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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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• c# regex split and extract multiple parts from a string

  - by nLL

  Hi, I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) - 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to diffrent string objects instead of single

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• Why is cell phone software is still so primitive?

  - by Tomislav Nakic-Alfirevic

  I don't do mobile development, but it strikes me as odd that features like this aren't available by default on most phones: full text search: searches all address book contents, messages, anything else being a plus better call management: e.g. a rotating audio call log, meaning you always have the last N calls recorded for your listening pleasure later (your little girl just said her first "da-da" while you were on a business trip, you had a telephone job interview, you received complex instructions to do something etc.) bluetooth remote control (like e.g. anyRemote, but available by default on a bluetooth phone) no multitasking capabilities worth mentioning and in general no e.g. weekly software updates, making the phone much more usable (even if it had to be done over USB, rather than over the network). I'm sure I was dumbfounded by the lack or design of other features as well, but they don't come to mind right now. To clarify, I'm not talking about smartphones here: my plain, 2-year old phone has a CPU an order of magnitude faster than my first PC, about as much storage space and it's ridiculous how bad (slow, unwieldy) the software is and it's not one phone or one manufacturer. What keeps the (to me) obvious software functionality vacuum on a capable hardware platform from being filled up?

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• Best way to handle huge strings in c#

  - by srk

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

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• Django: Serving Media Behind Custom URL

  - by TheLizardKing

  So I of course know that serving static files through Django will send you straight to hell but I am confused on how to use a custom url to mask the true location of the file using Django. http://stackoverflow.com/questions/2681338/django-serving-a-download-in-a-generic-view but the answer I accepted seems to be the "wrong" way of doing things. urls.py: url(r'^song/(?P<song_id>\d+)/download/$', song_download, name='song_download'), views.py: def song_download(request, song_id): song = Song.objects.get(id=song_id) fsock = open(os.path.join(song.path, song.filename)) response = HttpResponse(fsock, mimetype='audio/mpeg') response['Content-Disposition'] = "attachment; filename=%s - %s.mp3" % (song.artist, song.title) return response This solution works perfectly but not perfectly enough it turns out. How can I avoid having a direct link to the mp3 while still serving through nginx/apache? EDIT 1 - ADDITIONAL INFO Currently I can get my files by using an address such as: http://www.example.com/music/song/1692/download/ But the above mentioned method is the devil's work. How can I accomplished what I get above while still making nginx/apache serve the media? Is this something that should be done at the webserver level? Some crazy mod_rewrite? http://static.example.com/music/Aphex%20Twin%20-%20Richard%20D.%20James%20(V0)/10%20Logon-Rock%20Witch.mp3

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• Ruby ICalendar Gem: How to get e-mail reminders working.

  - by Jenny

  I'm trying to work out how to use the icalendar ruby gem, found at: http://icalendar.rubyforge.org/ According to their tutorial, you do something like: cal.event.do # ...other event properties alarm do action "EMAIL" description "This is an event reminder" # email body (required) summary "Alarm notification" # email subject (required) attendees %w(mailto:[email protected] mailto:[email protected]) # one or more email recipients (required) add_attendee "mailto:[email protected]" remove_attendee "mailto:[email protected]" trigger "-PT15M" # 15 minutes before add_attach "ftp://host.com/novo-procs/felizano.exe", {"FMTTYPE" => "application/binary"} # email attachments (optional) end alarm do action "DISPLAY" # This line isn't necessary, it's the default summary "Alarm notification" trigger "-P1DT0H0M0S" # 1 day before end alarm do action "AUDIO" trigger "-PT15M" add_attach "Basso", {"VALUE" => ["URI"]} # only one attach allowed (optional) end So, I am doing something similar in my code. def schedule_event puts "Scheduling an event for " + self.title + " at " + self.start_time start = self.start_time endt = self.start_time title = self.title desc = self.description chan = self.channel.name # Create a calendar with an event (standard method) cal = Calendar.new cal.event do dtstart Program.convertToDate(start) dtend Program.convertToDate(endt) summary "Want to watch" + title + "on: " + chan + " at: " + start description desc klass "PRIVATE" alarm do action "EMAIL" description desc # email body (required) summary "Want to watch" + title + "on: " + chan + " at: " + start # email subject (required) attendees %w(mailto:[email protected]) # one or more email recipients (required) trigger "-PT25M" # 25 minutes before end end However, I never see any e-mail sent to my account... I have even tried hard coding the start times to be Time.now, and sending them out 0 minutes before, but no luck... Am I doing something glaringly wrong?

