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• fourier transform to transpose key of a wav file

  - by tbischel

  I want to write an app to transpose the key a wav file plays in (for fun, I know there are apps that already do this)... my main understanding of how this might be accomplished is to 1) chop the audio file into very small blocks (say 1/10 a second) 2) run an FFT on each block 3) phase shift the frequency space up or down depending on what key I want 4) use an inverse FFT to return each block to the time domain 5) glue all the blocks together But now I'm wondering if the transformed blocks would no longer be continuous when I try to glue them back together. Are there ideas how I should do this to guarantee continuity, or am I just worrying about nothing?

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• Can AVAudioSession do full duplex?

  - by Eric Christensen

  It would seem like it should be able to, but the following breakout test code can't do both: //play a file: NSArray *pathsArray = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES); NSString *documentsDirectory = [pathsArray objectAtIndex:0]; NSString* playFilePath=[documentsDirectory stringByAppendingPathComponent:@"testplayfile.wav"]; AVAudioPlayer *tempplayer = [[AVAudioPlayer alloc] initWithContentsOfURL:[NSURL fileURLWithPath:playFilePath] error:nil]; [tempplayer prepareToPlay]; [tempplayer play]; //and record a file: NSString* recFilePath=[documentsDirectory stringByAppendingPathComponent:@"testrecordfile.wav"]; AVAudioRecording *soundrecording = [[AVAudioRecorder alloc] initWithURL:[NSURL fileURLWithPath:recFilePath] settings:nil error:nil]; [soundrecording prepareToRecord]; [soundrecording record]; This is the minimum I can think of to individually play one file and record another. And this works just fine in the simulator. I can play back a file and record at the same time. But it doesn't work on the iphone itself. If I comment out either function, the other performs fine. The playback plays fine either alone or with both, if it's first. If I comment out the playback, the record records fine. (There's additional code to stop the recording not shown here.) So each works fine, but not together. I know audioQueue has a setting to allow both, but I don't see an analogue for AVAudioSessions. Any idea if it's possible, and if so, what I need to add? Thanks!

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• UIWebView/MPMoviePlayerController and the "Done" button

  - by David Sowsy

  I am using the UIWebView to load both streaming audio and video. I have properly set up the UIWebView delegate and I am receiving webViewDidStartLoading and webViewFinishedLoading events perfectly. The webview launches a full screen window (likely a MPMoviePlayerController) Apple's MoviePlayer example gets the array of Windows to determine which window the moviePlayerWindow is for adding custom drawing/getting at the GUI components. I believe this to be a bad practice/hack. My expectation is that I should be able to figure out when that button was clicked by either a delegate method or an NSNotification. It may also be the case that I have to poke around subviews or controllers with isKindOf calls, but I don't think those are correct approaches. Are my expectations incorrect, and if so, why? What is the correct way to bind an action to that "Done" button?

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• EXC_BAD_ACCESS NSUrlConnection

