, error or none, none being the same as unchecking the check.   Now go to the properties window and set a new name and description for your ruleset. Then save (File/Save as) the ruleset using the new name as its name, and use it for your projects as detailed above. It can also be wise to add the ruleset to your solution as a solution item. That way it’s there if you want to enable Code Analysis in some of your TFS builds.   Running the code analysis In Visual Studio 2010 you can either do your code analysis project by project using the context menu in the solution explorer and choose “Run Code Analysis”, you can define a new solution configuration, call it for example Debug (Code Analysis), in for each project here enable the Enable Code Analysis on Build   In Visual Studio Dev-11 it is all much simpler, just go to the Solution root in the Solution explorer, right click and choose “Run code analysis on solution”.     The ruleset checks The following list is the essential and critical memory checks.  CheckID Message Can be ignored ? Link to description with fix suggestions CA1001 Types that own disposable fields should be disposable No  http://msdn.microsoft.com/en-us/library/ms182172.aspx CA1049 Types that own native resources should be disposable Only if the pointers assumed to point to unmanaged resources point to something else  http://msdn.microsoft.com/en-us/library/ms182173.aspx CA1063 Implement IDisposable correctly No  http://msdn.microsoft.com/en-us/library/ms244737.aspx CA2000 Dispose objects before losing scope No  http://msdn.microsoft.com/en-us/library/ms182289.aspx CA2115 1 Call GC.KeepAlive when using native resources See description  http://msdn.microsoft.com/en-us/library/ms182300.aspx CA2213 Disposable fields should be disposed If you are not responsible for release, of if Dispose occurs at deeper level  http://msdn.microsoft.com/en-us/library/ms182328.aspx CA2215 Dispose methods should call base class dispose Only if call to base happens at deeper calling level  http://msdn.microsoft.com/en-us/library/ms182330.aspx CA2216 Disposable types should declare a finalizer Only if type does not implement IDisposable for the purpose of releasing unmanaged resources  http://msdn.microsoft.com/en-us/library/ms182329.aspx CA2220 Finalizers should call base class finalizers No  http://msdn.microsoft.com/en-us/library/ms182341.aspx Notes: 1) Does not result in memory leak, but may cause the application to crash   The list below is a set of optional checks that may be enabled for your ruleset, because the issues these points too often happen as a result of attempting to fix up the warnings from the first set.   ID Message Type of fault Can be ignored ? Link to description with fix suggestions CA1060 Move P/invokes to NativeMethods class Security No http://msdn.microsoft.com/en-us/library/ms182161.aspx CA1816 Call GC.SuppressFinalize correctly Performance Sometimes, see description http://msdn.microsoft.com/en-us/library/ms182269.aspx CA1821 Remove empty finalizers Performance No http://msdn.microsoft.com/en-us/library/bb264476.aspx CA2004 Remove calls to GC.KeepAlive Performance and maintainability Only if not technically correct to convert to SafeHandle http://msdn.microsoft.com/en-us/library/ms182293.aspx CA2006 Use SafeHandle to encapsulate native resources Security No http://msdn.microsoft.com/en-us/library/ms182294.aspx CA2202 Do not dispose of objects multiple times Exception (System.ObjectDisposedException) No http://msdn.microsoft.com/en-us/library/ms182334.aspx CA2205 Use managed equivalents of Win32 API Maintainability and complexity Only if the replace doesn’t provide needed functionality http://msdn.microsoft.com/en-us/library/ms182365.aspx CA2221 Finalizers should be protected Incorrect implementation, only possible in MSIL coding No http://msdn.microsoft.com/en-us/library/ms182340.aspx   Downloadable ruleset definitions I have defined three rulesets, one called Inmeta.Memorycheck with the rules in the first list above, and Inmeta.Memorycheck.Optionals containing the rules in the second list, and the last one called Inmeta.Memorycheck.All containing the sum of the two first ones.  All three rulesets can be found in the  zip archive  “Inmeta.Memorycheck” downloadable from here.   Links to some other resources relevant to Static Code Analysis MSDN Magazine Article by Mickey Gousset on Static Code Analysis in VS2010 MSDN :  Analyzing Managed Code Quality by Using Code Analysis, root of the documentation for this Preventing generated code from being analyzed using attributes Online training course on Using Code Analysis with VS2010 Blogpost by Tatham Oddie on custom code analysis rules How to write custom rules, from Microsoft Code Analysis Team Blog Microsoft Code Analysis Team Blog

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  • Books or help on OO Analysis

    - by Pat
    I have this course where we learn about the domain model, use cases, contracts and eventually leap into class diagrams and sequence diagrams to define good software classes. I just had an exam and I got trashed, but part of the reason is we barely have any practical material, I spent at least two good months without drawing a single class diagram by myself from a case study. I'm not here to blame the system or the class I'm in, I'm just wondering if people have some exercise-style books that either provide domain models with glossaries, system sequence diagrams and ask you to use GRASP to make software classes? I could really use some alone-time practicing going from analysis to conception of software entities. I'm almost done with Larman's book called "Applying UML and Patterns An Introduction to Object-Oriented Analysis and Design and Iterative Development, Third Edition". It's a good book, but I'm not doing anything by myself since it doesn't come with exercises. Thanks.

