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• Converting Milliseconds to Timecode

  - by Jeff

  I have an audio project I'm working on using BASS from Un4seen. This library uses BYTES mainly but I have a conversion in place that let's me show the current position of the song in Milliseconds. Knowing that MS = Samples * 1000 / SampleRate and that Samples = Bytes * 8 / Bits / Channels So here's my main issue and it's fairly simple... I have a function in my project that converts the Milliseconds to TimeCode in Mins:Secs:Milliseconds. Public Function ConvertMStoTimeCode(ByVal lngCurrentMSTimeValue As Long) ConvertMStoTimeCode = CheckForLeadingZero(Fix(lngCurrentMSTimeValue / 1000 / 60)) & ":" & _ CheckForLeadingZero(Int((lngCurrentMSTimeValue / 1000) Mod 60)) & ":" & _ CheckForLeadingZero(Int((lngCurrentMSTimeValue / 10) Mod 100)) End Function Now the issue comes within the Seconds calculation. Anytime the MS calculation is over .5 the seconds place rounds up to the next second. So 1.5 seconds actually prints as 2.5 seconds. I know for sure that using the Int conversion causes a round down and I know my math is correct as I've checked in a calculator 100 times. I can't figure out why the number is rounding up. Any suggestions?

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• Debugging Key-Value-Observing overflow.

  - by Paperflyer

  I wrote an audio player. Recently I started refactored some of the communication flow to make it fully MVC-compliant. Now it crashes, which in itself is not surprising. However, it crashes after a few seconds inside the Cocoa key-value-observing routines with a HUGE stack trace of recursive calls to NSKeyValueNotifyObserver. Obviously, it is recursively observing a value and thus overflowing the NSArray that holds pending notifications. According to the stack trace, the program loops from observeValueForKeyPath to setMyValue and back. Here is the according code: - (void)observeValueForKeyPath:(NSString *)keyPath ofObject:(id)object change:(NSDictionary *)change context:(void *)context { if ([keyPath isEqual:@"myValue"] && object == myModel && [self myValue] != [myModel myValue]) { [self setMyValue:[myModel myValue]; } } and - (void)setMyValue:(float)value { myValue = value; [myModel setMyValue:value]; } myModel changes myValue every 0.05 seconds and if I log the calls to these two functions, they get called only every 0.05 seconds just as they should be, so this is working properly. The stack trace looks like this: -[MyDocument observeValueForKeyPath:ofObject:change:context:] NSKeyValueNotifyObserver NSKeyValueDidChange -[NSObject(NSKeyValueObserverNotification) didChangeValueForKey:] -[MyDocument setMyValue:] _NSSetFloatValueAndNotify …repeated some ~8k times until crash Do you have any idea why I could still be spamming the KVO queue?

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• AVAudioPlayer only initializes with some files

  - by Brendan

  Hi everyone, I'm having trouble playing some files with AVAudioPlayer. When I try to play a certain m4a, it works fine. It also works with an mp3 that I try. However it fails on one particular mp3 every time (15 Step, by Radiohead), regardless of the order in which I try to play them. The audio just does not play, though the view loading and everything that happens concurrently happens correctly. The code is below. I get the "Player loaded." log output on the other two songs, but not on 15 Step. I know the file path is correct (I have it log outputted earlier in the app, and it is correct). Any ideas? NSData *musicData = [NSData dataWithContentsOfURL:[[NSURL alloc] initFileURLWithPath:[[NSBundle mainBundle] pathForResource:[song filename] ofType:nil]]]; NSLog([[NSBundle mainBundle] pathForResource:[song filename] ofType:nil]); if(musicData) { NSLog(@"File found."); } self.songView.player = [[AVAudioPlayer alloc] initWithData:musicData error:nil]; if(self.songView.player) { NSLog(@"Player loaded."); } [self.songView.player play]; NSLog(@"You should be hearing something now."); Thanks, Brendan

