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  • No Audio Output Device is installed

    - by mabho
    Hi, this is an intermitent problem in my Sony Vaio model PCG-5K1L. I keep on getting a "No Audio Output Device is installed" when hovering my loudspeaker icon in Windows Vista. I have tried System Device Manager Sound Realtek High Definition Album Update Driver Software. The update process went through, but nothing happens. Still Vista does not seem to recognize my audio software. The strange part is that out of nothing my sound card can resume working to stop again hours later... If someone has any clues to solve this, please, help. Thanks a lot.

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  • Redirect audio from laptop to desktop over LAN

    - by Ram Rachum
    I want to be able to play a song on my laptop and have it sound through my desktop's (infinitely better) speakers. If you're familiar with Input Director: I want something that is to audio what Input Director is to mouse/keyboard. I want something that automatically redirects all audio from the laptop to the desktop in real time, and I want that solution to require, like Input Director, minimum maintenance. Beyond the initial setup, I don't want to have to babysit the program that does this. I want something that launches automatically with Windows and just works, and also allows me to cancel it whenever I want. And also doesn't go crazy when the laptop is turned on in a different network where the desktop computer isn't available. Any suggestions for such a program? (I use Windows XP on both computers.)

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  • Vmware Player 3.0 - cannot ping 32 bits guest from 64 bits (guest or host)

    - by npmj
    I'm stuck with what seems a bug in VmWare Player (build 203739). I'm using W7 Ultimate 64bits as host and have a CentOS 5.4 (64 bits) as a guest and a Windows XP Professional SP3 (32 bits) as another guest. From the 64 bits machines (the host and the linux guest) I cannot ping the windows XP. Off course, I already turned off the windows firewall in the guest and also in the host. The network is pretty basic, I'm using Vmnet8 (NAT), with DHCP and port forwarding (to the windows XP's IP). Everything is working ok, I have internet access from host and from both guests. Port forwarding to the XP guest is working ok too. The only problem is that I cannot access the XP guest through the Vmnet8. I monitored the traffic using wireshark (in the host and in the windows guest). If I try to ping the XP guest from the host, what I see is the ARP request leaving the host, being answered by the guest and, after that, there is no echo request leaving the host. The same occurs if I try to ping the XP from the CentOs guest. From the windows XP guest I can ping both the host and the CentOs guest. From the XP guest I can access the host shares. Obviously, from the host I cannot see the XP shares (as I cannot even ping the guest). I want to maintain this setup (using NAT to share the host's internet connection). Any suggestions?

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  • Shuffling in windows media player

    - by Crazy Buddy
    I think media player has several issues indeed. You see, I'll be hearing songs most of the time using WMP 11 (in WinXP SP3). Today - While I was wasting my time poking some sleepy questions in SE, I also noticed this... My "Now-playing" list contains some 500 mp3s (doesn't matter). I've enabled both Shuffle and Repeat. I play those songs. When I get irritated with some song (say - the 10th song), I change it. Something mysterious happened (happens even now). A sequence of atleast 3 songs (already played before the 10th song) repeat again in the same way following the selected one... Then, I skip those somehow and arrive at another boring song (say now - 20th) and now, the sequence would've increased by about 5 songs (sometimes)... Sometimes, I even notice a specific "sequence of songs" (including the skipped one) repeating again & again. I doubt most guys would've noticed. This makes me ask a question - Why? There are a lot songs in my playlist. Why the same sets of songs? Does WMP really chooses a sequence at start and follows it. Once a change is encountered, it starts the sequence again after several songs. Is it so? Feel free to shoot it down. I don't know whether it's acceptable here. Just curious about it... Note: This is only observed when both shuffle and repeat are enabled. To confirm, I tried it in two other PCs of mine (thereby dumped 2 hours). BTW, I also didn't observe this magic in VLC, Winamp, K-Lite and not even my Nokia cellphone. I think I'm not a good Googler and so, I can't find any such issues :-)

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  • How can I select an audio output device in directshow

    - by Vibhore Tanwer
    I was wondering how I can select the output device for audio in directshow. I am able to get available audio output devices in directshow. But how can I make one of these to be audio output device. Its always going for the default audio device. I want to be able to output audio on my choice of device. I have been struggling through google but couldn't find anything useful. All I could get was this link but it doesn't really solve my problem. Any help will be really helpful for me.

