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• Should I sell video tutorials on my own or via publishers like lynda.com? [closed]

  - by Derfder

  I am asking this because I am deciding between two models right now. One way is to create video tutorials on my own (make some short free videos and long pay per download/stream videos) or sell them to lynda.com or tutsplus. The 2nd way is easier, because they will do all the boring business stuff, will host the files to download etc. In that case, everything I need is a good microphone and obey their guidelines. On the other side if I do it on my own, I have to do all the unwanted business stuff, pay the server and other stuff. This is quite a big downside, however, I will have all the videos under my control in the future. I know that lynda.com has bigger attention and marketing that I am capable, but if you take e.g. phpvideotutrials.com (r.i.p ;), I think Leigh was very successful with relatively small budget. The interesting question will be the cost or how much will they pay me. Would it be less than if I sell it myself+monthly server hosting+other expenses? Any advice from people who actively sell their videos to some companies or do it on they own is highly appreciated.

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• How to use SAPI's SetNotifyCallbackFunction() in a CLR project with Windows Form as the interface wi

  - by manuel

  Hi, I'm trying to write a dll plugin for Winamp. I'm using Microsoft Visual Studio 2008 and Microsoft SAPI 5.1. I created the interface window using Windows Form (System::Windows::Forms::Form). I tried to use SetNotifyWIndowMessage(), but the method is never called when I speak to the microphone. So I tried using SetNotifyCallbackFunction(), but I got a compile error saying that I should use '&' in front of the method name in the parameter. However, when I add the '&', I got another compile error saying that i can't take the address of the method unless creating delegate instance. What should I do? Someone please help me..

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• Streaming input to System.Speech.Recognition.SpeechRecognitionEngine

  - by spurserh

  I am trying to do "streaming" speech recognition in C# from a TCP socket. The problem I am having is that SpeechRecognitionEngine.SetInputToAudioStream() seems to require a Stream of a defined length which can seek. Right now the only way I can think to make this work is to repeatedly run the recognizer on a MemoryStream as more input comes in. Here's some code to illustrate: SpeechRecognitionEngine appRecognizer = new SpeechRecognitionEngine(); System.Speech.AudioFormat.SpeechAudioFormatInfo formatInfo = new System.Speech.AudioFormat.SpeechAudioFormatInfo(8000, System.Speech.AudioFormat.AudioBitsPerSample.Sixteen, System.Speech.AudioFormat.AudioChannel.Mono); NetworkStream stream = new NetworkStream(socket,true); appRecognizer.SetInputToAudioStream(stream, formatInfo); // At the line above a "NotSupportedException" complaining that "This stream does not support seek operations." Does anyone know how to get around this? It must support streaming input of some sort, since it works fine with the microphone using SetInputToDefaultAudioDevice(). Thanks, Sean

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• Operation could not be completed. AVAudioRecorder iphone SDK

  - by Jonathan

  I am trying to record using the iphones microphone: This is my code: NSArray *paths = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString *documentsDirectory = [paths objectAtIndex:0]; // the path to write file NSString *appFile = [documentsDirectory stringByAppendingPathComponent:@"testing.mp3"]; NSURL *url = [NSURL fileURLWithPath:appFile isDirectory:NO]; NSDictionary *settings = [NSDictionary dictionaryWithObjectsAndKeys: [NSNumber numberWithFloat: 44100.0], AVSampleRateKey, [NSNumber numberWithInt: kAudioFormatMPEGLayer3], AVFormatIDKey, [NSNumber numberWithInt: 1], AVNumberOfChannelsKey, [NSNumber numberWithInt: AVAudioQualityLow], AVEncoderAudioQualityKey, nil]; NSError *error; recorder = [[AVAudioRecorder alloc] initWithURL:url settings:settings error:&error]; if ([recorder prepareToRecord] == YES){ [recorder record]; }else { int errorCode = CFSwapInt32HostToBig ([error code]); NSLog(@"Error: %@ [%4.4s])" , [error localizedDescription], (char*)&errorCode); } NSLog(@"BOOL = %d", (int)recorder.recording); This is the error I get: Operation could not be completed. (OSStatus error 1718449215.) And I can not work out why this doesn't work, as a lot of the code I got from a website. Jonathan

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• Java Speech Example: Encode, Stream, Decode, Play

  - by Dewayne

  I have been trying to find an example of this that I could use for a couple years, I'm ashamed to admit. I would like to see a working, compileable example (most that I find online don't compile or don't actually work) of reading from the microphone, encoding the voice data in a speech-friendly encoding such as Speex, and streaming that information in real time to a Decoder which then plays the audio. I suppose this example would simply echo what is said. I would like to ultimately use this to learn to make an audio mixing chat server.

