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  • Reduce size of MP4

    - by testing
    I have a MP4 file with a lengh of 22:44. Here are the details: Video: width: 720 px height: 404 px data bitrate: 1022 kBit/s overall bitrate: 1182 kBit/s fps: 24 codec: H264 - MPEG4 AVC (part 10) (avc1) Audio: bitrate: 159 kBit/s stereo sample rate: 48 kHz codec: MPEG AAC Audio (mp4a) I thought I can reduce the current filesize (about 200 MB) by reducing the width and the height (420 x 236). I tried different programs: Handbrake, Format Factory, Next Video Converter and Super. The first three didn't worked as expected: Handbrake has a bug by setting the width and the height, the next two doesn't allow the fine setting of the videosize (only presets of width and height). Super seems to be the best, but I didn't found a setting which reduces the file size. I reduced the width and the height but only got 20 MB less. Now I tried the xth setting and I still get a too high file size. I want to reduce the filesize to 100 MB or less. The ouput format should be FLV or MP4, because I need this for flowplayer. Which settings of SUPER or which program should I use to reduce the file size? Of course the video should still be viewable.

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  • How do I install Revenge of the Titans?

    - by Akash
    I've downloaded the .deb file of Revenge of the Titans, and installed it using Ubuntu Software Center. Now, when I try to launch it using the software launcher nothing happens. Any ideas? The .deb file was downloaded from the Humble Indie Bundle. I am unable to launch it from the terminal ( the command revenge-of-the-titans says command not found ). I also tried the .tar.gz. When I extract it and run ./revenge.sh , nothing happens. No output on the terminal or anything at all. I have set chmod 777 revenge.sh as well. The command /opt/revengeofthetitans/revenge.sh does not give any output. If I run gedit /opt/revengeofthetitans/revenge.sh in the terminal: > #!/bin/bash > # > # revenge.sh > # > ############################################################################### > > SCRIPT="`basename $0`" > GAMEDIR="${HOME}/.revenge_of_the_titans_1.80" LOGFILE="${GAMEDIR}/${SCRIPT}.log" > INSTDIR="`dirname $0`" ; cd > "${INSTDIR}" ; INSTDIR="`pwd`" > > [[ ! -d "${GAMEDIR}" ]] && mkdir -m > 0755 "${GAMEDIR}" > > JARPATH="patch.jar:RevengeOfTheTitans.jar:data-hib.jar:gfx.jar:fonts.jar:images.jar:music.jar:fx-mono.jar:fx-stereo.jar:gamecommerce.jar:common.jar:spgl-lite.jar:lwjgl.jar:lwjgl_util.jar:jorbis.jar:jinput.jar" > > # XMODIFIERS is cleared here to prevent SCIM screwing up keyboard > input XMODIFIERS= java \ > -noverify \ > -Djava.library.path="${INSTDIR}" \ > -Dorg.lwjgl.util.NoChecks=true \ > -Dorg.lwjgl.librarypath="${INSTDIR}" \ > -Dnet.puppygames.applet.Launcher.resources=/resources-hib.dat > \ > -Dnet.puppygames.applet.Game.gameResource=game.hib > \ > -XX:MaxGCPauseMillis=3 \ > -Xms64m \ > -Xmx375m \ > -Xincgc \ > -cp "${JARPATH}" \ > net.puppygames.applet.Launcher \ > "$@" \ > >"${LOGFILE}" 2>&1 > > exit 0 > > # > # EOF > # > ###############################################################################

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  • Ubuntu 12.04 - No sound - HELP!