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  • Read the audio input level peak in Cocoa

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

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  • Chrome/Webkit audio tag bug?

    - by Ronald
    I'm trying to get HTML5's audio tag to work in Chrome. The following code works flawlessly in Firefox, any ideas why it isn't working in Webkit? <html> <head> <script type="text/javascript"> function init(){ audio = new Audio("chat.ogg"); audio.play(); } </script> </head> <body onload="init()"> </body> I should also note that I tried this with an mp3 as well. Regardless of what format, whenever .play() is called on audio, Chrome responds with "undefined".

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  • Traktor Audio 2 DJ soundcard configuration

    - by Jaroslav
    I have a Traktor Audio 2 DJ USB sound card (the first version of what it's now called simply Traktor Audio 2) The problem in settings it only sees one output, when there should be two (I need that for Mixxx etc.) Also I want to be able set the sample rate to one of these: 44.1, 48, 88.2, 96 kHz or at least check which one is set. Additionally if possible setting the latency would be an advantage. Some info: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: TraktorAudio2 [Traktor Audio 2], device 0: Traktor Audio 2 [Traktor Audio 2] Subdevices: 1/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ cat /proc/asound/cards 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfdcfc000 irq 45 1 [TraktorAudio2 ]: snd-usb-caiaq - Traktor Audio 2 Native Instruments Traktor Audio 2 (usb-0000:00:1d.7-8)

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  • VMware Player loses internet connectivity

    - by Martha
    Periodically, the internet simply stops working in my virtual machine, and the only way I can get it working again is to restart the host computer. Since I use the virtual machine specifically for testing web pages, this is, shall we say, a bother. Details: I have Windows XP Pro running in VMware Player (v. 3.0.0 build-203739) on a Windows 7 host. It's set to NAT (shared IP address) because the firewall won't allow a bridged connection. Every couple of days or so, first the internet slows down to a crawl, then eventually it stops working altogether. Both VMWare and the virtual OS report that they are connected, everything looks just peachy, I can reach the internet from the host, but on the VM, all web pages time out and/or report that the server could not be found. (Browser-independent; tried with IE, FF, Chrome, Safari, and Opera.) When this happens, the only way I've found to restore the internet connectivity is to restart the host machine. Restarting the VM doesn't help, nor does refreshing network connections on either the host or the guest. (Although I'm not entirely sure I've found the proper way to refresh a network connection in Windows 7...) I have not noticed any predictability about when the problem occurs, i.e. it's not immediately after I do anything special. It seems to occur mostly after putting the host to sleep once or twice, but it has happened even if the host has been in continuous use. It also seems independent of when I start using the VM - sometimes, I wake up the VM and the internet is really slow in it, then eventually stops working altogether; other times, I wake up the VM, use it perfectly happily for a while, then suddenly the internet is gone. Does anyone know why this is occurring? Failing that, is there a workaround that's less drastic than restarting the host? (Windows 7 startup times are blazingly fast compared to previous versions of Windows, but it's still a hassle to close all my programs and reopen them again.) Edit: while badges overall are nice, the Tumbleweed badge isn't helping me to solve my problem. Hasn't anyone encountered anything even remotely similar?

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  • moving audio over a local network using GStreamer

    - by James Turner
    I need to move realtime audio between two Linux machines, which are both running custom software (of mine) which builds on top of Gstreamer. (The software already has other communication between the machines, over a separate TCP-based protocol - I mention this in case having reliable out-of-band data makes a difference to the solution). The audio input will be a microphone / line-in on the sending machine, and normal audio output as the sink on the destination; alsasrc and alsasink are the most likely, though for testing I have been using the audiotestsrc instead of a real microphone. GStreamer offers a multitude of ways to move data round over networks - RTP, RTSP, GDP payloading, UDP and TCP servers, clients and sockets, and so on. There's also many examples on the web of streaming both audio and video - but none of them seem to work for me, in practice; either the destination pipeline fails to negotiate caps, or I hear a single packet and then the pipeline stalls, or the destination pipeline bails out immediately with no data available. In all cases, I'm testing on the command-line just gst-launch. No compression of the audio data is required - raw audio, or trivial WAV, uLaw or aLaw encoding is fine; what's more important is low-ish latency.

