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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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  • 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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  • 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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  • 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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  • 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 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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  • .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 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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  • LGPL library with plugins of varied licenses

    - by Chris
    Note: "Plugins" here refers to shared objects that are accessed via dlopen() and friends. I'm writing a library that I'm planning on releasing under the LGPL. Its functionality can be extended (supporting new audio file formats, specifically) through plugins. I'm planning on creating an exception to the LGPL for this library so that plugins can be released under any license. So far so good. I've written a number of plugins already, some of which use LGPL and some of which use GPL libraries. I'm wary of releasing them with the main library, however, due to licensing issues. The LGPL-based ones would generally be fine, but for my "any license" clause. Would distributing these LGPL-based plugins with the library require the consent of the other license holders to create this exception? Along the same lines, would the inclusion of GPL-based plugins with my library force the whole thing to go GPL? I could also release the plugins separately. The advantage, I presume, is that the plugins an d library will now not be distributed together, creating more separation. But this seems to be no different, really, in the end. Boiled down: Can I include, with my LGPL library, plugins of varied licenses? If not, is it really any different releasing them separately? And if so, there's no real need to create an exception for non-LGPL plugins, is there? It's LGPL or nothing. I'd prefer asking a lawyer, of course, but this is just a hobby and I can't afford to hire a lawyer when I don't expect or want monetary compensation. I'm just hoping others have been in similar situations and have insight.

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  • What can cause my code to run slower when the server JIT is activated?

    - by durandai
    I am doing some optimizations on an MPEG decoder. To ensure my optimizations aren't breaking anything I have a test suite that benchmarks the entire codebase (both optimized and original) as well as verifying that they both produce identical results (basically just feeding a couple of different streams through the decoder and crc32 the outputs). When using the "-server" option with the Sun 1.6.0_18, the test suite runs about 12% slower on the optimized version after warmup (in comparison to the default "-client" setting), while the original codebase gains a good boost running about twice as fast as in client mode. While at first this seemed to be simply a warmup issue to me, I added a loop to repeat the entire test suite multiple times. Then execution times become constant for each pass starting at the 3rd iteration of the test, still the optimized version stays 12% slower than in the client mode. I am also pretty sure its not a garbage collection issue, since the code involves absolutely no object allocations after startup. The code consists mainly of some bit manipulation operations (stream decoding) and lots of basic floating math (generating PCM audio). The only JDK classes involved are ByteArrayInputStream (feeds the stream to the test and excluding disk IO from the tests) and CRC32 (to verify the result). I also observed the same behaviour with Sun JDK 1.7.0_b98 (only that ist 15% instead of 12% there). Oh, and the tests were all done on the same machine (single core) with no other applications running (WinXP). While there is some inevitable variation on the measured execution times (using System.nanoTime btw), the variation between different test runs with the same settings never exceeded 2%, usually less than 1% (after warmup), so I conclude the effect is real and not purely induced by the measuring mechanism/machine. Are there any known coding patterns that perform worse on the server JIT? Failing that, what options are available to "peek" under the hood and observe what the JIT is doing there?

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  • Android SDK: hello world does not run

    - by Alex
    I have installed Java x64, Eclipse Classic Judo x64 + ADT Pluggin. OS win 7 x64. I did installation everything according to the manual. Then created first application and launched it. Emulator was launched but hello world was not. I have not idea what doing wrong. Do anyone knows of such error and my problem as a whole? thx Console log: [2012-10-06 13:35:42 - test] ------------------------------ [2012-10-06 13:35:42 - test] Android Launch! [2012-10-06 13:35:42 - test] adb is running normally. [2012-10-06 13:35:42 - test] Performing com.example.test.MainActivity activity launch [2012-10-06 13:35:42 - test] Automatic Target Mode: launching new emulator with compatible AVD 'AVD_41' [2012-10-06 13:35:42 - test] Launching a new emulator with Virtual Device 'AVD_41' [2012-10-06 13:35:42 - Emulator] Failed to create Context 0x3005 [2012-10-06 13:35:42 - Emulator] emulator: WARNING: Could not initialize OpenglES emulation, using software renderer. [2012-10-06 13:35:42 - Emulator] WARNING: Data partition already in use. Changes will not persist! [2012-10-06 13:35:42 - Emulator] WARNING: SD Card image already in use: C:\Users\Zewisa\.android\avd\AVD_41.avd/sdcard.img [2012-10-06 13:35:42 - Emulator] WARNING: Cache partition already in use. Changes will not persist! [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] could not get wglGetExtensionsStringARB [2012-10-06 13:35:42 - Emulator] emulator: warning: opening audio input failed [2012-10-06 13:35:42 - Emulator]

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  • reactivating or binding a hover function in jquery??

