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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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  • Java Simple Chat Box

    - by Allen
    I am trying to create a very simple chat window that simply has the ability to display some text, which i add to from time to time. However I get the following run time error when attempting to append text to the window: java.lang.ClassCastException: javax.swing.JViewport cannot be cast to javax.swing.JTextPane at ChatBox.getTextPane(ChatBox.java:41) at ChatBox.getDocument(ChatBox.java:45) at ChatBox.addMessage(ChatBox.java:50) at ImageTest2.main(ImageTest2.java:160) Here is the class to handle the basic operations: public class ChatBox extends JScrollPane { private Style style; public ChatBox() { StyleContext context = new StyleContext(); StyledDocument document = new DefaultStyledDocument(context); style = context.getStyle(StyleContext.DEFAULT_STYLE); StyleConstants.setAlignment(style, StyleConstants.ALIGN_LEFT); StyleConstants.setFontSize(style, 14); StyleConstants.setSpaceAbove(style, 4); StyleConstants.setSpaceBelow(style, 4); JTextPane textPane = new JTextPane(document); textPane.setEditable(false); this.add(textPane); } public JTextPane getTextPane() { return (JTextPane) this.getComponent(0); } public StyledDocument getDocument() { return (StyledDocument) getTextPane().getStyledDocument(); } public void addMessage(String speaker, String message) { String combinedMessage = speaker + ": " + message; StyledDocument document = getDocument(); try { document.insertString(document.getLength(), combinedMessage, style); } catch (BadLocationException badLocationException) { System.err.println("Oops"); } } } if there is a simpler way to do this, by all means let me know. I only need the text to be of a single font type, and uneditable by the user. Aside from that, I just need to be able to append text on the fly.

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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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  • 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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  • How to get Facebook-Feeds in Trillian without connecting to Facebook-Chat?

    - by Protron
    I installed the trial of Trillian 5.0 Build 22 (I don't know if the version 4 Astra is any different). I successfully added my Facebook account so I can see my facebook news on Trillian and works great. The problem is that I don't want to use the Facebook-chat, and if I disconnect the Facebook-chat the rest of the facebook stuff (news, events, groups, request) also disappears. I tried by setting the Facebook status to Away; and I'm not sure if someone can chat with me if I choose that; but anyway I still have the contacts list full of people I don't want. EDIT: Maybe I wasn't clear enough about the Facebook part. And that might be why some people fell this is off-topic. But when I talk about Facebook in this question, I'm not referring to the Facebook site at any point. I'm mean the Facebook plugin of Trillian (and it's not an user plugin; it's a built-in plugin).

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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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  • 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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  • PPPD Is Locking the Modem and Not Releasing It