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• How to post a poll on the Facebook wall

  - by Bengt

  Hi, I'm trying to convert my poll app into a Facebook iframe app. My app is written in PHP and uses some Ajax calls to vote at a poll. In the application canvas everything is working fine, but of course I want to get the poll on the wall of a user too. Unfortunately I'm not able to find out how I can post a simple poll with some radio buttons for the options on the wall. I know how to publish images, text, audio files and links to the wall, but I have no idea how to publish my poll on the wall. And I don't just want to use links to vote, I want the user be able to choose a radio button. Does anyone have an idea how to do this or where to find information about doing this? I'm stuck there now for a while and it gets pretty frustrating. I'm using the new Graph API by the way. Or is this impossible? But I don't think so. Any help is appreciated. Bengt

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• DSP - Filtering frequencies using DFT

  - by Trap

  I'm trying to implement a DFT-based 8-band equalizer for the sole purpose of learning. To prove that my DFT implementation works I fed an audio signal, analyzed it and then resynthesized it again with no modifications made to the frequency spectrum. So far so good. I'm using the so-called 'standard way of calculating the DFT' which is by correlation. This method calculates the real and imaginary parts both N/2 + 1 samples in length. To attenuate a frequency I'm just doing: float atnFactor = 0.6; Re[k] *= atnFactor; Im[k] *= atnFactor; where 'k' is an index in the range 0 to N/2, but what I get after resynthesis is a slighty distorted signal, especially at low frequencies. The input signal sample rate is 44.1 khz and since I just want a 8-band equalizer I'm feeding the DFT 16 samples at a time so I have 8 frequency bins to play with. Can someone show me what I'm doing wrong? I tried to find info on this subject on the internet but couldn't find any. Thanks in advance.

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• INSERT INTO sql server error : invalid object name

  - by thormayer

  I have a problem with some statement on SQL SERVER the error I get is that I have an invalid object name 'TBL_VIDEOS' INSERT INTO TBL_VIDEOS ( TBL_VIDEOS.ID, TBL_VIDEOS.TITLE, TBL_VIDEOS.V_DESCRIPTION, TBL_VIDEOS.UPLOAD_DATE, TBL_VIDEOS.V_VIEWS, TBL_VIDEOS.USERNAME, TBL_VIDEOS.RATING, TBL_VIDEOS.V_SOURCE, TBL_VIDEOS.FLAG ) VALUES ('Z8MTRH3LmTVm', 'Why Creativity is the New Economy', 'Dr Richard Florida, one of the world&#39;s leading experts on economic competitiveness, demographic trends and cultural and technological innovation shows how developing the full human and creative capabilities of each individual, combined with institutional supports such as commercial innovation and new industry, will put us back on the path to economic and social prosperity. Listen to the podcast of the full event including audience Q&amp;A: http://www.thersa.org/events/audio-and-past-events/2012/why-creativity-is-the-new-economy Our events are made possible with the support of our Fellowship. Support us by donating or applying to become a Fellow. Donate: http://www.thersa.org/support-the-rsa Become a Fellow: http://www.thersa.org/fellowship/apply', CURRENT_TIMESTAMP, 0, 1, 0, 'http://www.youtube.com/watch?v=VPX7gowr2vE&feature=g-all-u' ,0) and I wonder what i've done wrong ? (btw, the error refer to line 1.. guess its the table name.. but it correct!

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• not able to register sip user on red5server, using red5phone

  - by sunil221

  I start the red5, and then i start red5phone i try to register sip user , details i provide are username = 999999 password = **** ip = asteriskserverip and i got --- Registering contact -- sip:[email protected]:5072 the right contact could be --- sip :99999@asteriskserverip this is the log: SipUserAgent - listen -> Init... Red5SIP register [SIPUser] register RegisterAgent: Registering contact <sip:[email protected]:5072> (it expires in 3600 secs) RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout RegisterAgent: Failed Registration stop try. Red5SIP Client leaving app 1 Red5SIP Client closing client 35C1B495-E084-1651-0C40-559437CAC7E1 Release ports: sip port 5072 audio port 3002 Release port number:5072 Release port number:3002 [SIPUser] close1 [SIPUser] hangup [SIPUser] closeStreams RTMPUser stopStream [SIPUser] unregister RegisterAgent: Unregistering contact <sip:[email protected]:5072> SipUserAgent - hangup -> Init... SipUserAgent - closeMediaApplication -> Init... [SIPUser] provider.halt RegisterAgent: Registration failure: No response from server. [SIPUser] SIP Registration failure Timeout please let me know if i am doing anything wrong. regards Sunil