  - by Lars

  Hi all, i got an EXC_BAD_ACCESS when i perform the last line of the function (webData). -(void)requestSoap{ NSString *requestUrl = @"http://www.website.com/webservice.php"; NSString *soapMessage = @"the soap message"; //website and soapmessage are valid in original code. NSError **error; NSURLResponse *response; //Convert parameter string to url NSURL *url = [NSURL URLWithString:requestUrl]; NSMutableURLRequest *theRequest = [NSMutableURLRequest requestWithURL:url cachePolicy:NSURLRequestReloadIgnoringCacheData timeoutInterval:10]; NSString *msgLength = [NSString stringWithFormat:@"%d", [soapMessage length]]; //Create an XML message for webservice [theRequest addValue: @"text/xml; charset=utf-8" forHTTPHeaderField:@"Content-Type"]; [theRequest addValue: msgLength forHTTPHeaderField:@"Content-Length"]; [theRequest setHTTPMethod:@"POST"]; [theRequest setHTTPBody: [soapMessage dataUsingEncoding:NSUTF8StringEncoding]]; NSData *webData = [NSURLConnection sendSynchronousRequest:theRequest returningResponse:&response error:error]; } I tried not to release a thing, because what i read on the net is it's almost always a memory thing. When i debug the code (NSZombieEnabled = YES) this is what i get: [Session started at 2010-05-31 15:56:13 +0200.] GNU gdb 6.3.50-20050815 (Apple version gdb-1461.2) (Fri Mar 5 04:43:10 UTC 2010) Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "x86_64-apple-darwin".sharedlibrary apply-load-rules all Attaching to process 19856. test(19856) malloc: recording malloc stacks to disk using standard recorder test(19856) malloc: enabling scribbling to detect mods to free blocks test(19856) malloc: process 19832 no longer exists, stack logs deleted from /tmp/stack-logs.19832.test.w9Ek4L.index test(19856) malloc: stack logs being written into /tmp/stack-logs.19856.test.URRpQF.index Program received signal: “EXC_BAD_ACCESS”. Does anybody have a clue?? Thanks a lot! Lars

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• multi thread apps crashes in release mode

  - by etzarfat

  Hello, I'm using Visual Studio 2008 (programming in c). I've a weird problem I worte a program that has 2 threads that runs simultaneously, a recording thread (using audio card to record into memory) and a translation thread (using a speech engine to recognize the words). when I run my program in debug mode (aka setting a breakpoint in the code) it runs great, however when I run in debug mode or release mode (outside the visual studio enviroment) it crashes and give me the following exception: "Unhandled exception at 0x7c911129 in LowLevel.exe: 0xC0000005: Access violation reading location 0x014c7245." My stack looks: LowLevel.exe!__set_flsgetvalue() Line 256 + 0xc bytes C LowLevel.exe!_isleadbyte_l(int c=4359676, localeinfo_struct * plocinfo=0x00000001) Line 57 C++ 014b00d8() LowLevel.exe!PlayDateOfExam(int option=1) Line 2240 + 0x7 bytes C++ LowLevel.exe!NSCThread(void * arg=0x00000000) Line 1585 + 0xb bytes C++ kernel32.dll!7c80b729() winmm.dll!76b5b294() I uses the following file in my project "nsc.lib" and WinMM.lib" I'm not really familiar with threads I used a sample (which works great) and built on it. I saw a similiar question year on the forum but I didn't really understand the answers since I'm not familiar with with threads. Can someone help me? Thanks

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• DirectSound affects system volume on WinXP

  - by Anton

  Hello guys, I'm currently developing an audio engine that is used in voice network chat software. Everything is working fine - capture/playback/mixing channels. The problem is in using it under Windows XP. I've been getting user reports with information that their global system volume is set to zero after launching the application. I'm assuming that happens because of WaveOut/DSound conflict. How can I force DSound not to affect system volume? Playback device is initialized: DirectSoundCreate8(&GUID, &pAudio, NULL); and: pAudio-SetCooperativeLevel(parentWnd, DSSCL_PRIORITY); I'm currently not able to debug the application, cause I'm using Vista and everything is OK. Hope you can help me with this issue! Thanks a Lot! Regards, Anton.

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• ActionController::MethodNotAllowed

  - by Lowgain

  I have a rails model called 'audioclip'. I orginally created a scaffold with a 'new' action, which I replaced with 'new_record' and 'new_upload', becasue there are two ways to attach audio to this model. Going to /audioclips/new_record doesn't work because it takes 'new_record' as if it was an id. Instead of changing this, I was trying to just create '/record_clip' and '/upload_clip' paths. So in my routes.db I have: map.record_clip '/record_clip', :controller => 'audioclips', :action => 'new_record' map.upload_clip '/upload_clip', :controller => 'audioclips', :action => 'new_upload' When I navigate to /record_clip, I get ActionController::MethodNotAllowed Only get, head, post, put, and delete requests are allowed. I'm not extremely familiar with the inner-workings of routing yet. What is the problem here? (If it helps, I have these two statements above map.resources = :audioclips

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• How do I force one method to be executed before another method?