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• Tablet as Car Computer

  - by Austin Fitzpatrick

  Okay, so forward this off to the right place if this isn't the right place to ask this question. I want to use a tablet computer as a car-computer. Minimum features would be to run my music (through iPod, Pandora, whatever I want) and GPS Navigation, watch TV or movies while I'm parked waiting for people, and the hard one: it needs to answer my phone calls with a pleasant interface much like in-dash systems do. It needs to detect that my phone is ringing in my pocket and provide an on-screen answer/ignore and then route the audio through the cars speakers. It would be nice to dial out and have address book access, but that is somewhat secondary. I have an iPhone myself and I figured that an iPad with 3G might make a good system for this - but I'm open to other options if an iPad can't do everything I need. I'm willing to write code, and I'm willing to jailbreak one or both devices. I haven't done much work in Obj-C, but I'm not opposed to learning a new language for this project. It's self enrichment for the most part, as I'm sure I can buy an indash entertainment system for less. Does anyone have experience with the iPhone/iPad SDK that can tell me whether or not it would be possible to get it an iPad to answer my calls in the car? What about an Android tablet? (that Adam looks promising, too).

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• Touch friendly GUI in Windows Mobile

  - by vonolsson

  I'm porting an audio processing application written in C++ from Windows to Windows Mobile (version 5+). Basically what I need to port is the GUI. The application is quite complicated and the GUI will need to be able to offer a lot of functionality. I would like to create a touch friendly user interface that also looks good. Which basically means that standard WinMo controls are out the window. I've looked at libraries such as Fluid and they look like something I would like to use. However, as I said I'm developing i C++. Even though it would be possible to only write the GUI part i some .NET language I rather not. My experience with .NET on Windows Mobile is that it doesn't work very well... Can anyone either suggest a C/C++ touch friendly GUI library for Windows Mobile or some kind of "best practices" document/how-to on how to use the standard Windows Mobile controls in order to make the touch friendly and also work and look well in later versions of Windows Mobile (in particular version 6.5)?

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• custom view on iphone's native media player(MPMoviePlayerController)

  - by sneha

  I am building an application that implements a custom view on iPhone’s native media player. I want your help in deciding directions to lay this effort. At present I have find out that iPhone SDK doesn’t support APIs to customize media player. I need these things in the player: I would like to have custom views i.e. want to change all control buttons on player like Play/Pause, seek bar etc. The background of player will also need to be different. The player has to play audio or video file from local/remote location. Can i use MPMoviePlayerController if it can be customized (How to do it ??). However, any other third party player approved by iPhone which has an ability to download and play the media file from local/remote location is also fine. It will be great to have an access to media player buffer so that it can be encrypted. I have following questions: 1.Any help in building/customizing player..... 2.Do you see issues in signing of application? 3.Does Apple have any restrictions on customizing media player? 4.Any sample iPhone application where media player is customized Any help in this regard is highly appreciated.

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• error to start Windows Media Encoder

  - by George2

  Hello everyone, I am using the following code snippet to run on Windows Server 2003 x64 edition. I met with the following error when invoking encoder.start method. I am using Windows Media Encoder 9. System.Runtime.InteropServices.COMException 0xC00D1B67 My code snippet is below, does anyone have any ideas what is wrong? IWMEncSourceGroup SrcGrp; IWMEncSourceGroupCollection SrcGrpColl; SrcGrpColl = encoder.SourceGroupCollection; SrcGrp = (IWMEncSourceGroup)SrcGrpColl.Add("SG_1"); IWMEncVideoSource2 SrcVid; IWMEncSource SrcAud; SrcVid = (IWMEncVideoSource2)SrcGrp.AddSource(WMENC_SOURCE_TYPE.WMENC_VIDEO); SrcAud = SrcGrp.AddSource(WMENC_SOURCE_TYPE.WMENC_AUDIO); SrcVid.SetInput("ScreenCap://ScreenCapture1", "", ""); SrcAud.SetInput("Device://Default_Audio_Device", "", ""); // Specify a file object in which to save encoded content. IWMEncFile File = encoder.File; string CurrentFileName = Guid.NewGuid().ToString(); File.LocalFileName = CurrentFileName; CurrentFileName = File.LocalFileName; // Choose a profile from the collection. IWMEncProfileCollection ProColl = encoder.ProfileCollection; IWMEncProfile Pro; for (int i = 0; i < ProColl.Count; i++) { Pro = ProColl.Item(i); if (Pro.Name == "Screen Video/Audio High (CBR)") { SrcGrp.set_Profile(Pro); break; } } encoder.Start(); thanks in advance, George

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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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• How can I attach a file using Wordpress custom fields / meta boxes?