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• How to use Festival Text To Speech C/C++ API

  - by Peeyush

  I want to use Festival TTS with my C++ programme. So i have downloaded all files form http://www.cstr.ed.ac.uk/downloads/festival/2.0.95/ then i start reading manual(http://www.cstr.ed.ac.uk/projects/festival/manual/festival_28.html) for C++ API but in manual they says that: In order to use Festival you must include festival/src/include/festival.h' which in turn will include the necessary other include files infestival/src/include' and speech_tools/include' you should ensure these are included in the include path for you your program. Also you will need to link your program withfestival/src/lib/libFestival.a', speech_tools/lib/libestools.a',speech_tools/lib/libestbase.a' and `speech_tools/lib/libeststring.a' as well as any other optional libraries such as net audio. " I am using UBUNTU 10.04(festival package is by default installed and i can use it form terminal by festival command) and GCC 4.4.3 but the problem is that i am new to GCC and i am not understanding which files i have to include in order to run my C++ code and i also don't know how to link libraries with my c++ code. So please tell me exactly which files i have to include and how to link with libraries if anyone already use festival tts with c++ then please post your code Thanks

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• Cross-Platform Language + GUI Toolkit for Prototyping Multimedia Applications

  - by msutherl

  I'm looking for a language + GUI toolkit for rapidly prototyping utility applications for multimedia installations. I've been working with Max/MSP/Jitter for many years, but I'd like to add a text-based language to my 'arsenal' for tasks apart from 'content production'. (When it comes to actual media synthesis, my choices are clear [SuperCollider + MSP for audio, Jitter + Quartz + openFrameworks for video]). I'm looking for something that maintains some of the advantages of Max, but is lower-level, faster, more cross-platfrom (Linux support), and text-based. Integration with powerful sound/video libraries is not a requirement. Some requirements: Cross-platform (at least OSX and Linux, Windows is a plus) Fast and easy cross-platform GUIs with no platform-specific modification GUI code separated from backend code as much as possible Good for interfacing with external serial devices (micro-controllers) Good network support (UDP/TCP) Good libraries for multi-media (video, sound, OSC) are a plus Asynchronous synchronous UNIX integration is a plus The options that come to mind: AS3/Flex (not a fan of AS3 or the idea of running in the Flash Player) openFrameworks (C++ framework, perhaps a bit too low level [looking for fast development time] and biased toward video work) Java w/ Processing libraries (like openFrameworks, just slower) Python + Qt (is Qt appropriate for rapid prototyping?) Python + Another GUI toolkit SuperCollider + Swing (yucky GUI development) Java w/ SWT Any other options? What do you recommend?

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• Cocos2d shake/accelerometer issue.

  - by Ryan Poolos

  So I a little backstory. I wanted to implement a particle effect and sound effect that both last about 3 sec or so when the user shakes their iDevice. But first issue arrived when the build in UIEvent for shakes refused to work. So I took the advice of a few Cocos veterans to just use some script to get "violent" accelerometer inputs as shakes. Worked great until now. The problem is that if you keep shaking it just stacks the particle and sounds over and over. Now this wouldn't be that big of a deal except it happens even if you are careful to try and not do so. So what I am hoping to do is disable the accelerometer when the particle effect/sound effect start and then reenable it as soon as they finish. Now I don't know if I should do this by schedule, NStimer, or some other function. I am open to ALL suggestions. here is my current "shake" code. - (void)accelerometer:(UIAccelerometer *)accelerometer didAccelerate:(UIAcceleration *)acceleration { const float violence = 1; static BOOL beenhere; BOOL shake = FALSE; if (beenhere) return; beenhere = TRUE; if (acceleration.x > violence * 1.5 || acceleration.x < (-1.5* violence)) shake = TRUE; if (acceleration.y > violence * 2 || acceleration.y < (-2 * violence)) shake = TRUE; if (acceleration.z > violence * 3 || acceleration.z < (-3 * violence)) shake = TRUE; if (shake) { id particleSystem = [CCParticleSystemQuad particleWithFile:@"particle.plist"]; [self addChild: particleSystem]; // Super simple Audio playback for sound effects! [[SimpleAudioEngine sharedEngine] playEffect:@"Sound.mp3"]; shake = FALSE; } beenhere = FALSE; }

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• How to get a fully transparent backbuffer in directx 9 without vista Desktop Window Manager