    - by Bruno Tacca
    I'm panicking... my sound stopped working after I tried to set-up my notebook speakers, plus two headphone jacks... My idea was to multichannel the sound to 3 channels, built-in speakers, and sound-card 2 headphone jacks. After a couple efforts I did it with 2 channels, speakers and 1 headphone jack, but the other wasn't working. After more tries and tries, sound stop working. I just want my sound back... crying like a baby on the floor. And, if possible, but not necessary, a simple guide to active the 3 channels. xD I will post the diagnosis according to https://help.ubuntu.com/community/SoundTroubleshootingProcedure STEP 1 Did it, still no sound. STEP 2 Did it, still no sound. STEP 3 and #STEP 4 (I removed the log cause there is a limit of characters to be posted.) The log can be found here: https://answers.launchpad.net/ubuntu/+source/alsa-driver/+question/238653 STEP 5 Rebooted, still no sound. STEP 6 Did it. In the Output Devices tab, nothing is muted. I play a music with the Rhythmbox Music Player, I don't hear anything but in the pavucontrol I can see in the Built-in Audio Analog Stereo a sound bar shaking... but, no sound. STEP 7 In alsamixer, AlsaMixer v1.0.25 Card: HDA Intel PCH Chip: Creative CA0132 information View: F3:[Playback] F4: Capture F5: All Item: Headphone [dB gain: 25.00, 25.00] Then, I have 5 columns Headphone, Speaker, PCM, S/PDIF, S/PDIF Default PCM A little weird when I try to mute the Headphone and the Speaker, here what happens: Starting both unmutted, mutting headphone cause speaker being mutted automaticaly. Starting both unmutted, mutting speaker cause headphone being mutted automaticaly. Starting both mutted, possible to unmute both separately. STEP 8 I cannot hear sound on both (headphone and/or speaker). STEP 9 Dual boot... Restarted, windows was with sound at max volume. Restarted again, still no sound at ubuntu. I heard something when ubuntu started, a little noise, then silence again. The sound icon always start mutted, after unmutting, I have no sound. STEP 10 I dont have this command in my ubuntu. STEP 11 Tried at STEP 8, no sound. There are no problem with jumpers or hardware, cause I have sound working on windows. STEP 12 No way to open my alienware and loss the warranty x.X" STEP 13 I think it's loaded, judging my the logs STEP 14 Alienware M17xR4, the hardware is listed in the logs above, at STEP 4. There are two headphone hacks, one with just an headphone printed above, and the other with an headset (with mic) printed, there is a mic jack too, and a spdif (optical) too. STEP 15 I dont want to enable S/PDIF STEP 16 I never used the HDMI output, yet... Thanks in advance. I hope I listed all the information you need.

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  • Improve Playback Using Enhancements in Windows Media Player 12

    - by DigitalGeekery
    Are you looking for ways to improve the playback of your media in Windows Media Player 12? We’ll show you how to do that by using the enhancements in WMP 12. If you are in Library mode, you’ll need to click the icon at the lower right to switch to Now Playing mode. Right-click anywhere in Media Player while in Now Playing mode, select Enhancements, and select any of the available options.   You can switch between the individual enhancements by clicking the right and left buttons at the top left.   Crossfading and Auto Volume Leveling The Auto Volume Leveling setting is just a simple toggle on and off. If your MP3 or WMA files have volume leveling information values.   You can automatically add volume leveling information values to all files you add to your library by switching to Library view, going to Tools > Options, and selecting Add volume leveling information values for new files on the Library tab. Click OK when finished.   Crossfading will gradually decrease the volume of the song that is ending (fade out) and increase volume of the song that is beginning. Click Turn on Crossfading and then click and drag the slider left or right change the amount of overlap between tracks. Graphic Equalizer The graphic equalizer is toggled on and off by clicking Turn on / Turn off at the top left. You can select pre-defined equalizer settings by music genre by clicking the Default list. The radio buttons on the left allow you to move the sliders individually, in a loose group or a tight group. You can always return to the default settings by clicking Reset. Play Speed Settings Choose a pre-defined settings by clicking Slow, Normal, or Fast. Uncheck the Snap slider to common speeds the move the slider right and left to your desired speed. If nothing else, these settings provide a little fun and amusement. Quiet Mode Quiet mode will level out any sharp volume highs and lows within a single track. Simply toggle the setting on or off and select whether you prefer Medium difference or Little difference by selecting one of the radio buttons. SRS WOW effects SRS WOW effects enhance low-frequency and stereo sound performance. Click Turn on to enable the TruBass and WOW Effect sliders. You can also optimize for your speaker type. Click to switch between Regular, Large, and Headphones. Video Settings Video Settings allow you to adjust the Hue, Brightness, Saturation, and Contrast.   You can also adjust the zoom settings by clicking Select video zoom settings.   Dolby Digital Settings Choose between Normal, Night, and Theater settings to adjust the audio for Dolby Digital content. This setting will only effect media with Dolby Digital sound. Looking for more ways to improve your media experience in WMP 12? Check out how to update metadata and cover art and how to share media with other Windows 7 computers on your home network. Similar Articles Productive Geek Tips Fixing When Windows Media Player Library Won’t Let You Add FilesInstall and Use the VLC Media Player on Ubuntu LinuxHow To Rip a Music CD in Windows 7 Media CenterStream Media from Windows 7 to XP with VLC Media PlayerInstalling Windows Media Player Plugin for Firefox TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Acronis Online Backup DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows Check these Awesome Chrome Add-ons iFixit Offers Gadget Repair Manuals Online Vista style sidebar for Windows 7 Create Nice Charts With These Web Based Tools Track Daily Goals With 42Goals Video Toolbox is a Superb Online Video Editor