  - by shipshape

  I am using Wordpress's add_meta_box() function to add customized meta fields to the Add New Post page, like this. I want one of these fields to allow the user to upload a file, so that a single image, pdf, audio file, or video can be associated with the post. The closest example I've seen is this one. Unfortunately it does not suit my needs, as I want my file to be processed by Wordpress's Media Uploader - so it should appear in the Media Library afterwards, and thumbnails should be generated according to the Media settings. I think ideally there would be a way to tap into Wordpress's existing Add Media dialog, and simply output the URL of the uploaded file into a text box, but I don't see how to do that. This question is similar, but the answers are a little clunky - I would like to keep this super simple for my end users. How can I accomplish this? Please do not recommend plugins such as Flutter or Magic Fields - I have tried these and they do not suit my purposes (I want the images to be processed by Wordpress's Media Uploader). I am using Wordpress 3.0-alpha. Thanks!

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• Dynamically created iframe used to download file triggers onload with firebug but not without

  - by justkt

  EDIT: as this problem is now "solved" to the point of working, I am looking to have the information on why. For the fix, see my comment below. I have an web application which repeatedly downloads wav files dynamically (after a timeout or as instructed by the user) into an iframe in order to trigger the a default audio player to play them. The application targets only FF 2 or 3. In order to determine when the file is downloaded completely, I am hoping to use the window.onload handler for the iframe. Based on this stackoverflow.com answer I am creating a new iframe each time. As long as firebug is enabled on the browser using the application, everything works great. Without firebug, the onload never fires. The version of firebug is 1.3.1, while I've tested Firefox 2.0.0.19 and 3.0.7. Any ideas how I can get the onload from the iframe to reliably trigger when the wav file has downloaded? Or is there another way to signal the completion of the download? Here's the pertinent code: HTML (hidden's only attribute is display:none;): <div id="audioContainer" class="hidden"> </div> JavaScript (could also use jQuery, but innerHTML is faster than html() from what I've read): waitingForFile = true; // (declared at the beginning of closure) $("#loading").removeClass("hidden"); var content = "<iframe id='audioPlayer' name='audioPlayer' src='" + /path/to/file.wav + "' onload='notifyLoaded()'></iframe>"; document.getElementById("audioContainer").innerHTML = content; And the content of notifyLoaded: function notifyLoaded() { waitingForFile = false; // (declared at beginning of the closure) $("#loading").addClass("hidden"); } I have also tried creating the iframe via document.createElement, but I found the same behavior. The onload triggered each time with firebug enabled and never without it. EDIT: Fixed the information on how the iframe is being declared and added the callback function code. No, no console.log calls here.

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• SoundManager / Jquery / Regular expression : Parse class name before certain character To Get SoundI

  - by j-man86

  So I am trying to access a jquery soundmanager variable from one script (wpaudio.js – from the wp-audio plugin) inside of another (init.js – my own javascript). I am creating an alternate pause/play button higher up on the page and need to resume the current soundID, which is contained as part of a class name in the DOM. Here is the code that creates that class name in wpaudio.js: function wpaButtonCheck() { if (!this.playState || this.paused) jQuery('#' + this.sID + '_play').attr('src', wpa_url + '/wpa_play.png'); else jQuery('#' + this.sID + '_play').attr('src', wpa_url + '/wpa_pause.png'); } Here is the output: where wpa0 would be the sID of the sound I need. My current script in init.js is: $('.mixesSidebar #currentSong .playBtn').toggle(function() { soundManager.pauseAll(); $(this).addClass('paused'); }, function() { soundManager.resumeAll(); $(this).removeClass('paused'); }); I need to change resumeAll to "resume(this.sID)", but I need to somehow store the sID onclick and call it in the above function. Alternately, I think a regular expression that could get the class name of the current play button and either parse the string up to the "_play" or use a trim function to get rid of "_play"– but I'm not sure how to do this. Thanks for your help!