  - by RexOnRoids

  I've got 2 methods. One method starts playing an audio file (.mp3), the other method updates a UIToolBar to show a button (PLAY or PAUSE). These two methods are called in the following order: //Adds some UIBarButtonItems to a UIToolBar [self togglePlayer]; //Uses AVAudioPlayer [audioPlayer play]; I call the methods in the above order so that the (pause) button will be shown at the time the song starts playing. But, the problem is that the song starts playing first, and the UIToolBar remains unchanged for quite a while (from 2 to 5 secs) until the button is added and shown. What I want is for the button to be shown at the same time the song starts playing (i.e. NO DELAY). Is there any way to do this?

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• How to make QT support HTML 5 database?

  - by Mickey Shine

  I am using Qt 4.7.1 and embedded a webview in my app. But I got the following error when trying to visit http://webkit.org/demos/sticky-notes/ to test the HTML 5 database feature Failed to open the database on disk. This is probably because the version was bad or there is not enough space left in this domain's quota I compiled my static Qt library with the following command: configure --prefix=/usr/local/qt-static-release-db --accessibility --multimedia --audio-backend --svg --webkit --javascript-jit --script --scripttools --declarative --release -nomake examples -nomake demos --static --openssl -I /usr/local/ssl/include -L /usr/local/ssl/lib -confirm-license -sql-qsqlite -sql-qmysql -sql-qodbc

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• super dealloc error using multiple table view calsses

  - by padatronic

  I am new to Iphone apps and I am trying to build a tab based app. I am attempting to have a table ontop of an image in both tabs. On tab with a table of audio links and the other tab with a table of video links. This has all gone swimmingly, I have created two viewControllers for the two tables. All the code works great apart from to get it to work I have to comment out the super dealloc in the - (void)dealloc {} in the videoTableViewController for the second tab. If I don't I get the error message: FREED(id): message numberOfSectionsInTableView: sent to freed object please help, i have no idea why it is doing this...

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• Alternative to Rtmp and red5 for Iphone application

  - by IeN

  I am using red5 + rtmp in client-server flash application. There isnt audio/video streams in my applications, rtmp used for transfering messages from app to server and back. Now i need to develop application for Iphone and need help: 1) is there any rtmp implementation on Iphone ?? 2) If not, how could i solve this problem? Is there is any alternative to rtmp on iphone? And most important question : could it be solved without rewriting whole server part of application? (red5+ rtmp) Thanks

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• AVAudioPlayer Output to Speaker Problem

  - by Max

  After searching around for how to send AVAudioPlayer output to the iPhone's speaker, I found this: http://stackoverflow.com/questions/1064846/iphone-audio-playback-force-through-internal-speaker Despite setting the category correctly to AVAudioSessionCategoryPlayAndRecord, this solution doesn't seem to be working for me and won't even let the build compile, giving me this error: "_AudioSessionSetProperty", referenced from: ... ... ld: symbol(s) not found collect2: ld returned 1 exit status Am I not including something? I'm importing AudioToolbox, AVFoundation, and CoreAudio. My class implements AVAudioSessionDelegate, AVAudioRecorderDelegate, AVAudioPlayerDelegate, and UITextFieldDelegate. Any help would be greatly appreciated!

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• Why does ffmpeg stop randomly in the middle of a process?