  - by flawlesslyfaulted

  I currently have an activex control that initiates a media (video/audio) framework another development group in my company developed and I am providing a window handle to that code. That handle is being used by their rendering plugin in the pipeline that uses Direct3d for rendering the video using that handle. I have seperate LPDIRECT3D9EX and LPDIRECT3DDEVICE9EX pointers that I initialize in my activex control. I am trying to clear a backbuffer to transparent and then use directx drawing primatives to draw on that backbuffer producing a transparent window with my drawing primatives over the streaming video on the directx surface below. It appears that clearing a device backbuffer with full alpha transparency is ignored by directx. d3ddev->Clear(0, NULL, D3DCLEAR_TARGET, D3DCOLOR_RGBA(0, 0, 1, 0 /*full alpha*/), 1.0f, 0); I can see the object I draw but they are drawn on top of a backbuffer that has the RGB color specified without the alpha value. The project linked (http://www.codeproject.com/KB/directx/umvistad3d.aspx) to in the stackoverflow question below does what I want but requires vista's Desktop Window Manager and won't work for XP. http://stackoverflow.com/questions/148275/how-do-i-draw-transparent-directx-content-in-a-transparent-window I have tried with D3DRS_ALPHABLENDENABLE true with configured blend with no avail. I have also tried to have pixels with full alpha values not rendered using D3DRS_ALPHATESTENABLE, D3DRS_ALPHAREF, and D3DRS_ALPHAFUNC setup but this doesn't work either. I have tried using ColorFill with alpha after retrieving the backbuffer with GetBackBuffer but this doesn't work either. (again only RGB is used) Finally I have tried creating a texture, selecting a surface, colorfilling that surface with a fully transparent alpha value, then loading that surface onto the backbuffer but only the RGB values appear to be used. I have checked the capabilities using the DXCapsViewer.exe and the D3DFMT_A8R8G8B8 backbuffer format that I am using for the backbuffer is valid so it can't be that. Has anyone gotten a transparent backbuffer in directx to work in XP?

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• regex split and extract multiple parts from a string

  - by nLL

  I am trying to extract some parts of the "Video:" line from below text. Seems stream 0 codec frame rate differs from container frame rate: 30000.00 (300 00/1) -> 14.93 (1000/67) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'C:\a.3gp': Metadata: major_brand : 3gp5 minor_version : 0 compatible_brands: 3gp5isom Duration: 00:00:45.82, start: 0.000000, bitrate: 357 kb/s Stream #0.0(und): Video: mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb /s, 14.93 fps, 14.93 tbr, 90k tbn, 30k tbc Stream #0.1(und): Audio: aac, 16000 Hz, mono, s16, 11 kb/s Stream #0.2(und): Data: mp4s / 0x7334706D, 0 kb/s Stream #0.3(und): Data: mp4s / 0x7334706D, 0 kb/s* This is an output from ffmpeg command line where i can get Video: part with private string ExtractVideoFormat(string rawInfo) { string v = string.Empty; Regex re = new Regex("[V|v]ideo:.*", RegexOptions.Compiled); Match m = re.Match(rawInfo); if (m.Success) { v = m.Value; } return v; } and result is mpeg4, yuv420p, 352x276 [PAR 1:1 DAR 88:69], 344 kb What i am trying to do is to somehow split that line and get mpeg4 yuv420p 352x276 [PAR 1:1 DAR 88:69] 344 kb assigned to different string objects instead of single

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• .NET Speech recognition plugin Runtime Error: Unhandled Exception. What could possibly cause it?