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• Voice transmission over LAN using java?

  - by Ala ABUDEEB

  Hello I'm building a java application which works in a LAN environment, every computer on that LAN have this application installed on it, at some point i need this application to transfer voice simultaneously to all computer over the LAN (voice broadcasting) according to the following mechanism: Only one computer of the LAN can send voice using a microphone(the administrator) All computers receive that voice simultaneously (of course using my application) The voice should be recorded on the administrator computer after finishing the session. Could anyone give me an idea of how to use java in working with voice transmission? What java library can help me do that? Please help, thank you

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• Touchscreen using sound input?

  - by ricardowong

  Hi, i don't really know if it is actually possible, but i believe that it can be made. How possible is it to make a program that recognizes different sound bouncing from the screen and turn it into a position that will obviously be later fed to the mouse. I know that it sounds kind of dumb, but lately i've been noticing that a very dull, strong sound is made when touching the screen, and that sound varies when doing so at different positions. Probably the microphone "hears" differently because the screen acts as a drum with the casing. Anyways, what do you think, anyone has any experience programming with sound?

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• RockBand-like voice app for PC/OSX / Real time pitch display software

  - by Sai Emrys

  I played Rock Band 2 for the first time a little while ago (at Notacon). One thing I enjoyed about it was getting real-time feedback about my singing. I think it'd be neat to have something like that to run alongside my usual music, so that I can sing to random stuff in my music collection and know when I'm hitting the notes. Is there something like this for PC - ideally for OSX, and ideally that can just operate on arbitrary songs? I don't really care if it's game-like (though that's neat too); I just want it for the singing feedback. And I have no need for pitch correction - ideally what I'd see is just the pitches of the notes in the music and (on the same scale, differently displayed) of the live microphone. I tried to STFW but got no salient hits. :-/ Thanks!

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• Quickest and easiest way to implement speech to text conversion for a small speech subset.

  - by sgtpeppers

  Hi, I want to implement a system that receives speech through a microphone on my Mac OS x. I know arbitrary speech recognition is close to impossible without training the system so I'm willing to restrict it to 10 simple sentences. It must recognize with a high degree of accuracy which of these 10 sentences are being spoken, generate the text and add an entry to a remote MySQL database. With these being the architecture of the system I want to implement, could anyone give me an overview of what would be the best way to go about implementing this system? I'm looking for ideas like open source libraries to minimize the coding as this is just a prototype application for a demonstration. Basically I'm looking for a quick and easy solution. Thanks!

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• showSettings callback in Flex?

  - by Jim Robert

  I am pretty new to flex, so forgive me if this is an obvious question. Is there a way to open the Security.showSettings (flash.system.Security) with a callback? or at least to detect if it is currently open or not? My flex application is used for streaming audio, and is normally controlled by javascript, so I keep it hidden for normal use (via absolute positioning it off the page). When I need microphone access I need to make the flash settings dialog visible, which works fine, I move it into view and open the dialog. When the user closes it, I need to move it back off the screen so they don't see an empty flex app sitting there after they change their settings. thanks :)

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• Custom component which displays voice recognition button if available

  - by steff

  Hi evereyone, I'd like to create a custom component which supports voice recognition. It will primarily be an extended EditText which should show the microphone button for voice recognition if it is available. I wanted to to look at the search app-widget on the homescreen but I don't find it in the source. This is intended to use the voice recognition as some sort of dictation device, i.e. the user does not have to type but use his voice instead. So could anyone please point me in some direction? Thanks in advance, Steff

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• I need to consume an ocx for voice recording and playblack

  - by reinaldo Crespo

  Hi. The current ocx controls I'm using for voice recording and playback are not compatible with Windows 7. I'm already feeling the pressure to produce a Windows 7 compatible version of my software. The author has already stated that he is not planning to write a Windows 7 compatible ocx. I work from xharbour so I need to consume an OCX or write the whole thing (which I'd like to avoid and don't even know where to start). My basic needs are (1) to record dictation from the microphone with methods to pause and vox preferably, (2) save to file, (3) and later playback with methods to ff and rew. Thank you, Reinaldo.