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  • "continue" and "break" for static analysis

    - by B. VB.
    I know there have been a number of discussions of whether break and continue should be considered harmful generally (with the bottom line being - more or less - that it depends; in some cases they enhance clarity and readability, but in other cases they do not). Suppose a new project is starting development, with plans for nightly builds including a run through a static analyzer. Should it be part of the coding guidelines for the project to avoid (or strongly discourage) the use of continue and break, even if it can sacrifice a little readability and require excessive indentation? I'm most interested in how this applies to C code. Essentially, can the use of these control operators significantly complicate the static analysis of the code possibly resulting in additional false negatives, that would otherwise register a potential fault if break or continue were not used? (Of course a complete static analysis proving the correctness of an aribtrary program is an undecidable proposition, so please keep responses about any hands-on experience with this you have, and not on theoretical impossibilities) Thanks in advance!

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  • How to use OO for data analysis? [closed]

    - by Konsta
    In which ways could object-orientation (OO) make my data analysis more efficient and let me reuse more of my code? The data analysis can be broken up into get data (from db or csv or similar) transform data (filter, group/pivot, ...) display/plot (graph timeseries, create tables, etc.) I mostly use Python and its Pandas and Matplotlib packages for this besides some DB connectivity (SQL). Almost all of my code is a functional/procedural mix. While I have started to create a data object for a certain collection of time series, I wonder if there are OO design patterns/approaches for other parts of the process that might increase efficiency?

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  • How to fix these compiler errors?

    - by Sandra Schlichting
    I have this source code from 2001 that I would like to compile. It gives this: $ make g++ -O99 -Wall -DLINUX -pedantic -c -o audio.o audio.cpp In file included from audio.cpp:7: audio.h:14: error: use of enum ‘mad_flow’ without previous declaration audio.h:15: error: use of enum ‘mad_flow’ without previous declaration audio.h:17: error: use of enum ‘mad_flow’ without previous declaration audio.cpp: In function ‘mad_flow audio::input(void*, mad_stream*)’: audio.cpp:19: error: new declaration ‘mad_flow audio::input(void*, mad_stream*)’ audio.h:14: error: ambiguates old declaration ‘int audio::input(void*, mad_stream*)’ audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:23: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:23: error: within this context audio.h:10: error: ‘char* audio::stream::Buffer’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:26: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferPos’ is private audio.cpp:27: error: within this context audio.h:11: error: ‘size_t audio::stream::BufferSize’ is private audio.cpp:27: error: within this context audio.cpp: In function ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’: audio.cpp:49: error: new declaration ‘mad_flow audio::output(void*, const mad_header*, mad_pcm*)’ audio.h:15: error: ambiguates old declaration ‘int audio::output(void*, const mad_header*, mad_pcm*)’ audio.cpp: In function ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’: audio.cpp:83: error: new declaration ‘mad_flow audio::error(void*, mad_stream*, mad_frame*)’ audio.h:17: error: ambiguates old declaration ‘int audio::error(void*, mad_stream*, mad_frame*)’ audio.cpp: In constructor ‘audio::stream::stream(const char*)’: audio.cpp:119: error: ‘input’ was not declared in this scope audio.cpp:122: error: ‘output’ was not declared in this scope audio.cpp:123: error: ‘error’ was not declared in this scope make: *** [audio.o] Error 1 audio.h contains #include <stdlib.h> #include "mad.h" namespace audio { class stream { private: char* Buffer; size_t BufferSize, BufferPos; struct mad_decoder Decoder; friend enum mad_flow input(void* Data, struct mad_stream* MadStream); friend enum mad_flow output(void* Data, const struct mad_header* Header, struct mad_pcm* PCM); friend enum mad_flow error(void* Data, struct mad_stream* MadStream, struct mad_frame* Frame); public: stream(const char* FileName); ~stream(); void play(); }; } I have tried to just insert enum mad_flow {}; but that just gave a new problem. Can anyone see how to fix this?

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