  - by manuel

  I'm writing a plugin (dll file) for speech recognition, and I'm creating a WinForm as its interface/dialog. When I run the plugin and click the 'Speak' to start the initialization, I get an unhandled exception. Here is a piece of the code: public ref class Dialog : public System::Windows::Forms::Form { public: SpeechRecognitionEngine^ sre; private: System::Void btnSpeak_Click(System::Object^ sender, System::EventArgs^ e) { Initialize(); } protected: void Initialize() { if (System::Threading::Thread::CurrentThread->GetApartmentState() != System::Threading::ApartmentState::STA) { throw gcnew InvalidOperationException("UI thread required"); } //create the recognition engine sre = gcnew SpeechRecognitionEngine(); //set our recognition engine to use the default audio device sre->SetInputToDefaultAudioDevice(); //create a new GrammarBuilder to specify which commands we want to use GrammarBuilder^ grammarBuilder = gcnew GrammarBuilder(); //append all the choices we want for commands. //we want to be able to move, stop, quit the game, and check for the cake. grammarBuilder->Append(gcnew Choices("play", "stop")); //create the Grammar from th GrammarBuilder Grammar^ customGrammar = gcnew Grammar(grammarBuilder); //unload any grammars from the recognition engine sre->UnloadAllGrammars(); //load our new Grammar sre->LoadGrammar(customGrammar); //add an event handler so we get events whenever the engine recognizes spoken commands sre->SpeechRecognized += gcnew EventHandler<SpeechRecognizedEventArgs^> (this, &Dialog::sre_SpeechRecognized); //set the recognition engine to keep running after recognizing a command. //if we had used RecognizeMode.Single, the engine would quite listening after //the first recognized command. sre->RecognizeAsync(RecognizeMode::Multiple); //this->init(); } void sre_SpeechRecognized(Object^ sender, SpeechRecognizedEventArgs^ e) { //simple check to see what the result of the recognition was if (e->Result->Text == "play") { MessageBox(plugin.hwndParent, L"play", 0, 0); } if (e->Result->Text == "stop") { MessageBox(plugin.hwndParent, L"stop", 0, 0); } } };

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• Toggling between instances of NiftyPlayer on a page - won't stop playing when hidden on IE

  - by Ashley

  Hi, i've got a page with links to MP3s, when the link is clicked I use javascript to show a small Flash player (NiftyPlayer) under the link. When a different link is clicked, the old player is hidden and the new player is revealed. The player auto-starts when the element is shown, and auto-stops when hidden - in Firefox. In IE it will only auto-start and NOT auto-stop. This is what I would like to solve. This is an example HTML with link and player <a href="Beat The Radar - Misunderstood What You Said.mp3" onclick="toggle_visibility('player662431');return false;" class="mp3caption">Misunderstood What You Said</a> <div id="player662431" class="playerhide"><embed src="http://www.xxx.com/shop/flash/player.swf?file=/mp3/Beat The Radar - Misunderstood What You Said.mp3&as=1" quality="high" bgcolor="#000000" width="161" height="13" name="niftyPlayer662431" align="" type="application/x-shockwave-flash" swLiveConnect="true" pluginspage="http://www.macromedia.com/go/getflashplayer"></embed> Here is the javascript (i've got jquery installed to let me hide all the open players on this page apart from the new one) function toggle_visibility(id) { $('.playerhide').hide(); var e = document.getElementById(id); e.style.display = 'block'; } I think what I need to do is start the player manually with javascript (rather than using the autostart as=1 function in the URL string) There is some javascript that comes with NiftyPlayer to allow this EG niftyplayer('niftyPlayer1').play() there is also a stop method. I need some help with javascript - how do I add this call to play into my toggle_visibility function (it has the same unique ID number added to the name of the player as the ID of the div that's being shown, but I don't know how to pull this ID number out of one thing and put it in another) I also would like to be able to do niftyplayer('niftyPlayer1').stop() to stop the audio of the previously running player. Is it possible to store the current ID number somewhere and call it back when needed? Thanks for the help, i'm a PHP programmer who needs some support with Javascript - I know what I want to achieve, just don't know the commands to do it! Thanks

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• .net real time stream processing - needed huge and fast RAM buffer

  - by mack369

  The application I'm developing communicates with an digital audio device, which is capable of sending 24 different voice streams at the same time. The device is connected via USB, using FTDI device (serial port emulator) and D2XX Drivers (basic COM driver is to slow to handle transfer of 4.5Mbit). Basically the application consist of 3 threads: Main thread - GUI, control, ect. Bus reader - in this thread data is continuously read from the device and saved to a file buffer (there is no logic in this thread) Data interpreter - this thread reads the data from file buffer, converts to samples, does simple sample processing and saves the samples to separate wav files. The reason why I used file buffer is that I wanted to be sure that I won't loose any samples. The application doesn't use recording all the time, so I've chosen this solution because it was safe. The application works fine, except that buffered wave file generator is pretty slow. For 24 parallel records of 1 minute, it takes about 4 minutes to complete the recording. I'm pretty sure that eliminating the use of hard drive in this process will increase the speed much. The second problem is that the file buffer is really heavy for long records and I can't clean this up until the end of data processing (it would slow down the process even more). For RAM buffer I need at lest 1GB to make it work properly. What is the best way to allocate such a big amount of memory in .NET? I'm going to use this memory in 2 threads so a fast synchronization mechanism needed. I'm thinking about a cycle buffer: one big array, the Bus Reader saves the data, the Data Interpreter reads it. What do you think about it? [edit] Now for buffering I'm using classes BinaryReader and BinaryWriter based on a file.