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  • I got my z-5 Logitech speakers to work, but whenever I restart, I have to reconfigure them

    - by The Bill
    This is the content of my alsa-base.conf file (for some reason, the entries preceded by # are bolded--anyway): autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-usb-audio index=0 options snd-hda-intel index=1 Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 options snd-usb-audio index=-2 options snd-usb-audio index=0 alias snd-card-0 snd-usb-audio alias snd-card-1 snd-hda-intel options snd-hda-intel index=1 I deleted a line that said something like "#Keep usb-audio from being loaded as first soundcard" and that made the speakers work for the first time (before this, they never showed up). I also added the last four lines. Anyway, what can I add to this so that I don't have to reconfigure them each time I restart? Currently, I have to open Sound Settings, then under the hardware tab, select Analog Stereo Output, and then unplug my USB speakers and plug them back in. This makes them pop up so that I can see them. Otherwise, it will not show my Z-5 speakers as a device that can be configured.

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  • The Power to Control Power

    - by speakjava
    I'm currently working on a number of projects using embedded Java on the Raspberry Pi and Beagle Board.  These are nice and small, so don't take up much room on my desk as you can see in this picture. As you can also see I have power and network connections emerging from under my desk.  One of the (admittedly very minor) drawbacks of these systems is that they have no on/off switch.  Instead you insert or remove the power connector (USB for the RasPi, a barrel connector for the Beagle).  For the Beagle Board this can potentially be an issue; with the micro-SD card located right next to the connector it has been known for people to eject the card when trying to power off the board, which can be quite serious for the hardware. The alternative is obviously to leave the boards plugged in and then disconnect the power from the outlet.  Simple enough, but a picture of underneath my desk shows that this is not the ideal situation either. This made me think that it would be great if I could have some way of controlling a mains voltage outlet using a remote switch or, even better, from software via a USB connector.  A search revealed not much that fit my requirements, and anything that was close seemed very expensive.  Obviously the only way to solve this was to build my own.Here's my solution.  I decided my system would support both control mechanisms (remote physical switch and USB computer control) and be modular in its design for optimum flexibility.  I did a bit of searching and found a company in Hong Kong that were offering solid state relays for 99p plus shipping (£2.99, but still made the total price very reasonable).  These would handle up to 380V AC on the output side so more than capable of coping with the UK 240V supply.  The other great thing was that being solid state, the input would work with a range of 3-32V and required a very low current of 7.5mA at 12V.  For the USB control an Arduino board seemed the obvious low-cost and simple choice.  Given the current requirments of the relay, the Arduino would not require the additional power supply and could be powered just from the USB.Having secured the relays I popped down to Homebase for a couple of 13A sockets, RS for a box and an Arduino and Maplin for a toggle switch.  The circuit is pretty straightforward, as shown in the diagram (only one output is shown to make it as simple as possible).  Originally I used a 2 pole toggle switch to select the remote switch or USB control by switching the negative connections of the low voltage side.  Unfortunately, the resistance between the digital pins of the Arduino board was not high enough, so when using one of the remote switches it would turn on both of the outlets.  I changed to a 4 pole switch and isolated both positive and negative connections. IMPORTANT NOTE: If you want to follow my design, please be aware that it requires working with mains voltages.  If you are at all concerned with your ability to do this please consult a qualified electrician to help you.It was a tight fit, especially getting the Arduino in, but in the end it all worked.  The completed box is shown in the photos. The remote switch was pretty simple just requiring the squeezing of two rocker switches and a 9V battery into the small RS supplied box.  I repurposed a standard stereo cable with phono plugs to connect the switch box to the mains outlets.  I chopped off one set of plugs and wired it to the rocker switches.  The photo shows the RasPi and the Beagle board now controllable from the switch box on the desk. I've tested the Arduino side of things and this works fine.  Next I need to write some software to provide an interface for control of the outlets.  I'm thinking a JavaFX GUI would be in keeping with the total overkill style of this project.