    - by Skid
    Got an issue with PPPD on one of our system, we have a PC that is used to talk to remote sites via a dial up connection, the modem can both connect out to the sites and the sites also dial back in. Currently I'm having an issue where some times a site ether dials in or we dial out, and it connects, but then blocks the modem and throws and error to kern.log. Aug 26 14:23:57 TM-SCADA kernel: [191233.503745] INFO: task pppd:8142 blocked for more than 120 seconds. Aug 26 14:23:57 TM-SCADA kernel: [191233.503750] "echo 0 > /proc/sys/kernel/hung_task_timeout_secs" disables this message. Aug 26 14:23:57 TM-SCADA kernel: [191233.503753] pppd D ffffffff8180cb40 0 8142 1 0x00000000 Aug 26 14:23:57 TM-SCADA kernel: [191233.503759] ffff8800ac1f5dc8 0000000000000086 ffff8800ac1f5fd8 00000000000137c0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503765] ffff8800ac1f4010 00000000000137c0 00000000000137c0 00000000000137c0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503770] ffff8800ac1f5fd8 00000000000137c0 ffffffff81c13020 ffff880135df5b80 Aug 26 14:23:57 TM-SCADA kernel: [191233.503775] Call Trace: Aug 26 14:23:57 TM-SCADA kernel: [191233.503784] [<ffffffff8166ba29>] schedule+0x29/0x70 Aug 26 14:23:57 TM-SCADA kernel: [191233.503790] [<ffffffff813db005>] tty_ldisc_ref_wait+0x65/0xb0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503796] [<ffffffff813f3061>] ? uart_ioctl+0xd1/0x1c0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503801] [<ffffffff81076960>] ? wake_up_bit+0x40/0x40 Aug 26 14:23:57 TM-SCADA kernel: [191233.503806] [<ffffffff813d3fa0>] tty_ioctl+0x2c0/0x9a0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503810] [<ffffffff811d0549>] ? fcntl_setlk+0x69/0x200 Aug 26 14:23:57 TM-SCADA kernel: [191233.503815] [<ffffffff81195f79>] do_vfs_ioctl+0x99/0x330 Aug 26 14:23:57 TM-SCADA kernel: [191233.503820] [<ffffffff81195212>] ? do_fcntl+0x232/0x410 Aug 26 14:23:57 TM-SCADA kernel: [191233.503823] [<ffffffff811962b1>] sys_ioctl+0xa1/0xb0 Aug 26 14:23:57 TM-SCADA kernel: [191233.503829] [<ffffffff81674e69>] system_call_fastpath+0x16/0x1b The syslog trace stops at "Serial connection established". Aug 28 06:00:03 TM-SCADA pppd[10358]: pppd 2.4.5 started by root, uid 0 Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (NO CARRIER) Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (NO DIALTONE) Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (ERROR) Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (NO ANSWER) Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (BUSY) Aug 28 06:00:04 TM-SCADA chat[10360]: abort on (Username/Password Incorrect) Aug 28 06:00:04 TM-SCADA chat[10360]: send (atz^M) Aug 28 06:00:04 TM-SCADA chat[10360]: expect (OK) Aug 28 06:00:04 TM-SCADA chat[10360]: atz^M^M Aug 28 06:00:04 TM-SCADA chat[10360]: OK Aug 28 06:00:04 TM-SCADA chat[10360]: -- got it Aug 28 06:00:04 TM-SCADA chat[10360]: send (atx0^M) Aug 28 06:00:04 TM-SCADA chat[10360]: expect (OK) Aug 28 06:00:04 TM-SCADA chat[10360]: ^M Aug 28 06:00:04 TM-SCADA chat[10360]: atx0^M^M Aug 28 06:00:04 TM-SCADA chat[10360]: OK Aug 28 06:00:04 TM-SCADA chat[10360]: -- got it Aug 28 06:00:04 TM-SCADA chat[10360]: send (atdt0123456789^M) Aug 28 06:00:04 TM-SCADA chat[10360]: expect (CONNECT/ARQ) Aug 28 06:00:04 TM-SCADA chat[10360]: ^M Aug 28 06:00:30 TM-SCADA chat[10360]: atdt0123456789^M^M Aug 28 06:00:30 TM-SCADA chat[10360]: CONNECT/ARQ Aug 28 06:00:30 TM-SCADA chat[10360]: -- got it Aug 28 06:00:30 TM-SCADA pppd[10358]: Serial connection established. I've only found two ways to release the modem in this condition, the first is to turn the modem off and on again, the second is to delete the serial lock file, and then SIGKILL pppd. Now I could write into our software to do the latter if the modem is locked, but I would rather stop it from locking in the first place if at all possible. The reason I put this issue in the askubuntu is because we used to use OpenSuse and never had an issue with it, admittedly that was version 11.2 or earlier so its still and old kernel, but I figured I would ask here first anyway. Any suggestions of places to look would be appreciated.

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  • Realtek HD Audio 5.1 on Windows 7

    - by Darth
    I have problem with Realtek 5.1 driver for Windows 7 x64. I've installed newest drivers for Realtek HD Audio, but 5.1 still doesn't work, the only thing that works is front stereo. However, when I click on single speaker in sound settings test, every one of them work. Sorry that the image isn't in english, but I hope you can understand the point.

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  • How to change remote desktop audio quality?

    - by Vili
    When listening to some music on my remote computer, the audio quality is low. I've found this: HKLM\SOFTWARE\Microsoft\Windows NT\CurrentVersion\Drivers32\Terminal Server\RDP There are two interesting keys: EnableMP3Codec - with 0x00001 MaxBandwidth - this is 0x000056b9 Does anybody know anything about these keys? How should they work? What if i set EMP3C to 0? What if i change MaxBandwidth to higher number? Should i change these on my local computer and/or on the remote computer?

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  • Speech to text software (audio transcribing) for mac

    - by GiH
    What is the best speech to text software for mac? I have an hours worth of audio that I need to transcribe, and I'd really like to not have to do it manually :-). I prefer free options and I like open source so if there is a project I'd like to know. All answers are welcome though.

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  • Good Audio Splitter

    - by Jeremy White
    I need to get audio from my computer's headphone jack and push the output to 2 sets of speakers. I have tried using a cheap splitter from Fry's, but one set of speakers ends up acting as a microphone (!?) for the other set of speakers. What's the best way to split headphone output and get best quality with no interference on each set of speakers? I would, of course, also be interested in why the cheap splitter causes one set of speakers to start acting as a microphone.

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