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• How to get the selected option value of a drop down box in PHP code

  - by Angeline Aarthi

  I have a dropdown box which lists a set of logos,like flower,butterfly etc. <p class="title1">Logo</p> <select name="logoMenu" class="select" size="7"> <?php foreach($logos as $logo):?> <option id="<?php echo $logo['Subproperty']['id'];?>" value="<?php echo $logo['Subproperty']['values'];?>"><?php echo $logo['Subproperty']['values'];?> </option> <?php endforeach;?> </select> Suppose If I select the logo 'Flower' from the drop down box, I want the flower pic to be displayed in a div.This is the div that I have to display the pictures. <div id="theme_logos" class="float_left spaceleft" style="display:none;"> <?php foreach($defaultLogos as $logo): //if($logo['Subproperty']['values']==clicked option value){?> <img height="50" width="50" src="/FormBuilder/app/webroot/img/themes/<?php echo $logo['Subproperty']['images'];?>" class="float_left user_profile_image user_profile_image" alt="Default50"/> <?php endforeach;?> </div> The problem with this code is that it displaya all the pictures found in the table. Because im My controller code, I give only the property id as that of 'Logo',but do not give which logo. $this->set('defaultLogos',$this->Subproperty->find('all',array('conditions'=>array('Subproperty.property_id'=>1,'Subproperty.values'=>"Flower")))); Here I have hard coded as 'flower' so that I get the flower picture alone.. If I select the logo from the drop down box, how to pass that selected value to the controller code? Or if I get the selected logo name thro' jquery,how to use that value in the if condition inside the for each loop? someone help me out with this.. I'm using CakePHP framework. $("#logoMenu option").click(function(){ selectedLogo=$(this).attr("value"); $('#subproperty_id').val($(this).attr("id")); if(selectedLogo=="Your logo"){ $("#themes_upload").show(); } else{ alert(selectedLogo); $("#themes_upload").hide(); $("#theme_logos").show(); } }); EDIT Now I have tried an ajax post where I pass the selected logo to the same action of the controller. I get the value when I alert the passed value in the success function of the ajax function. I the picture doesn't appear. $("#logoMenu option").click(function(){ selectedLogo=$(this).attr("value"); $('#subproperty_id').val($(this).attr("id")); if(selectedLogo=="Your logo"){ $("#themes_upload").show(); } else{ alert(selectedLogo); $.ajax({ type: "POST", url: "http://localhost/FormBuilder/index.php/themes/themes/", async: false, data: "selectedLogo="+selectedLogo, success: function(msg){ alert( "Data Saved: " + msg); } }); $("#themes_upload").hide(); $("#theme_logos").show(); } }); function themes(){ $this->set('themes',$this->Theme->find('all')); $logo=$this->params['form']['selectedLogo']; echo "logo:".$logo; $this->set('defaultLogos',$this->Subproperty->find('all',array('conditions'=>array('Subproperty.property_id'=>1,'Subproperty.values'=>$logo)))); } But when I trry to display the img in the page,it doesn't appear. Is it because the div show command is after the ajax request?

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• How to control the system volume using javascript

  - by Geetha

  Hi, I am using media player to play audio and video. I am creating own button to increase and decrease the volume of the media player. working fine too. Problem: Even after reaches 0% volume its audible. If the player volume increase the system volume also be increased. Is it possible. How to achieve this task. Control: <object id="mediaPlayer" classid="clsid:22D6F312-B0F6-11D0-94AB-0080C74C7E95" codebase="http://activex.microsoft.com/activex/controls/mplayer/en/nsmp2inf.cab#Version=5,1,52,701" height="1" standby="Loading Microsoft Windows Media Player components..." type="application/x-oleobject" width="1"> <param name="fileName" value="" /> <param name="animationatStart" value="true" /> <param name="transparentatStart" value="true" /> <param name="autoStart" value="true" /> <param name="showControls" value="true" /> <param name="volume" value="70" /> </object> Code: function decAudio() { if (document.mediaPlayer.Volume >= -1000) { var newVolume = document.mediaPlayer.Volume - 100; if (newVolume >= -1000) { document.mediaPlayer.Volume = document.mediaPlayer.Volume - 100; } else { document.mediaPlayer.Volume = -1000; } } }

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