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  • No external microphone Acer AO722

    - by Leeghwater
    The ACER AO722 comes with an external mic input, and this input is not recognised by Alsa mixer or Sound (in System Settings). There are various comments on this problem, but no real solutions. For example External Mic not working but Internal Mic works on an Acer Aspiron AO722. Using the internal mic is not an option, as I need to use skype professionally. I have tried everything in alsamixer (accessible through the Terminal Ctrl+Alt+t, command: alsamixer), and in Sound (under System Settings). I have also installed Pulseaudio. But to no avail. The headset is working normally under Skype in Windows. My AO722 came with Windows 7 on it, so I have installed Skype there too. My headset has separate connectors for ears and mic, and these go into the respective output and input on the right side of the laptop. This location: http://bernaerts.dyndns.org/linux/202-ubuntu-acer-ao722 sounds like an effective solution but it is for Ubuntu Natty 11.04. The solution suggested sounds drastic to me: replace the kernel 2.6.38-13 with version 2.6.38-12. I use Ubuntu 12.04, and my kernel is 3.2.0-30-generic-pae. Question: could I try this solution with Ubuntu 12.04? Is this a risky thing to do? I have found harware work around this problem. The audio output seems to be a combi output with also a microphone connection. I have made an adapter for this output. I used a 4 contacts 3,5 mm audio jack plug. To this plug I have soldered 2 female (common stereo) connectors, one for ears and one for the mic of my headset. The 4 contacts jack, which goes into the laptop (in audio OUTput) is wired as follows: tip = hot audio right; first sleeve after tip = hot audio left; second sleeve = common earth (for both ears and microphone); the 3rd sleeve = microphone signal input. In the connector which I could buy, the 3rd sleeve is not so much a sleeve, but part of the metal base of the connector; normally you would expect this one to be connect to earth. But connecting the mic signal to it works. Maybe ready made adapters of this kind and even headsets with a combi jack can simply be purchased; I didn't check. When I plug in the 4 contacts jack, Sound and Alsamixer immediately recognise an external microphone (even if no mic is connected to the adapter). In Sound, under the Input tab, 'Settings for internal microphone' changes into 'Setting for microphone'. The microphone comes through loud and clear, however there is a constant noise in the background. Others have reported this too. If I disconnect the external mic from the adapter, or shortcircuit the external microphone, the noise gets less but does not disappear. Therefore, it is not background noise from the room, but it comes from the computer itself. However, if you talk directly in the microphone of the headset, the noise level is acceptable for VOIP. The headset of my mobile phone Nokia C1 mobile comes wwith a 4 contacts combi 3,5mm jack plug. However, this one works (ear and mic) with the AO722 only if not inserted fully. Possibly the wiring of this headset jack is different. I cannot find detailed specs of the AO722, and don't know whether the audio 'output' was actually designed as a combi input/output. I have seen that at least one other AO model has a combi connector only. In any case, I do not believe that connecting your headset in this way will harm your computer. I would still appreciate a software solution. This must be possible, because the proper microphone input connector works under MS Windows.