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• SoundManager / Jquery : Get SoundID sID

  - by j-man86

  So I am trying to access a jquery soundmanager variable from one script (wpaudio.js – from the wp-audio plugin) inside of another (init.js – my own javascript). I am creating an alternate pause/play button higher up on the page and need to resume the current soundID, which is contained as part of a class name in the DOM. Here is the code that creates that class name in wpaudio.js: function wpaButtonCheck() { if (!this.playState || this.paused) jQuery('#' + this.sID + '_play').attr('src', wpa_url + '/wpa_play.png'); else jQuery('#' + this.sID + '_play').attr('src', wpa_url + '/wpa_pause.png'); } Here is the output: <img src="http://24.232.185.173/wordpress/wp-content/plugins/wpaudio-mp3-player/wpa_play.png" class="wpa_play" id="wpa0_play"> where wpa0 would be the sID of the sound I need. My current script in init.js is: $('.mixesSidebar #currentSong .playBtn').toggle(function() { soundManager.pauseAll(); $(this).addClass('paused'); }, function() { soundManager.resumeAll(); $(this).removeClass('paused'); }); I need to change resumeAll to "resume(this.sID)", but I need to somehow store the sID onclick and call it in the above function. Alternately, I think a regular expression that could get the class name of the current play button and either parse the string up to the "_play" or use a trim function to get rid of "_play"– but I'm not sure how to do this. Thanks for your help!

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• process of connecting RTP with SIP via SDP & land lines

  - by TacB0sS

  Hello to everyone, I have a problem with starting a media session and to combine it with my SIP client. I've designed a recursive SIP client that reuse the same request template to send the next requests to server, according to the acceptable sequences noted in the RFC's, and examples that I read. as far as I could tell the SIP part is working fine registers to server invites, and authenticates. I didn't complete any calls to clients yet because of the content header needs to be filled up (which I didn't yet so I get a 503 from the server which is OK I guess). for a long time I didn't know where to start with the media session, and slowly learned how to use the JMF and I've constructed an object that handles RTP transmitting, now I'm standing at the cross road, on the one hand I have my SIP signaling but it needs the SDP content header to complete the invite, and on the other I have the RTP which is knows how to p2p. For me to complete my design I require your help with the following questions: Is there an easy//a simple//an implemented way to convert the Audio/Video format from the JMF into SDP media headers? or even a generator that I would input all the parameters for the content header, and it would generate a content header fast, or do I have to implement this myself? Once I've finished constructing the SDK and the SIP is up and running and I get an OK response from the server (after ringing and all), how do I start the media session? do I connect p2p according to caller details I send in the SIP invite? If 2 is correct, then how does a connection to land lines would be? does land lines knows that once they send an OK back to server they listen/start RTP session on a specific port? Or did I get everything wrong? :-/ I really appreciate any help I could I get, I looked every where for answers but they are not clear, they ignore question 2 as if it was an obvious thing, but for me it just isn't. Thank in advance, Adam Zehavi.

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• C# StripStatusText Update Issue