  - by acidzombie24

  ffmpeg feels like its taking a long time. I then look at my output file and i see it stops between 6 and 8mbs. A fully encoded file is about 14mb. Why does ffmpeg stop? My code locks up on StandardOutput.ReadToEnd();. I had to kill the process (after seeing it not move for more then 10 seconds when i see it update every second previously) then i get the results of stdout and err. stdout is "" stderr is below. The output msg shows the filesize ended. I also see a drop in my CPU usage when it stops. I copyed the argument from visual studios. CD to the same working directory and ran the cmd (bin/ffmpeg) and pasted the argument. It was able to complete. int soundProcess(string infn, string outfn) { string aa, aa2; aa = aa2 = "DEAD"; var app = new Process(); app.StartInfo.UseShellExecute = false; app.StartInfo.RedirectStandardOutput = true; app.StartInfo.RedirectStandardError = true; //*/ app.StartInfo.FileName = @"bin\ffmpeg.exe"; app.StartInfo.Arguments = string.Format(@"-i ""{0}"" -ab 192k -y {2} ""{1}""", infn, outfn, param); app.Start(); try { app.PriorityClass = ProcessPriorityClass.BelowNormal; } catch (Exception ex) { if (!Regex.IsMatch(ex.Message, @"Cannot process request because the process .*has exited")) throw ex; } aa = app.StandardOutput.ReadToEnd(); aa2 = app.StandardError.ReadToEnd(); app.WaitForExit(); if (aa2.IndexOf("could not find codec parameters") != -1) return 1; else if (aa == "DEAD" || aa2 == "DEAD") return -1; else if (aa2.Length != 0) return -2; else return 0; } The output of stderr. stdout is empty. FFmpeg version SVN-r15815, Copyright (c) 2000-2008 Fabrice Bellard, et al. configuration: --enable-memalign-hack --enable-postproc --enable-swscale --enable-gpl --enable-libfaac --enable-libfaad --enable-libgsm --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libx264 --enable-libxvid --disable-ffserver --disable-vhook --enable-avisynth --enable-pthreads libavutil 49.12. 0 / 49.12. 0 libavcodec 52. 3. 0 / 52. 3. 0 libavformat 52.23. 1 / 52.23. 1 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 6. 1 / 0. 6. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 13 2008 10:28:29, gcc: 4.2.4 (TDM-1 for MinGW) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\dev\src\trunk\prjname\prjname\App_Data/temp/m/o/6304266424778814852': Duration: 00:12:53.36, start: 0.000000, bitrate: 154 kb/s Stream #0.0(und): Audio: aac, 44100 Hz, stereo, s16 Output #0, ipod, to 'C:\dev\src\trunk\prjname\prjname\App_Data\temp\m\o\2.m4a': Stream #0.0(und): Audio: libfaac, 44100 Hz, stereo, s16, 192 