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  • AS3 microphone recording/saving works, in-flash PCM playback double speed

    - by Lowgain
    I have a working mic recording script in AS3 which I have been able to successfully use to save .wav files to a server through AMF. These files playback fine in any audio player with no weird effects. For reference, here is what I am doing to capture the mic's ByteArray: (within a class called AudioRecorder) public function startRecording():void { _rawData = new ByteArray(); _microphone.addEventListener(SampleDataEvent.SAMPLE_DATA, _samplesCaptured, false, 0, true); } private function _samplesCaptured(e:SampleDataEvent):void { _rawData.writeBytes(e.data); } This works with no problems. After the recording is complete I can take the _rawData variable and run it through a WavWriter class, etc. However, if I run this same ByteArray as a sound using the following code which I adapted from the adobe cookbook: (within a class called WavPlayer) public function playSound(data:ByteArray):void { _wavData = data; _wavData.position = 0; _sound.addEventListener(SampleDataEvent.SAMPLE_DATA, _playSoundHandler); _channel = _sound.play(); _channel.addEventListener(Event.SOUND_COMPLETE, _onPlaybackComplete, false, 0, true); } private function _playSoundHandler(e:SampleDataEvent):void { if(_wavData.bytesAvailable <= 0) return; for(var i:int = 0; i < 8192; i++) { var sample:Number = 0; if(_wavData.bytesAvailable > 0) sample = _wavData.readFloat(); e.data.writeFloat(sample); } } The audio file plays at double speed! I checked recording bitrates and such and am pretty sure those are all correct, and I tried changing the buffer size and whatever other numbers I could think of. Could it be a mono vs stereo thing? Hope I was clear enough here, thanks!

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  • FFMpeg Error av_interleaved_write_frame():

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

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  • 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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  • JMF microphone volume controller

    - by TacB0sS
    How to obtain the Microphone volume controller in JMF? this is what I have: I tried this implementation concept of yours, but I keep getting a null from the first volume processor when I try to get the stream, here is how I do it: // the device is the media device specifically audio Processor processorForVolume = Manager.createProcessor(device.getLocator()); // wait until configured ProcessorStates newState = new ProcessorStateListener(Processor.Configured).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // setting the content descriptor to null - read in another thread this allows to get the gain control processorForVolume.setContentDescriptor(null); // set the track control format to one supported by the device and the track control. // I didn't match it to an RTP allowed format, but I don't think this has anything to do with it... TrackControl[] trackControls = processorForVolume.getTrackControls(); if (trackControls.length == 0) throw new MC_Exception("No track controls where found for this device:", new Object[]{device}); for (TrackControl control : trackControls) trackManipulator.manipulateTrackControls(control); // wait until the processor is realized newState = new ProcessorStateListener(Controller.Realized).waitForProcessorState(processorForVolume); System.out.println("volumeProcessorState: "+newState); // receives the gain control micVolumeController = processorForVolume.getGainControl(); // cannot get the output stream to process further... any suggestions? processor = Manager.createProcessor(processorForVolume.getDataOutput()); new ProcessorStateListener(Processor.Configured).waitForProcessorState(processor); processor.setContentDescriptor(DeviceCapturingManager.RAW_RTP); new ProcessorStateListener(Controller.Realized).waitForProcessorState(processor); this is the output It generates: volumeProcessorState: Configured format set to track control - com.sun.media.ProcessEngine$ProcTControl@1627c16: LINEAR, 48000.0 Hz, 16-bit, Stereo, LittleEndian, Signed volumeProcessorState: Realized and the data output from the processor is Null. I should make clear that when the content descriptor != null I do get an output stream but not the volume controller, and the when it is null I get the controller, but no stream. I try to connect to an audio microphone device Adam.