  - by ikurtz

  I am here due to a strange behaviour in Button_Click event. The code is attached. The issue is the first StripStatus message is never displayed. any ideas as to why? private void FireBtn_Click(object sender, EventArgs e) { // Control local controls for launching attack AwayTableLayoutPanel.Enabled = false; AwayCancelBtn.Enabled = false; FireBtn.Enabled = false; ////////////// Below statusBar message is never displayed but the folowing sound clip is. GameToolStripStatusLabel.Text = "(Home vs. Away)(Attack Coordinate: (" + GameModel.alphaCoords(GridLock.Column) + "," + GridLock.Row + "))(Action: Fire)"; //////////////////////////////////////////// if (audio) { SoundPlayer fire = new SoundPlayer(Properties.Resources.fire); fire.PlaySync(); fire.Dispose(); } // compile attack message XmlSerializer s; StringWriter w; FireGridUnit fireGridUnit = new FireGridUnit(); fireGridUnit.FireGridLocation = GridLock; s = new XmlSerializer(typeof(FireGridUnit)); w = new StringWriter(); s.Serialize(w, fireGridUnit); ////////////////////////////////////////////////////////// // send attack message GameMessage GameMessageAction = new GameMessage(); GameMessageAction.gameAction = GameMessage.GameAction.FireAttack; GameMessageAction.statusMessage = w.ToString(); s = new XmlSerializer(typeof(GameMessage)); w = new StringWriter(); s.Serialize(w, GameMessageAction); SendGameMsg(w.ToString()); GameToolStripStatusLabel.Text = "(Home vs. Away)(Attack Coordinate: (" + GameModel.alphaCoords(GridLock.Column) + "," + GridLock.Row + "))(Action: Awaiting Fire Result)"; } EDIT: if I put in a messageBox after the StripStatus message the status is updated.

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• Playing a .TS file on iOS

  - by Jonathan Grynspan

  We're working with some hardware that produces files in the .TS format, and we'd like to play them on an iOS device. (The files are internally consistent with what iOS supports--MPEG-4 video, AAC audio.) We've been investigating three options so far: Roll our own integrated HTTP Live Streaming server and serve up a faux M3U8 playlist from within the app. This... doesn't seem to want to play nice, and we've had mixed luck actually getting the .TS files to play on devices. Unwrap the MPEG-4 and AAC data from the TS file and re-wrap it as MP4. This, I'm told, is exceedingly difficult to do, and I haven't found anything useful online that could shed light on how to do it. We've got code in the pipeline to do it but it won't be ready until long after we need it. If we could do it, I could easily subclass NSURLProtocol and have it working within a matter of hours minutes. Use FFmpeg to implement option #2. FFmpeg seems like a possible solution but it isn't configured to build for iOS and I don't have the background to get it working (whereas the rest of our engineers don't have the Apple background needed.) I think #2 is our best bet, but as I don't know the ins and outs of MPEG-2 TS and MPEG-4, I don't have the ability to put it together myself. Does anybody have any insight into this problem? Perhaps some experience playing local TS files on iOS, or some tips on converting from TS to MP4?

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• How can I filter these Django records?

  - by mipadi

  I have a set of Django models as shown in the following diagram (the names of the reverse relationships are shown in the yellow bubbles): In each relationship, a Person may have 0 or more of the items. Additionally, the slug field is (unfortunately) not unique; multiple Person records may have the same slug fields. Essentially these records are duplicates. I want to obtain a list of all records that meet the following criteria: All duplicate records (that is, having the same slug) with at least one Entry OR at least one Audio OR at least one Episode OR at least one Article. So far, I have the following query: Person.objects.values('slug').annotate(num_records=Count('slug')).filter(num_records__gt=1) This groups all records by slug, then adds a num_records attribute that says how many records have that slug, but the additional filtering is not performed (and I don't even know if this would work right anyway, since, given a set of duplicate records, one may have, e.g., and Entry and the other may have an Article). In a nutshell, I want to find all duplicate records and collapse them, along with their associated models, into one record. What's the best way to do this with Django?

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• How do I begin reading source code?

  - by anonnoir

  I understand the value of reading source code, and I am trying my best to read as much as I can. However, every time I try getting into a 'large' (i.e. complete) project of sorts, I am overwhelmed. For example, I use Anki a lot when revising languages. Also, I'm interested in getting to know how an audio player works (because I have some project ideas), hence quodlibet on Google Code. But whenever I open the source code folders for the above programs, there are just so many files that I don't know where or what to begin with. I think that I should start with files marked init.py but I can't see the logical structure of the programs, or what reasoning was applied when the original writer divided his modules the way he did. Hence, my questions: How/where should I begin reading source? Any general tips or ideas? How does a programmer keep in mind the overall structure and logic of the program, especially for large projects, and is it common not to document that structure? As an open source reader, must I look through all of the code and get a bird's eye view of the code and libraries, before even being able to proceed? Would an IDE like Eclipse SDK (with PyDev) help with code-reading? Thanks for the help; I really appreciate your helping me.