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• (fluxus) learning curve

  - by Inaimathi

  I'm trying to have some fun with fluxus, but its manual and online docs all seem to assume that the reader is already an expert network programmer who's never heard of Scheme before. Consequently, you get passages that try to explain the very basics of prefix notation, but assume that you know how to pipe sound-card data into the program, or setup and connect to an OSC process. Is there any tutorial out there that goes the opposite way? IE, assumes that you already have a handle on the Lisp/Scheme thing, but need some pointers before you can properly set up sound sources or an OSC server? Barring that, does anyone know how to get (for example) the system microphone to connect to (fluxus), or how to get it to play a sound file from disk?

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• Redundancy algorithm for reading noisy bitstream

  - by Tedd Hansen

  I'm reading a lossy bit stream and I need a way to recover as much usable data as possible. There can be 1's in place of 0's and 0's in palce of 1's, but accuracy is probably over 80%. A bonus would be if the algorithm could compensate for missing/too many bits as well. The source I'm reading from is analogue with noise (microphone via FFT), and the read timing could vary depending on computer speed. I remember reading about algorithms used in CD-ROM's doing this in 3? layers, so I'm guessing using several layers is a good option. I don't remember the details though, so if anyone can share some ideas that would be great! :) Edit: Added sample data Best case data: in: 0000010101000010110100101101100111000000100100101101100111000000100100001100000010000101110101001101100111000101110000001001111011001100110000001001100111011110110101011100111011000100110000001000010111 out: 0010101000010110100101101100111000000100100101101100111000000100100001100000010000101110101001101100111000101110000001001111011001100110000001001100111011110110101011100111011000100110000001000010111011 Bade case (timing is off, samples are missing): out: 00101010000101101001011011001110000001001001011011001110000001001000011000000100001011101010011011001 in: 00111101001011111110010010111111011110000010010000111000011101001101111110000110111011110111111111101 Edit2: I am able to controll the data being sent. Currently attempting to implement simple XOR checking (though it won't be enough).

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• Flash and ActionScript

  - by Sonesh Dabhi

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

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• Audio processing in C# or C++

  - by melculetz

  Hi, I would like to create an application that uses AI techniques and allows the user to record a part of a song and then tries to find that song in a database of wav files. I would have liked to use some already existing libraries for the audio processing part. So, could you recommend any libraries in C# which can read a wav file, get input from microphone, have some audio filters (low pass, high pass, FFT etc) and maybe have the ability to plot the audio signal as well. I would prefer to develop in C#, but if there aren't good libraries for audio processing, I guess I could work in C++ as well. As far as I know, Mathlab already has the above mentioned functionalities, but I can't use it in my application.

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• Android: Using MediaRecorder to crop an existing audio file?

  - by user141146

  Hi, I'd like to take an existing mp3 file located on an SD card and arbitrarily crop it (e.g. crop from 0:12 to 1:14 in a 3 minute song). The only class that I've seen that seems remotely relevant to do this is the MediaRecorder class. My 'hope' would be to "record" an existing file like this: MediaRecorder recorder = new MediaRecorder(); recorder.setAudioSource(###some magical way of specifying an existing file??###); But this obviously doesn't work (setAudioSource() takes an int and seems to default to the phone's microphone). Is there a class or an approach that can be used to crop audio on the phone itself? TKS!!

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• Read the audio input level peak in Cocoa

  - by Kenneth Ballenegger

  I'm trying to make an audio-sensitive animation, and for that purpose, I'm looking for a way to look up the current audio level. I'm looking for the peak within a set amount of time. (Think the red bar that stays on for a second or so, on an audio meter.) I've searched around for for something like this, and the only thing I could find was how to read a movie's audio levels, and how Quartz Compositions have access to this thru their iTunes Visualizer protocol. I'm looking for a way to read this from the microphone, although I'm also interested if you know how to read this from an audio file. Thanks!