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  • Rapid calls to fread crashes the application

    - by Slynk
    I'm writing a function to load a wave file and, in the process, split the data into 2 separate buffers if it's stereo. The program gets to i = 18 and crashes during the left channel fread pass. (You can ignore the couts, they are just there for debugging.) Maybe I should load the file in one pass and use memmove to fill the buffers? if(params.channels == 2){ params.leftChannelData = new unsigned char[params.dataSize/2]; params.rightChannelData = new unsigned char[params.dataSize/2]; bool isLeft = true; int offset = 0; const int stride = sizeof(BYTE) * (params.bitsPerSample/8); for(int i = 0; i < params.dataSize; i += stride) { std::cout << "i = " << i << " "; if(isLeft){ std::cout << "Before Left Channel, "; fread(params.leftChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Left Channel, "; } else{ std::cout << "Before Right Channel, "; fread(params.rightChannelData+offset, sizeof(BYTE), stride, file + i); std::cout << "After Right Channel, "; offset += stride; std::cout << "After offset incr.\n"; } isLeft != isLeft; } } else { params.leftChannelData = new unsigned char[params.dataSize]; fread(params.leftChannelData, sizeof(BYTE), params.dataSize, file); }

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  • How accurately (in terms of time) does Windows play audio?

    - by MusiGenesis
    Let's say I play a stereo WAV file with 317,520,000 samples, which is theoretically 1 hour long. Assuming no interruptions of the playback, will the file finish playing in exactly one hour, or is there some occasional tiny variation in the playback speed such that it would be slightly more or slightly less (by some number of milliseconds) than one hour? I am trying to synchronize animation with audio, and I am using a System.Diagnostics.Stopwatch to keep the frames matching the audio. But if the playback speed of WAV audio in Windows can vary slightly over time, then the audio will drift out of sync with the Stopwatch-driven animation. Which leads to a second question: it appears that a Stopwatch - while highly granular and accurate for short durations - runs slightly fast. On my laptop, a Stopwatch run for exactly 24 hours (as measured by the computer's system time and a real stopwatch) shows an elapsed time of 24 hours plus about 5 seconds (not milliseconds). Is this a known problem with Stopwatch? (A related question would be "am I crazy?", but you can try it for yourself.) Given its usage as a diagnostics tool, I can see where a discrepancy like this would only show up when measuring long durations, for which most people would use something other than a Stopwatch. If I'm really lucky, then both Stopwatch and audio playback are driven by the same underlying mechanism, and thus will stay in sync with each other for days on end. Any chance this is true?

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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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  • Syncing two AS3 NetStreams

    - by Lowgain
    I'm writing an app that requires an audio stream to be recording while a backing track is played. I have this working, but there is an inconsistent gap in between playback and record starting. I don't know if I can do anything to make the sync perfect every time, so I've been trying to track what time each stream starts so I can calculate the delay and trim it server-side. This also has proved to be a challenge as no events seem to be sent when a connection starts (as far as I know). I've tried using various properties like the streams' buffer sizes, etc. I'm thinking now that as my recorded audio is only mono, I may be able to put some kind of 'control signal' on the second stereo track which I could use to determine exactly when a sound starts recording (or stick the whole backing track in that channel so I can sync them that way). This leaves me with the new problem of properly injecting this sound into the NetStream. If anyone has any idea whether or not any of these ideas will work, how to execute them, or some alternatives, that would be extremely helpful! Been working on this issue for awhile

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  • PHP : How to get specific data from array