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• Converting UnicodeString to PAnsiChar in Delphi XE

  - by moodforaday

  In Delphi XE I am using the BASS audio library, which contains this function: function BASS_StreamCreateURL(url: PAnsiChar; offset: DWORD; flags: DWORD; proc: DOWNLOADPROC; user: Pointer):HSTREAM; stdcall; external bassdll; The 'url' parameter is of type PAnsiChar, so in my code I do a cast: FStreamHandle := BASS_StreamCreateURL(PAnsiChar( url ) [...] The compiler emits a warning on this line: "suspicious typecast of string to PAnsiChar". In trying to eliminate the warning, I found that the recommended way is to use a double cast: FStreamHandle := BASS_StreamCreateURL(PAnsiChar( AnsiString( url )) [...] This does eliminate the warning, but the BASS function now returns error code 2 ("cannot open file"), which tells me the URL string it receives is somehow broken. I cannot see what the bass DLL actually receives, but using a breakpoint in the debugger the string looks good: var s : PAnsiChar; begin s := PAnsiChar( AnsiString( url )); At this point string s appears fine, but the BASS function fails when I pass it. My initial code: PAnsiChar( url ) works well with BASS, but emits a warning. So what's the correct way of getting from UnicodeString to PAnsiChar without a warning?

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• PlaySound linker error in C++

  - by logic-unit

  Hello, I'm getting this error: [Linker error] undefined reference to 'PlaySoundA@12' Id returned 1 exit status From this code: // c++ program to generate a random sequence of numbers then play corresponding audio files #include <windows.h> #include <mmsystem.h> #include <iostream> #pragma comment(lib, "winmm.lib") using namespace std; int main() { int i; i = 0; // set the value of i while (i <= 11) // set the loop to run 11 times { int number; number = rand() % 10 + 1; // generate a random number sequence // cycling through the numbers to find the right wav and play it if (number == 0) { PlaySound("0.wav", NULL, SND_FILENAME); // play the random number } else if (number == 1) { PlaySound("1.wav", NULL, SND_FILENAME); // play the random number } //else ifs repeat to 11... i++; // increment i } return 0; } I've tried absolute and relative paths for the wavs, the file size of them is under 1Mb each too. I've read another thread here on the subject: http://stackoverflow.com/questions/1565439/how-to-playsound-in-c As you may well have guessed this is my first C++ program, so my knowledge is limited with where to go next. I've tried pretty much every page Google has on the subject including MSDN usage page. Any ideas? Thanks in advance...

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• FFMpeg Error av_interleaved_write_frame():

  - by rajaneesh

  this my code . after running php code FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/usr --libdir=/usr/lib --shlibdir=/usr/lib --mandir=/usr/share/man --incdir=/usr/include --enable-libamr-nb --enable-libamr-wb --enable-libdirac --enable-libfaac --enable-libfaad --enable-libmp3lame --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-x11grab libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Nov 6 2009 19:05:03, gcc: 4.1.2 20080704 (Red Hat 4.1.2-46) Seems stream 0 codec frame rate differs from container frame rate: 50.00 (50/1) - 25.00 (25/1) Input #0, flv, from 'demo.flv': Duration: 00:00:30.83, start: 0.000000, bitrate: 546 kb/s Stream #0.0: Video: h264, yuv420p, 640x360 [PAR 1:1 DAR 16:9], 546 kb/s, 25 tbr, 1k tbn, 50 tbc Stream #0.1: Audio: aac, 44100 Hz, stereo, s16 Output #0, image2, to 'demo.jpg': Stream #0.0: Video: mjpeg, yuvj420p, 640x360 [PAR 1:1 DAR 16:9], q=2-31, 200 kb/s, 90k tbn, 1 tbc Stream mapping: Stream #0.0 - #0.0 Press [q] to stop encoding av_interleaved_write_frame(): I/O error occurred Usually that means that input file is truncated and/or corrupted.