kb/s Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding size= 87kB time=4.74 bitrate= 150.7kbits/s size= 168kB time=9.06 bitrate= 151.9kbits/s size= 265kB time=14.28 bitrate= 151.8kbits/s size= 377kB time=20.29 bitrate= 152.1kbits/s size= 487kB time=26.22 bitrate= 152.1kbits/s size= 594kB time=32.02 bitrate= 152.1kbits/s size= 699kB time=37.64 bitrate= 152.1kbits/s size= 808kB time=43.54 bitrate= 152.0kbits/s size= 930kB time=50.09 bitrate= 152.2kbits/s size= 1058kB time=57.05 bitrate= 152.0kbits/s size= 1193kB time=64.23 bitrate= 152.1kbits/s size= 1329kB time=71.63 bitrate= 152.0kbits/s size= 1450kB time=78.16 bitrate= 152.0kbits/s size= 1578kB time=85.05 bitrate= 152.0kbits/s size= 1706kB time=92.00 bitrate= 152.0kbits/s size= 1836kB time=98.94 bitrate= 152.0kbits/s size= 1971kB time=106.25 bitrate= 151.9kbits/s size= 2107kB time=113.57 bitrate= 152.0kbits/s size= 2214kB time=119.33 bitrate= 152.0kbits/s size= 2345kB time=126.39 bitrate= 152.0kbits/s size= 2479kB time=133.56 bitrate= 152.0kbits/s size= 2611kB time=140.76 bitrate= 152.0kbits/s size= 2745kB time=147.91 bitrate= 152.1kbits/s size= 2880kB time=155.20 bitrate= 152.0kbits/s size= 3013kB time=162.40 bitrate= 152.0kbits/s size= 3146kB time=169.58 bitrate= 152.0kbits/s size= 3277kB time=176.61 bitrate= 152.0kbits/s size= 3412kB time=183.90 bitrate= 152.0kbits/s size= 3540kB time=190.80 bitrate= 152.0kbits/s size= 3670kB time=197.81 bitrate= 152.0kbits/s size= 3805kB time=205.08 bitrate= 152.0kbits/s size= 3932kB time=211.93 bitrate= 152.0kbits/s size= 4052kB time=218.38 bitrate= 152.0kbits/s size= 4171kB time=224.82 bitrate= 152.0kbits/s size= 4277kB time=230.55 bitrate= 152.0kbits/s size= 4378kB time=235.96 bitrate= 152.0kbits/s size= 4486kB time=241.79 bitrate= 152.0kbits/s size= 4592kB time=247.50 bitrate= 152.0kbits/s size= 4698kB time=253.21 bitrate= 152.0kbits/s size= 4804kB time=258.95 bitrate= 152.0kbits/s size= 4906kB time=264.41 bitrate= 152.0kbits/s size= 5012kB time=270.09 bitrate= 152.0kbits/s size= 5118kB time=275.85 bitrate= 152.0kbits/s size= 5234kB time=282.10 bitrate= 152.0kbits/s size= 5331kB time=287.39 bitrate= 151.9kbits/s size= 5445kB time=293.55 bitrate= 152.0kbits/s size= 5555kB time=299.40 bitrate= 152.0kbits/s size= 5665kB time=305.37 bitrate= 152.0kbits/s size= 5766kB time=310.80 bitrate= 152.0kbits/s size= 5876kB time=316.70 bitrate= 152.0kbits/s size= 5984kB time=322.50 bitrate= 152.0kbits/s size= 6094kB time=328.49 bitrate= 152.0kbits/s size= 6212kB time=334.76 bitrate= 152.0kbits/s size= 6327kB time=340.99 bitrate= 152.0kbits/s