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  • Audio output from Silverlight

    - by leecarter
    I'm looking to develop a Silverlight application which will take a stream of data (not an audio stream as such) from a web server. The data stream would then be manipulated to give audio of a certain format (G.711 a-Law for example) which would then be converted into PCM so that additional effects can be applied (such as boosting the volume). I'm OK up to this point. I've got my data, converted the G.711 into PCM but my problem is being able to output this PCM audio to the sound card. I basing a solution on some C# code intended for a .Net application but in Silverlight there is a problem with trying to take a copy of a delegate (function pointer) which will be the topic of a separate question once I've produced a simple code sample. So, the question is... How can I output the PCM audio that I have held in a data structure (currently an array) in my Silverlight to the user? (Please don't say write the byte values to a text box) If it were a MP3 or WMA file I would play it using a MediaElement but I don't want to have to make it into a file as this would put a crimp on applying dynamic effects to the audio. I've seen a few posts from people saying low level audio support is poor/non-existant in Silverlight so I'm open to any suggestions/ideas people may have.

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  • Crash in audio resampler with some audio rates - FFMPEG PHP ( Solved! )

    - by Olaf Erlandsen
    i have a problem with this command( FFMPEG PHP ): Command: ffmpeg -i 62f76f050494f0ed6a5997967c00c0c0.wmv -ss 0 -t 99 -y -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 -f flv 62f76f050494f0ed6a5997967c00c0c0.flv Output: FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [asf @ 0xe81670]max_analyze_duration reached Input #0, asf, from '/var/www/resources/tmp/62f76f050494f0ed6a5997967c00c0c0.wmv': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 Duration: 00:00:50.87, bitrate: 2467 kb/s Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 256 kb/s Stream #0.1: Video: vc1, yuv420p, 950x460 [PAR 1:1 DAR 95:46], 25 fps, 25 tbr, 1k tbn, 25 tbc Output #0, flv, to '/var/www/resources/media/62f76f050494f0ed6a5997967c00c0c0.flv': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: flv, yuv420p, 950x460 [PAR 1:1 DAR 95:46], q=2-31, 200 kb/s, 1k tbn, 29.97 tbc Stream #0.1: Audio: libmp3lame, 11025 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 72 fps= 0 q=5.0 size= 0kB time=10.91 bitrate= 0.0kbits/s Multiple frames in a packet from stream 0 Warning, using s16 intermediate sample format for resampling frame= 141 fps=139 q=5.0 size= 103kB time=8.15 bitrate= 103.2kbits/s frame= 220 fps=144 q=5.0 size= 875kB time=10.92 bitrate= 656.6kbits/s frame= 290 fps=143 q=5.0 size= 1525kB time=13.74 bitrate= 909.1kbits/s frame= 356 fps=141 q=5.0 size= 2153kB time=15.99 bitrate=1103.1kbits/s frame= 427 fps=141 q=5.0 size= 2847kB time=18.70 bitrate=1247.0kbits/s frame= 497 fps=141 q=5.0 size= 3771kB time=21.16 bitrate=1460.0kbits/s frame= 575 fps=142 q=5.0 size= 4695kB time=24.61 bitrate=1563.0kbits/s frame= 639 fps=141 q=5.0 size= 5301kB time=26.80 bitrate=1620.2kbits/s frame= 703 fps=139 q=5.0 size= 5829kB time=29.36 bitrate=1626.2kbits/s frame= 774 fps=139 q=5.0 size= 6659kB time=32.39 bitrate=1684.0kbits/s frame= 842 fps=139 q=5.0 size= 7915kB time=35.27 bitrate=1838.6kbits/s frame= 911 fps=139 q=5.0 size= 9011kB time=37.98 bitrate=1943.4kbits/s frame= 975 fps=138 q=5.0 size= 9788kB time=40.59 bitrate=1975.3kbits/s frame= 1041 fps=138 q=5.0 size= 10904kB time=43.83 bitrate=2037.9kbits/s frame= 1115 fps=138 q=5.0 size= 11795kB time=46.24 bitrate=2089.8kbits/s frame= 1183 fps=138 q=5.0 size= 12678kB time=48.74 bitrate=2130.7kbits/s frame= 1247 fps=137 q=5.0 size= 13964kB time=51.36 bitrate=2227.5kbits/s frame= 1271 fps=136 q=5.0 Lsize= 15865kB time=58.86 bitrate=2208.1kbits/s video:15366kB audio:462kB global headers:0kB muxing overhead 0.238956% Problem: Warning, using s16 intermediate sample format for resampling I've also tried changing the parameter From -ar 44100 to -ar 11025 Thanks! Solution: Read this link: http://en.wikipedia.org/wiki/MP3#Bit_rate

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  • Is there a media player that works on HTTPS sites?