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• Array output for option of command in bash script

  - by dewaforex

  Hi, Sorry for my bad english I'm stuck figure out with my bash script with array for option of command I make bash script to extract attachments from mkv file, and at the end merge again that attachments to mkv file after the video/audio has been encoding.. this is for extract attachment #find the total of attachment A=$(mkvmerge -i input.mkv | grep -i attachment | awk '{printf $3 "\n"}' | sed 's;\:;;' | awk 'END { print NR }') #extract it for (( i=1; i<=$A; i++ )) do font[${i}]="$(mkvmerge -i input.mkv | grep -i attachment | awk '{for (i=11; i <= NF; i++) printf($i"%c" , (i==NF)?ORS:OFS) }' | sed "s/'//g" | awk "NR==$i")" mkvextract attachments input.mkv $i:"${font[${i}]}" done And now for merge again the attachment for (( i=1; i<=$A; i++ )) do #seach for space between file name and and '\' before the space because some attachment has space in filename font1[${i}]=$(echo ${font[${i}]} | sed 's/ /\\ /g') #make option for add attachment attachment[${i}]=$"--attach-file ${font1[${i}]}" done mkvmerge -o output.mkv -d 1 -S test.mp4 sub.ass ${attachment[*]} The problem, still can't work for file name with space. When I tried echo the ${attachment[*]}, It's seem all right --attach-file Beach.ttf --attach-file Candara.ttf --attach-file CASUCM.TTF --attach-file Complete\ in\ Him.ttf --attach-file CURLZ_.TTF --attach-file Frostys\ Winterland.TTF --attach-file stilltim.ttf But the output still recognize the file name with space only the first word. mkvmerge v3.0.0 ('Hang up your Hang-Ups') built on Dec 6 2010 19:19:04 Automatic MIME type recognition for 'Beach.ttf': application/x-truetype-font Automatic MIME type recognition for 'Candara.ttf': application/x-truetype-font Automatic MIME type recognition for 'CASUCM.TTF': application/x-truetype-font Error: The file 'Complete\' cannot be attached because it does not exist or cannot be read. I hope somebody can help me. Thanks

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• Encoding h.264 with libavcodec/x264

  - by Leviathan

  I am attempting to encode video using libavcodec/libavformat. I'm trying to change the standard output-example.c from ffmpeg source. The AVI file is created on the disk, but the only sound is encoded. I tried adding a lot of options for x264 from here. All the other codecs works fine, mpeg2, mpeg4, mjpeg, xvid. In addition to specifying the parameters x264, I also set the codec to AVOutputFormat structure. That's all I've done. AVOutputFormat *pOutFormat; // in header file av_register_all(); AVCodec *codec = avcodec_find_encoder_by_name("libx264"); pOutFormat = guess_format("avi", NULL, NULL); pOutFormat->video_codec = codec->id; The debug output of my application: Output #0, mp4, to 'D:\1.avi': Stream #0.0: Video: libx264, yuv420p, 320x240, q=10-51, 500 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: aac, 44100 Hz, 1 channels, s16, 128 kb/s [libx264 @ 0x694010]using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x694010]bitrate tolerance too small, using .01 [libx264 @ 0x694010]profile Main, level 2.0 [libx264 @ 0x694010]frame I:150 Avg QP:14.76 size: 2534 [libx264 @ 0x694010]mb I I16..4: 75.9% 0.0% 24.1% [libx264 @ 0x694010]final ratefactor: 17.57 [libx264 @ 0x694010]coded y,uvDC,uvAC intra: 42.7% 92.4% 47.4% [libx264 @ 0x694010]i16 v,h,dc,p: 11% 14% 2% 73% [libx264 @ 0x694010]i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 18% 29% 5% 8% 10% 3% 3% 2% [libx264 @ 0x694010]kb/s:506.79