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• Fast block placement algorithm, advice needed?

  - by James Morris

  I need to emulate the window placement strategy of the Fluxbox window manager. As a rough guide, visualize randomly sized windows filling up the screen one at a time, where the rough size of each results in an average of 80 windows on screen without any window overlapping another. It is important to note that windows will close and the space that closed windows previously occupied becomes available once more for the placement of new windows. The window placement strategy has three binary options: Windows build horizontal rows or vertical columns (potentially) Windows are placed from left to right or right to left Windows are placed from top to bottom or bottom to top Why is the algorithm a problem? It needs to operate to the deadlines of a real time thread in an audio application. At this moment I am only concerned with getting a fast algorithm, don't concern yourself over the implications of real time threads and all the hurdles in programming that that brings. So far I have two choices which I have built loose prototypes for: 1) A port of the Fluxbox placement algorithm into my code. The problem with this is, the client (my program) gets kicked out of the audio server (JACK) when I try placing the worst case scenario of 256 blocks using the algorithm. This algorithm performs over 14000 full (linear) scans of the list of blocks already placed when placing the 256th window. 2) My alternative approach. Only partially implemented, this approach uses a data structure for each area of rectangular free unused space (the list of windows can be entirely separate, and is not required for testing of this algorithm). The data structure acts as a node in a doubly linked list (with sorted insertion), as well as containing the coordinates of the top-left corner, and the width and height. Furthermore, each block data structure also contains four links which connect to each immediately adjacent (touching) block on each of the four sides. IMPORTANT RULE: Each block may only touch with one block per side. The problem with this approach is, it's very complex. I have implemented the straightforward cases where 1) space is removed from one corner of a block, 2) splitting neighbouring blocks so that the IMPORTANT RULE is adhered to. The less straightforward case, where the space to be removed can only be found within a column or row of boxes, is only partially implemented - if one of the blocks to be removed is an exact fit for width (ie column) or height (ie row) then problems occur. And don't even mention the fact this only checks columns one box wide, and rows one box tall. I've implemented this algorithm in C - the language I am using for this project (I've not used C++ for a few years and am uncomfortable using it after having focused all my attention to C development, it's a hobby). The implementation is 700+ lines of code (including plenty of blank lines, brace lines, comments etc). The implementation only works for the horizontal-rows + left-right + top-bottom placement strategy. So I've either got to add some way of making this +700 lines of code work for the other 7 placement strategy options, or I'm going to have to duplicate those +700 lines of code for the other seven options. Neither of these is attractive, the first, because the existing code is complex enough, the second, because of bloat. The algorithm is not even at a stage where I can use it in the real time worst case scenario, because of missing functionality, so I still don't know if it actually performs better or worse than the first approach. What else is there? I've skimmed over and discounted: Bin Packing algorithms: their emphasis on optimal fit does not match the requirements of this algorithm. Recursive Bisection Placement algorithms: sounds promising, but these are for circuit design. Their emphasis is optimal wire length. Both of these, especially the latter, all elements to be placed/packs are known before the algorithm begins. I need an algorithm which works accumulatively with what it is given to do when it is told to do it. What are your thoughts on this? How would you approach it? What other algorithms should I look at? Or even what concepts should I research seeing as I've not studied computer science/software engineering? Please ask questions in comments if further information is needed. [edit] If it makes any difference, the units for the coordinates will not be pixels. The units are unimportant, but the grid where windows/blocks/whatever can be placed will be 127 x 127 units.

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• Publishing SWF using Adobe Flash

  - by Kim

  Hello everyone, I have a SWF file which contains of an image (1keyframe) and also, it contains an AS3 file with the following codes: var loader:Loader=new Loader(); var ur:URLRequest=new URLRequest("1.swf"); loader.load(ur); addChild(loader); so basically, i am trying to play the swf file (1.swf - an audio) while the image is being displayed. What I want to know is how will I be able to publish this project into an SWF file which can still play as expected even without the raw 1.swf file. I can publish SWF right now but when I delete the 1.swf file, my generated swf can only display the image. Help me please. Thanks in advance :)

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• Streaming Media Server and Hosting

  - by Ryan Max

  My partner and I have a webcam site that basically runs the old-school method....Every 0.5 seconds the javascript reloads the image in the browser from the webcam. However we are wanting to upgrade to a streaming media server to get higher quality video, and possibly audio. We aren't tied to any one specific file format or server type, as of right now we are leaning towards slicehost (as scalability is important), and installing darwin streaming server or wowza. This is meant to be a live stream. Does anyone have any suggestions for hosts/server software?

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• How to use Speech 2 Text in Microsoft Surface

  - by Roflcoptr

  I'd like to use some speech 2 text in my microsoft surface application. I saw that it is possible, but I don't really know where to start. Is there any framework/library available, or a code snippet, or a tutorial?? I don't even know exactly what i should google for ;) ===EDIT=== I read that it is necessary to use a grammar to recognize words. So if I want to proceed free text, is there a predefined grammar for the english language? Or is it a better choice to don't use speech2text but just audio files instead?