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• Making of a "Babbelbox" where you can speak to for partys

  - by Spidfire

  Ive got a project to make for a party, its called in holland a "Babbelbox". its a computer with a webcam and microphone that can be used to make a kind of video log of everyone who wants to say something about the party. But the problem is that i dont know where to start. ive made a kind of video show system in c but i cant save any data to a good format so it wont jam my harddisk in one hour full. Requirements: Record video + audio Recoding has to start after pressing a button Good compression over the recorded videos (would be even better if it can to be read by final cut pro or premiere pro) Light wight programm would be nice but i could scale up the computer power

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• audio stream sampling rate in linux

  - by farhan

  Im trying read and store samples from an audio microphone in linux using C/C++. Using PCM ioctls i setup the device to have a certain sampling rate say 10Khz using the SOUND_PCM_WRITE_RATE ioctl etc. The device gets setup correctly and im able to read back from the device after setup using the "read". int got = read(itsFd, b.getDataPtr(), b.sizeBytes()); The problem i have is that after setting the appropriate sampling rate i have a thread that continuously reads from /dev/dsp1 and stores these samples, but the number of samples that i get for 1 second of recording are way off the sampling rate and always orders of magnitude more than the set sampling rate. Any ideas where to begin on figuring out what might be the problem?

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• audio power on AudioQueue

  - by Tomoyuki

  Hi everyone. I'm now creating an Application using speech recognition.To check the Audio Power coming in through the microphone, I wrote a method as follows. -(void)checkPower(AudioqueRef)queue{ UInt32 expectedSize= sizeof(AudioQueueLevelMeterState); AudioQueueGetProperty(queue, kAudioQueueProperty_CurrentLevelMeter, audioLevels, expectedSize); NSLog(@"average:%f peak:%f",audioLevels.mAveragePower,audioLevels.mPeakPower); } I found that sometimes mAveragePower was larger than mPeakPower, and when mAveragePower was 1.0, in other words, averagePower is regarded as max, mPeakPower was lower than 1.0. I think that generally this result is inpossible. please Let me know if you have any information about sound power on CoreAudio. thanks.

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• Creating an object in javascript pointing to a table row with an "id"

  - by Matt

  I'm having trouble finding information online for creating a object in javascript and pointing it to a id in the html. Here's what I have so far for the JavaScript: function countRecords() { headRow=new Object(); //point to specific id here? var rowCount = 0; The HTML: <table id="prodTable"> <tr><th colspan="8">Digital Cameras</th></tr> <tr id="titleRow"> <th>Model</th> <th>Manufacturer</th> <th>Resolution</th> <th>Zoom</th> <th>Media</th> <th>Video</th> <th>Microphone</th> </tr>

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• How does one record audio from a Javascript based webapp?

  - by username

  I'm trying to write a web-app that records WAV files (eg: from the user's microphone). I know Javascript alone can not do this, but I'm interested in the least proprietary method to augment my Javascript with. My targeted browsers are Firefox for PC and Mac (so no ActiveX). Please share your experiences with this. I gather it can be done with Flash (but not as a WAV formated file). I gather it can be done with Java (but not without code-signing). Are these the only options? @dominic-mazzoni I'd like to record the file as a WAV because because the purpose of the webapp will be to assemble a library of good quality short soundbites. I estimate upload will be 50 MB, which is well worth it for the quality. The app will only be used on our intranet. UPDATE: There's now an alternate solution thanks to JetPack's upcoming Audio API: See https://wiki.mozilla.org/Labs/Jetpack/JEP/18

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• How do I write the audio stream to a memory buffer instead of a file using DirectShow?

  - by yngvedh

  Hi, I have made a sample application which constructs a filter graph to capture audio from the microphone and stream it to a file. Is there any filter which allows me to stream to a memory buffer instead? I'm following the approach outlined in an article on msdn and are currently using the CLSID_FileWriter object to write the audio to file. This works nicely, but I cannot figure out how to write to a memory buffer. Is there such a memory sink filter or do I have to create it myself? (I would prefer one which is bundled with windows XP)

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• Problem using AudioRecord with 8-bit encoding in android

  - by maxsap

  Hello, I have made an application that records from the phones microphone using the AudioRecord and 16-bit encoding, and I am able to playback the recording. For some compatibility reason I need to use 8-bit encoding, but when I try to run the same program using that encoding I keep getting an Invalid Audio Format. my code is : int bufferSize = AudioRecord.getMinBufferSize(11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT); AudioRecord recordInstance = new AudioRecord( MediaRecorder.AudioSource.MIC, 11025, AudioFormat.CHANNEL_CONFIGURATION_MONO, AudioFormat.ENCODING_PCM_8BIT, bufferSize); Any one knows what is the problem? according to the documentation AudioRecord is capable of 8-bit encoding. thanks in advanced maxsap.

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