    - by mathiregister
    hi guys, with the following three lines: $( ".thumb" ).bind( "mousedown", function() { $('.thumb').not(this).unbind('mouseenter mouseleave'); }); i'm unbinding this hover-function: $(".thumb").hover( function () { $(this).not('.text, .file, .video, .audio').stop().animate({"height": full}, "fast"); $(this).css('z-index', z); z++; }, function () { $(this).stop().animate({"height": small}, "fast"); } ); i wonder how i can re-bind the exact same hover function again on mouseup? the follwoing three lines arent't working! $( ".thumb" ).bind( "mouseup", function() { $('.thumb').bind('mouseenter mouseleave'); }); to get what i wanna do here's a small explanation. I want to kind of deactivate the hover function for ALL .thumbs-elements when i click on one. So all (but not this) should not have the hover function assigned while i'm clicking on an object. If i release the mouse again, the hover function should work again like before. Is that even possible to do? thank you for your help!

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  • How to eliminate tearing from animation?

    - by MusiGenesis
    I'm running an animation in a WinForms app at 18.66666... frames per second (it's synced with music at 140 BPM, which is why the frame rate is weird). Each cel of the animation is pre-calculated, and the animation is driven by a high-resolution multimedia timer. The animation itself is smooth, but I am seeing a significant amount of "tearing", or artifacts that result from cels being caught partway through a screen refresh. When I take the set of cels rendered by my program and write them out to an AVI file, and then play the AVI file in Windows Media Player, I do not see any tearing at all. I assume that WMP plays the file smoothly because it uses DirectX (or something else) and is able to synchronize the rendering with the screen's refresh activity. It's not changing the frame rate, as the animation stays in sync with the audio. Is this why WMP is able to render the animation without tearing, or am I missing something? Is there any way I can use DirectX (or something else) in order to enable my program to be aware of where the current scan line is, and if so, is there any way I can use that information to eliminate tearing without actually using DirectX for displaying the cels? Or do I have to fully use DirectX for rendering in order to deal with this problem? Update: forgot a detail. My app renders each cell onto a PictureBox using Graphics.DrawImage. Is this significantly slower than using BitBlt, such that I might eliminate at least some of the tearing by using BitBlt?

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  • How do I post a .wav file from CS5 Flash, AS3 to a Java servlet?

    - by Muostar
    Hi, I am trying to send a byteArray from my .fla to my running tomcat server integrated in Eclipse. From flash I am using the following code: var loader:URLLoader = new URLLoader(); var header:URLRequestHeader = new URLRequestHeader("audio/wav", "application/octet-stream"); var request:URLRequest = new URLRequest("http://localhost:8080/pdp/Server?wav=" + tableID); request.requestHeaders.push(header); request.method = URLRequestMethod.POST; request.data = file;//wav; loader.load(request); And my java servlet looks as follows: try{ int readBytes = -1; int lengthOfBuffer = request.getContentLength(); InputStream input = request.getInputStream(); byte[] buffer = new byte[lengthOfBuffer]; ByteArrayOutputStream output = new ByteArrayOutputStream(lengthOfBuffer); while((readBytes = input.read(buffer, 0, lengthOfBuffer)) != -1) { output.write(buffer, 0, readBytes); } byte[] finalOutput = output.toByteArray(); input.close(); FileOutputStream fos = new FileOutputStream(getServletContext().getRealPath(""+"/temp/"+wav+".wav")); fos.write(finalOutput); fos.close(); When i run the flash .swf file and send the bytearray to the server, I receive following in the server's console window:: (loads of loads of Chinese symbols) May 20, 2010 7:04:57 PM org.apache.tomcat.util.http.Parameters processParameters WARNING: Parameters: Character decoding failed. Parameter '? (loads of loads of Chinese symbols) and then looping this for a long time. It is like I recieve the bytes but not encoding/decoding them correctly. What can I do?

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  • calculate camera up vector after glulookat()?

    - by carrots
    I'm just starting out teaching myself openGL and now adding openAL to the mix. I have some planets scattered around in 3D space and when I touch the screen, I'm assigning a sound to a random planet and then slowly and smoothly flying the "camera" over to look at it and listen to it. The animation/tweening part is working perfectly, but the openAL piece isn't quiet right. I move the camera around by doing a tiny translate() and gluLookAt() for every frame to keep things smooth (tweening the camera position and lookAt coords). The trouble seems to be with the stereo image I'm getting out of the headphones.. it seems like the left/right/up/down is mixed up sometimes after the camera rolls or spins. I am pretty sure the trouble is here: ALfloat listenerPos[]={camera->currentX,camera->currentY,camera->currentZ}; ALfloat listenerOri[]={camera->currentLookX, camera->currentLookY, camera->currentLookZ, 0.0,//Camera Up X <--- here 0.1,//Camera Up Y <--- here 0.0}//Camera Up Z <--- and here alListenerfv(AL_POSITION,listenerPos); alListenerfv(AL_ORIENTATION,listenerOri); I'm thinking I need to recompute the UP vector for the camera after each gluLookAt() to straighten out the audio positioning problem.. but after hours of googling and experimenting I'm stuck in math that suddenly got way over my head. 1) Am I right that I need to recalculate the up vector after each gluLookAt() i do? 2) If so, can someone please walk me though figuring out how to do that?

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