    - by Iain Hallam
    I'm currently using Yahoo! Media Player for a site that needs to play MP3 files that are stored on our server. In total, there's quite a bit more than the free limits at Soundcloud, but each file is only a few minutes long. YMP is pretty good, but causes security warnings on HTTPS pages, because it can only be served via HTTP. Is there an equivalent free player I can embed for the HTTPS pages? EDIT: Just to clarify, I'm initially looking for something that will scan the page and turn media links playable.

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  • How can I make Banshee re-encode FLAC to Ogg Vorbis when copying to my player?

    - by Michael E
    I have most of my music in FLAC on my large storage device, and would like to automatically re-encode it in Ogg Vorbis when copying it to my portable audio player (Sansa Fuze v2). I have set my Fuze to MTP mode and told Banshee to encode to Ogg Vorbis with quality 4 in the Device Properties dialog for the Fuze (I would use MSC mode, but don't have an encoding option in the device properties when I do that). However, when I copy music to the device, either by dragging it from the music library or by syncing a playlist, the full FLAC files are copied rather than transcoded and written as Oggs. How can I get my Banshee setup re-encoding the audio? If StackExchange supported bonus points, I'd give bonus points for a solution that only re-encoded music that was already losslessly encoded, but I don't think that's possible.

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  • No audio in my ubuntu system

    - by hap497
    Hi, I am running ubuntu 9.10. But there is no sound in my environment. When I go to System-Preference, there is no 'sound' entry there. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: I82801AAICH [Intel 82801AA-ICH], device 0: Intel ICH [Intel 82801AA-ICH] Subdevices: 1/1 Subdevice #0: subdevice #0 $ lsmod Module Size Used by usb_storage 52576 3 binfmt_misc 8356 1 vboxvfs 34620 0 vboxvideo 1884 1 drm 159584 2 vboxvideo agpgart 34988 1 drm snd_intel8x0 30168 2 snd_ac97_codec 101216 1 snd_intel8x0 ac97_bus 1532 1 snd_ac97_codec snd_pcm_oss 37920 0 snd_mixer_oss 16028 1 snd_pcm_oss snd_pcm 75296 3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss snd_seq_dummy 2656 0 snd_seq_oss 28576 0 iptable_filter 3100 0 snd_seq_midi 6432 0 ip_tables 11692 1 iptable_filter x_tables 16544 1 ip_tables snd_rawmidi 22208 1 snd_seq_midi snd_seq_midi_event 6940 2 snd_seq_oss,snd_seq_midi ppdev 6688 0 snd_seq 50224 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_mid i_event snd_timer 22276 2 snd_pcm,snd_seq snd_seq_device 6920 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi ,snd_seq psmouse 56500 0 serio_raw 5280 0 snd 59204 14 snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_ oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_ti mer,snd_seq_device i2c_piix4 9932 0 parport_pc 31940 0 soundcore 7264 1 snd snd_page_alloc 9156 2 snd_intel8x0,snd_pcm vboxguest 143836 7 vboxvfs lp 8964 0 parport 35340 3 ppdev,parport_pc,lp pcnet32 32644 0 mii 5212 1 pcnet32 floppy 54916 0 ~:987:2$ lspci 00:00.0 Host bridge: Intel Corporation 440FX - 82441FX PMC [Natoma] (rev 02) 00:01.0 ISA bridge: Intel Corporation 82371SB PIIX3 ISA [Natoma/Triton II] 00:01.1 IDE interface: Intel Corporation 82371AB/EB/MB PIIX4 IDE (rev 01) 00:02.0 VGA compatible controller: InnoTek Systemberatung GmbH VirtualBox Graphics Adapter 00:03.0 Ethernet controller: Advanced Micro Devices [AMD] 79c970 [PCnet32 LANCE] (rev 40) 00:04.0 System peripheral: InnoTek Systemberatung GmbH VirtualBox Guest Service 00:05.0 Multimedia audio controller: Intel Corporation 82801AA AC'97 Audio Controller (rev 01) 00:06.0 USB Controller: Apple Computer Inc. KeyLargo/Intrepid USB 00:07.0 Bridge: Intel Corporation 82371AB/EB/MB PIIX4 ACPI (rev 0 00:0b.0 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB2 EHCI Controller ~:988:3$