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• C++ from SpeakHere in iPhone app

  - by niklassaers

  Hi guys, I've made a template app where I've grabbed the recording part of the SpeakHere example and removed the file handling part, but I'm struggeling to get the C++ part of the app working right. As soon as it enters the C++ class, it gets syntax errors. If I don't import the header files from C++ (and then of course don't use the code) into my Objective C classes, all works fine. I cannot see the difference between how I'm doing it and the example is doing it. Can you see the difference? I've posted the entire code here: http://github.com/niklassaers/testFFT The build errors I get are: testFFT/CAStreamBasicDescription.h:91:0 testFFT/CAStreamBasicDescription.h:91: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'CAStreamBasicDescription' testFFT/CAStreamBasicDescription.h:298:0 testFFT/CAStreamBasicDescription.h:298: error: expected '=', ',', ';', 'asm' or '__attribute__' before '<' token testFFT/CAStreamBasicDescription.h:299:0 testFFT/CAStreamBasicDescription.h:299: error: expected '=', ',', ';', 'asm' or '__attribute__' before '==' token testFFT/CAStreamBasicDescription.h:301:0 testFFT/CAStreamBasicDescription.h:301: error: expected '=', ',', ';', 'asm' or '__attribute__' before '!=' token testFFT/CAStreamBasicDescription.h:302:0 testFFT/CAStreamBasicDescription.h:302: error: expected '=', ',', ';', 'asm' or '__attribute__' before '<=' token testFFT/CAStreamBasicDescription.h:303:0 testFFT/CAStreamBasicDescription.h:303: error: expected '=', ',', ';', 'asm' or '__attribute__' before '>=' token testFFT/CAStreamBasicDescription.h:304:0 testFFT/CAStreamBasicDescription.h:304: error: expected '=', ',', ';', 'asm' or '__attribute__' before '>' token testFFT/CAStreamBasicDescription.h:307:0 testFFT/CAStreamBasicDescription.h:307: error: expected ';', ',' or ')' before '&' token testFFT/CAXException.h:65:0 testFFT/CAXException.h:65: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'CAX4CCString' testFFT/CAXException.h:87:0 testFFT/CAXException.h:87: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'CAXException' testFFT/AQRecorder.h:59:0 testFFT/AQRecorder.h:59: error: expected '=', ',', ';', 'asm' or '__attribute__' before 'AQRecorder' testFFT/RecorderLink.h:57:0 testFFT/RecorderLink.h:57: error: expected specifier-qualifier-list before 'AQRecorder' testFFT/RecorderLink.h:62:0 testFFT/RecorderLink.h:62: error: expected specifier-qualifier-list before 'AQRecorder' Any idea what's going on here? Cheers Nik

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• Testing MPMoviePlayerViewController in iPad simulator

  - by hgpc

  I have a view that shows a MPMoviePlayerViewController modally. When testing it in the iPad simulator it works well on the first try. If I dismiss the video and then show the view again, the player only plays the audio, but not the video. Is this a simulator quirk or am I doing something wrong? Here's my code: - (void)viewWillAppear:(BOOL)animated { [super viewWillAppear:animated]; MPMoviePlayerViewController* v = [[MPMoviePlayerViewController alloc] initWithContentURL:url]; [[NSNotificationCenter defaultCenter] addObserver:self selector: @selector(playbackDidFinish:) name:MPMoviePlayerPlaybackDidFinishNotification object:v.moviePlayer]; [self presentMoviePlayerViewControllerAnimated:v]; [v release]; } -(void) playbackDidFinish:(NSNotification*)aNotification { MPMoviePlayerController *player = [aNotification object]; [[NSNotificationCenter defaultCenter] removeObserver:self name:MPMoviePlayerPlaybackDidFinishNotification object:player]; [player stop]; [self dismissMoviePlayerViewControllerAnimated]; }

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• Multimedia files written over WAN are getting truncated

  - by Dean

  I use the windows Multimedia API to create .wav files. 1. Open file with mmsioOpen 2. Creates WAVE,frm and data chunks using mmioCreateChunk 3. Write audio data using mmioWrite 4. Ascend out of the chunks using mmioAscend 5. Close file using mmioClose The file is being written into a temporary location, so after it has been closed it gets copied to another location using the CopyFile. This program is written in C++ and works great until the file it is writing resides over a WAN in a different city or country. The end result is a wav file that should be 20-30 seconds long ends up being 4 secodns long. It is always the last bit that is missing, so when you play it back it just stops before then of the recording. I initially thought that maybe I was copying the file too soon so as a test I put in a pause of 30 seconds after closing the file using Sleep(30000), but this made no difference to either it being truncated or by how much. I have modified the program to write to a file in parrallel using CreateFile and WriteFile, and the result is the same, so it is not an issue specifically with the mmio API's. Does anyone have any ideas why this is happening and if there is a work-around to it? I suspect that I may end up having the temporary location on the local drive, but this is quite a big change to the application as well as existing deployments. thanks for everyones time Dean