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• Getting nice sound from Java

  - by Peter Lang

  I managed to play midi files using Java, but it produces some distracting noise. I figured out that this is caused by the poor quality soundbank file shipped with Java 6 SDK/JRE. How can I improve that quality? Here is what I have so far: MidiNote example using a Receiver works fine (sounds the same as when playing midi files with other players), so it does not seem to use the Soundbank shipped with Java but the fallback mechanism that uses a hardware MIDI port. Using SimpleMidiPlayer example to play a Midi file works, but the quality is poor. When I delete lib/audio/soundbank.gm, the quality is not bad any more, so the fallback is used again. When I put soundbank-deluxe.gm into the same directory, it is used and produces much better sound. Messing with the clients soundbank file as described in the official Installation Instructions certainly isn't an option, so I tried to put the new soundbank-file into the jar-file and load it: Soundbank soundbank = MidiSystem.getSoundbank( getClass().getResourceAsStream("soundbank-deluxe.gm")); if(synthesizer.isSoundbankSupported(soundbank)) { System.out.println(synthesizer.loadAllInstruments(soundbank)); } This prints true, but the sound remains unchanged. What am I doing wrong loading the soundbank file? Can I force the hardware MIDI port to be used instead of the standard soundbank file?

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• Setting Position of source and listener has no effect

  - by Ben E

  Hi Guys, First time i've worked with OpenAL, and for the life of my i can't figure out why setting the position of the source doesn't have any effect on the sound. The sounds are in stero format, i've made sure i set the listener position, the sound is not realtive to the listener and OpenAL isn't giving out any error. Can anyone shed some light? Create Audio device ALenum result; mDevice = alcOpenDevice(NULL); if((result = alGetError()) != AL_NO_ERROR) { std::cerr << "Failed to create Device. " << GetALError(result) << std::endl; return; } mContext = alcCreateContext(mDevice, NULL); if((result = alGetError()) != AL_NO_ERROR) { std::cerr << "Failed to create Context. " << GetALError(result) << std::endl; return; } alcMakeContextCurrent(mContext); SoundListener::SetListenerPosition(0.0f, 0.0f, 0.0f); SoundListener::SetListenerOrientation(0.0f, 0.0f, -1.0f); The two listener functions call alListener3f(AL_POSITION, x, y, z); Real vec[6] = {x, y, z, 0.0f, 1.0f, 0.0f}; alListenerfv(AL_ORIENTATION, vec); I set the sources position to 1,0,0 which should be to the right of the listener but it has no effect alSource3f(mSourceHandle, AL_POSITION, x, y, z); Any guidance would be much appreciated

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• Linking AS code to symbols defined in an external SWC?

  - by Ender

  (apologies ahead of time, I only really know Flash; my Flex experience is basically nil. There may be a very standard and obvious workflow solution that Flex people know about) I have a number of UI elements that are graphically quite complex (they're not components, they're just Sprites). Since it takes a long time to compile them, I've been trying to move them into an external .swc. However, I want to associate some code with these classes, but I don't want to have to recompile the graphical assets every time I make a code change. At the moment I have it set up like this: UI elements are created in a separate FLA and exported to a SWC. In my primary FLA, I have actionscript classes that extend each of the graphical assets in the SWC. For example: external.swc: (some symbol defined in the Library and exported for actionscript in frame 1) class: com.foo.WidgetGraphic base: flash.display.Sprite main.fla: Widget.as: package com.foo { public class Widget extends WidgetGraphic { ... } } This works, but is time-consuming and prone to error. I'd rather be able to avoid having to inherit from each graphical asset, and just define them directly. Is there a better way to do what I'm trying to accomplish? Note: the main concern here is compile time. I don't have any movies or audio or fonts, just a lot of vector art assets that appear to be slowing down my compilation time significantly. When I'm debugging I'm only making code changes, and would rather not have to keep recompiling the art...

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