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  • Optical Audio out stuck on on a MacBook

    - by Clinton Blackmore
    Apple have made an interesting headphone port for the MacBook (and some other Intel Mac models). It works like a standard jack: nothing plugged in - audio comes out of built-in speakers headphones/external speakers plugged in - plays through headphones/external speakers but you can also use a special adapter (which trips a tiny microswitch) to get an optical audio out signal (which you can presumably plug into a nice surround-sound system). This is all well and good except when, like auto-tracking, it doesn't work, and you are left with nothing to adjust. Users report that they get no sound when they have nothing plugged in and that a red light emanates from the headphone port. If you go to System Preferences - Sound - Output, it will say (IIRC) "Optical Out" instead of "Internal Speakers". The only solution I'm aware of is to try to reset the switch by inserting and removing a set of headphones or a toothpick, perhaps wiggling it inside of the port, and hoping that you luck out and get it. Are there other ways to fix this problem? Does anyone know where the microswitch is or have a good technique to reset it?

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  • Ways to have audio output without wires

    - by viraptor
    I'm trying to find a way of using my home speakers/amp without actually having to connect them. There are two laptops that use them normally (so I don't like changing the connection all the time) and I'd rather move the speakers to a place that's away from the couch. I'm not sure how to do this though... The options I can think of are: some kind of wireless jack-jack connection finally getting a media server Unfortunately I can't find any good product for the first solution. I've seen some headphones which have the receiver integrated and a separate transmitted, so in general the idea is already out there, just not the way I need ;) I've seen also http://www.miccus.com/products/blubridge-mini-jack, but I'd have to have a compatible receiver which I can't find on its own (maybe there's some application that the media server could use?). As far as media server goes... many of the plug servers look really interesting, but I'm not sure how to create an audio output and how to redirect the input really. None of the plug servers I've seen so far advertises the option of audio output jack port. I think this part could be fixed by getting one with an usb port and a separate cheap usb soundcard. I hope that input can be sorted out in some rather simple way. I've got Linux running on both laptops so I hope that would be possible to configure jack/pulse/whatever to use the remote endpoint, or even write a simple local-/dev/dsp:network:media-server-/dev/dsp forwarder. So the main question is... are there better ways? Are there any out of the box solutions? Or maybe this was already done by someone and described somewhere?

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  • I can play "test" sounds, but no other audio works

    - by Callum
    I'm running Windows XP, and last night my PC was infected by a frustrating virus (one of those viruses that won't let you open virus checkers, etc). I finally killed it 2 hours later, but it involved some heavy duty anti-dote. One side effect is my audio is now gone. Except it's not entirely gone, because when I open the Realtek HD Audio Manager in the task bar, I can play all the "test" sounds. The speakers, the sound card, etc, are therefore working fine. But things like YouTube or Windows Media Player, there's no sound. I'm guessing there's a setting that needs to be reconfigured somewhere.. but where? Maybe relevant: One thing I did do last night was "play" with the system registry. Any help would be greatly appreciated. Thanks. SOLVED! The two hour battle with my computer virus resulted in my computer permanently thinking it was in Safe Mode, regardless of how it booted up. I was able to "fix" this by following the post by hsandler in this thread: http://www.petri.co.il/forums/showthread.php?t=23032&page=2 I then rebooted.. and let me tell you, the Windows Startup music has never sounded so sweet. Thanks to all, especially James, whose advice gave me a major clue as to what the problem was.

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  • How to slow down audio files?

    - by verve
    I need a program (with an easy learning curve) that lets me slow down mp3 (at the very least this format) music and audiobook files. The software needs to be able to slow down the audio at the chosen speeds without altering the pitch and accuracy of the words being pronounced. Perhaps like the language software "Byki Deluxe's" "SlowSound" feature? I'm learning a foreign language (German) and I find the speeds at which the books are being read too fast. I need to hear the pronunciation of each word much more clearly to learn how to pronounce the words myself. Is there such a product out there? Now, I know you can slow down stuff in VLC but it sounds really artificial. I need something that slows down audio files without altering the accuracy of the words being pronounced. It doesn't have to be freeware; ease of use and quality is more important to me. Win 7 64-bit. IE 8. Edit: Are there any software-for-pay like Audacity? Only the beta works in Win 7. Also, I'd prefer to be able to slow down a file live and not have to create a new file to use the feature.