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• Videoconference using Flash and SIP

  - by Júlio Santos

  The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 conference is required.

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• Using PHP to store results of a post request

  - by Paul M

  Im currently working with an API which requires we send our collection details in xml to their server using a post request. Nothing major there but its not working, so I want to output the sent xml to a txt file so I can look at actually whats being sent!! Instead of posting to the API im posting to a document called target, but the xml its outputting its recording seems to be really wrong. Here is my target script, note that the posting script posts 3 items, so the file being written should have details of each post request one after the other. <?php error_reporting(E_ALL); ini_set('display_errors', 1); // get the request data... $payload = ''; $fp = fopen('php://input','r'); $output_file = fopen('output.txt', 'w'); while (!feof($fp)) { $payload .= fgets($fp); fwrite($output_file, $payload); } fclose($fp); fclose($output_file); ?> I also tried the following, but this just recorded the last post request so only 1 collection item was recorded in the txt file, instead of all 3 output_file = fopen('output.txt', 'w'); while (!feof($fp)) { $payload .= fgets($fp); } fwrite($output_file, $payload); fclose($fp); fclose($output_file); I know im missing something really obvious, but ive been looking at this all morning!

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• DSP - How are frequency amplitudes modified using DFT?

  - by Trap

  I'm trying to implement a DFT-based equalizer (not FFT) for the sole purpose of learning. To check if it works I took an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. Now I tried to silence some frequency bands, just by setting their amplitudes to zero before resynthesis, but definitely it's not the way to go. What I get is a rather distorted signal. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. I first tried to modify the real part amplitudes only, then modifying both the real and imaginary part amplitudes. I also tried to convert the DFT output to polar notation, then modifying the magnitude and convert back to rectangular notation, but none of this is working. Can someone show me what I'm doing wrong? I tried to find info on this subject in the internet but couldn't find any. Thanks in advance.

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• C# How to Present Such Question?

  - by ikurtz

  greetings! i have a C# game program that im developing. it uses sound samples and winsock. when i test run the game most of the audio works fine but from time to time if it is multiple samples being played sequentially the application form shakes a little bit and then goes back to its old position. how do i go about debugging this or present it to you folks in a manageable manner? im sure no one is going to want the whole app code in fear of virus attacks. please guide me.. thanking you. EDIT: i have not been able to pin down any code section that produces this result. it just does and i cannot explain it. EDIT: no the x/y position are not changing. the window like shakes around a few pixels and then goes back to the position were it was before the shake.

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• How can HTML5 "replace" Flash?

  - by Kassini

  A topic of debate that's seen a resurgence since the unveiling of the iPad is the issue of Flash versus HTML5. There are those that suggest that HTML5 will one day supplant/replace Adobe Flash. I do not develop software that runs in a browser, so my (limited) understanding is: HTML is a pure-text markup language that is delivered over HTTP to a client browser. The client browser interprets the markup and renders (with varying degrees of success) the page according to an standard specification. Adobe Flash is a propriety framework for working with audio, video, sound and raster/vector graphics. It requires special authoring tools (a compiler perhaps?) and a custom player that's available as a plug-in to most common browsers. Could someone please explain (to this C/C++ developer) how it is possible from a technical/coding point-of-view that a text-based markup language (HTML5) could be considered a replacement to a multimedia framework (Flash)? Please no opinionated arguments - just technical facts.

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