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  • Converting Audio To Video Output and Attaching Text?

    - by ZeeMan
    I am currently working on a project and before i get started i thought it'd be nice to check with stackOverflow community, and see maybe they can help me with this. The Idea: I have about a thousand MP3 files that i need to convert into Video files to be upload on Youtube for my work. Here is where it gets tricky i need to also attach the Text associated with the Audio to the Video as an Image. I was thinking .ppt. The Problem: I can do this one audio file at a time but it would take me a zillion years. lol!! The Question: Can I Create Some Kind Of Program Using Let's Say XML or JavaScript Or XHTML or some other programming language to do a MASS content creation and all i have to do is feed it the Information?? possibly a script?? or is it possible to create an example .ppt file and then hack it so that i can have it reproduce itself with different information?? The Note: Thanks U In Advance For Helping Out!!! Regards, ZeeMan!!!

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  • Collision Detection on floor tiles Isometric game

    - by Anivrom
    I am having a very hard to time figuring out a bug in my code. It should have taken me 20 minutes but instead I've been working on it for over 12 hours. I am writing a isometric tile based game where the characters can walk freely amongst the tiles, but not be able to cross over to certain tiles that have a collides flag. Sounds easy enough, just check ahead of where the player is going to move using a Screen Coordinates to Tile method and check the tiles array using our returned xy indexes to see if its collidable or not. if its not, then don't move the character. The problem I'm having is my Screen to Tile method isn't spitting out the proper X,Y tile indexes. This method works flawlessly for selecting tiles with the mouse. NOTE: My X tiles go from left to right, and my Y tiles go from up to down. Reversed from some examples on the net. Here's the relevant code: public Vector2 ScreentoTile(Vector2 screenPoint) { //Vector2 is just a object with x and y float properties //camOffsetX,Y are my camera values that I use to shift everything but the //current camera target when the target moves //tilescale = 128, screenheight = 480, the -46 offset is to center // vertically + 16 px for some extra gfx in my tile png Vector2 tileIndex = new Vector2(-1,-1); screenPoint.x -= camOffsetX; screenPoint.y = screenHeight - screenPoint.y - camOffsetY - 46; tileIndex.x = (screenPoint.x / tileScale) + (screenPoint.y / (tileScale / 2)); tileIndex.y = (screenPoint.x / tileScale) - (screenPoint.y / (tileScale / 2)); return tileIndex; } The method that calls this code is: private void checkTileTouched () { if (Gdx.input.justTouched()) { if (last.x >= 0 && last.x < levelWidth && last.y >= 0 && last.y < levelHeight) { if (lastSelectedTile != null) lastSelectedTile.setColor(1, 1, 1, 1); Sprite sprite = levelTiles[(int) last.x][(int) last.y].sprite; sprite.setColor(0, 0.3f, 0, 1); lastSelectedTile = sprite; } } if (touchDown) { float moveX=0,moveY=0; Vector2 pos = new Vector2(); if (player.direction == direction_left) { moveX = -(player.moveSpeed); moveY = -(player.moveSpeed / 2); Gdx.app.log("Movement", String.valueOf("left")); } else if (player.direction == direction_upleft) { moveX = -(player.moveSpeed); moveY = 0; Gdx.app.log("Movement", String.valueOf("upleft")); } else if (player.direction == direction_up) { moveX = -(player.moveSpeed); moveY = player.moveSpeed / 2; Gdx.app.log("Movement", String.valueOf("up")); } else if (player.direction == direction_upright) { moveX = 0; moveY = player.moveSpeed; Gdx.app.log("Movement", String.valueOf("upright")); } else if (player.direction == direction_right) { moveX = player.moveSpeed; moveY = player.moveSpeed / 2; Gdx.app.log("Movement", String.valueOf("right")); } else if (player.direction == direction_downright) { moveX = player.moveSpeed; moveY = 0; Gdx.app.log("Movement", String.valueOf("downright")); } else if (player.direction == direction_down) { moveX = player.moveSpeed; moveY = -(player.moveSpeed / 2); Gdx.app.log("Movement", String.valueOf("down")); } else if (player.direction == direction_downleft) { moveX = 0; moveY = -(player.moveSpeed); Gdx.app.log("Movement", String.valueOf("downleft")); } //Player.moveSpeed is 1 //tileObjects.x is drawn in the center of the screen (400px,240px) // the sprite width is 64, height is 128 testX = moveX * 10; testY = moveY * 10; testX += tileObjects.get(player.zIndex).x + tileObjects.get(player.zIndex).sprite.getWidth() / 2; testY += tileObjects.get(player.zIndex).y + tileObjects.get(player.zIndex).sprite.getHeight() / 2; moveX += tileObjects.get(player.zIndex).x + tileObjects.get(player.zIndex).sprite.getWidth() / 2; moveY += tileObjects.get(player.zIndex).y + tileObjects.get(player.zIndex).sprite.getHeight() / 2; pos = ScreentoTile(new Vector2(moveX,moveY)); Vector2 pos2 = ScreentoTile(new Vector2(testX,testY)); if (!levelTiles[(int) pos2.x][(int) pos2.y].collides) { Vector2 newPlayerPos = ScreentoTile(new Vector2(moveX,moveY)); CenterOnCoord(moveX,moveY); player.tileX = (int)newPlayerPos.x; player.tileY = (int)newPlayerPos.y; } } } When the player is moving to the left (downleft-ish from the viewers point of view), my Pos2 X values decrease as expected but pos2 isnt checking ahead on the x tiles, it is checking ahead on the Y tiles(as if we were moving DOWN, not left), and vice versa, if the player moves down, it will check ahead on the X values (as if we are moving LEFT, instead of DOWN). instead of the Y values. I understand this is probably the most confusing and horribly written post ever, but I'm confused myself so I'm having a hard time explaining it to others lol. if you need more information please ask!! I'm so frustrated after over 12 hours of working on it I'm about to give up.

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  • Should I make the Cells in a Tiledmap as null when my player hits it

    - by Vishal Kumar
    I am making a Tile Based game using Libgdx. I took the idea from SuperKoalio platformer demo by Mario Zencher. When I wanted to implement Collectables in my game , I simply draw the coins using Tiled Map Editor. When my player hits that, I use to set that cell as null. Someday on this site suggested me not to do so... never use null. I agreed. What can be any other way. If I am using layer.setCell(x,y) to set the cell to any other cell... even if an transparent one .. my player seems to be stopped by an invisible object/hurdle. This is my code: for (Rectangle tile : tiles) { if (koalaRect.overlaps(tile)) { TiledMapTileLayer layer = (TiledMapTileLayer) map.getLayers().get(1); try{ type = layer.getCell((int) tile.x, (int) tile.y).getTile().getProperties().get("tileType").toString(); } catch(Exception e){ System.out.print("Exception in Tiles Property"+e); type="nonbreakable"; } //Let us destroy this cell if(("award".equals(type))){ layer.setCell((int) tile.x, (int) tile.y, null); listener.coin(); score+=100; test = ""+layer.getCell(0, 0).getTile().getProperties().get("tileType"); } //DOING THIS GIVES A BAD EFFECT if(("killer".equals(type))){ //player.health--; //layer.setCell((int) tile.x, (int) tile.y, layer.getCell(20,0)); } // we actually reset the player y-position here // so it is just below/above the tile we collided with // this removes bouncing :) if (player.velocity.y > 0) { player.position.y = (tile.y - Player.height); } Is this a right approach? OR I should create separate Sprite Class called Coin.

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  • audio and video data in RTP

    - by Banana
    Suppose a user wants to transmit both audio and video to another user, whose formats are AMR for audio and H.264 for video. Does the user have to transmit audio and video packets always separately? Meaning that it is not possible to mix audio and video within the same RTP packed, is that correct? If this is true I guess the RTP protocol will need to know the SSRC of both audio and video to be able to check the sync of the two streams.

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  • How to install audio-recorder

    - by Michael
    I have used Ubuntu serval years, and i am trying to install a audio recorder from the terminal, and this i want to work whit ubuntu as default audio recording system in the sound settings menu, and i installed it from the terminal and i had enter: sudo add-apt-repository ppa:osmoma/audio-recorder sudo apt-get update sudo apt-get install audio-recorder and it seams installed but how can you set it up as default audio recorder for ubuntu. Can some one please help.

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