    - by Giffary
    Hello! I try to use amazon API using PHP. If I use print_r($parsed_xml->Items->Item->ItemAttributes) it show me some result like SimpleXMLElement Object ( [Binding] = Electronics [Brand] = Canon [DisplaySize] = 2.5 [EAN] = 0013803113662 [Feature] = Array ( [0] = High-powered 20x wide-angle optical zoom with Optical Image Stabilizer [1] = Capture 720p HD movies with stereo sound; HDMI output connector for easy playback on your HDTV [2] = 2.5-inch Vari-Angle System LCD; improved Smart AUTO intelligently selects from 22 predefined shooting situations [3] = DIGIC 4 Image Processor; 12.1-megapixel resolution for poster-size, photo-quality prints [4] = Powered by AA batteries (included); capture images to SD/SDHC memory cards (not included) ) [FloppyDiskDriveDescription] = None [FormFactor] = Rotating [HasRedEyeReduction] = 1 [IsAutographed] = 0 [IsMemorabilia] = 0 [ItemDimensions] = SimpleXMLElement Object ( [Height] = 340 [Length] = 490 [Weight] = 124 [Width] = 350 ) [Label] = Canon [LensType] = Zoom lens [ListPrice] = SimpleXMLElement Object ( [Amount] = 60103 [CurrencyCode] = USD [FormattedPrice] = $601.03 ) [Manufacturer] = Canon [MaximumFocalLength] = 100 [MaximumResolution] = 12.1 [MinimumFocalLength] = 5 [Model] = SX20IS [MPN] = SX20IS [OpticalSensorResolution] = 12.1 [OpticalZoom] = 20 [PackageDimensions] = SimpleXMLElement Object ( [Height] = 460 [Length] = 900 [Weight] = 242 [Width] = 630 ) [PackageQuantity] = 1 [ProductGroup] = Photography [ProductTypeName] = CAMERA_DIGITAL [ProductTypeSubcategory] = point-and-shoot [Publisher] = Canon [Studio] = Canon [Title] = Canon PowerShot SX20IS 12.1MP Digital Camera with 20x Wide Angle Optical Image Stabilized Zoom and 2.5-inch Articulating LCD [UPC] = 013803113662 ) my goal is to get only Feature infomation and I try to use $feature = $parsed_xml->Items->Item->ItemAttributes->Feature it does'not work for me because it just show me the first feature only. How do i get all feature information? please help

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  • Split UInt32 (audio frame) into two SInt16s (left and right)?

    - by morgancodes
    Total noob to anything lower-level than Java, diving into iPhone audio, and realing from all of the casting/pointers/raw memory access. I'm working with some example code wich reads a WAV file from disc and returns stereo samples as single UInt32 values. If I understand correctly, this is just a convenient way to return the 32 bits of memory required to create two 16 bit samples. Eventually this data gets written to a buffer, and an audio unit picks it up down the road. Even though the data is written in UInt32-sized chunks, it eventually is interpreted as pairs of 16-bit samples. What I need help with is splitting these UInt32 frames into left and right samples. I'm assuming I'll want to convert each UInt32 into an SInt16, since an audio sample is a signed value. It seems to me that for efficiency's sake, I ought to be able to simply point to the same blocks in memory, and avoid any copying. So, in pseudo-code, it would be something like this: UInt32 myStereoFrame = getFramefromFilePlayer; SInt16* leftChannel = getFirst16Bits(myStereoFrame); SInt16* rightChannel = getSecond16Bits(myStereoFrame); Can anyone help me turn my pseudo into real code?

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  • Optical SPDIF audio from motherboard not working with receiver

    - by simon b
    Hi, I hope someone can help; I can't get my SPDIF optical out working through my receiver and all the responses I can see on the web assume you have a sound card, while I settled for the (seemingly high end) sound on my motherboard (Asus P7P55D-E PRO), which appears to limit some of my options. My set-up is a "new out of the box" one and is: *Windows 7 PC (using PowerDVD10 for DVDs/Blurays and Windows media player for music) *Asus P7P55D-E PRO motherboard - has 8-channel audio TRS jacks and SPDIF optical and coaxial out *An old Yamaha receiver, whose only multi-channel input options are optical in and 6 channel RCA in. However, it still can handle DTS and DD *Boston Acoustic Soundware XS 5.1 speakers I've currently got the SPDIF optical out from the motherboard connected to the in on my receiver, have SPDIF enabled in the sound menu and the light is glowing red down the fibre. But I'm getting no sound at all. What I want is to be able to play DVDs/BluRays in 5.1 but also to be able to play music in multi-channel mode (even though I know this will be "fake" multichannel; it's more about where I sit in the room and my requirement to use the sub because the Boston is a satellite/sub set-up) My questions are: *Will optical work at all for multi-channel? THe latest posts I can see suggest it does but some people seem to say optical only outputs stereo. Whom to believe? *Even if it does work, I've read that I have to disable AC-3 decoding, or make various other changes, which don't seem to be possible without the menu options that a sound-card brings. Is the motherboard-only option just too inflexible? *Although my SPDIF device is enabled in the sound menu, it insists under "Jack information" that it is a "rear panel RCA jack", when of course it is not (both TOSLINK and rCA jacks do exist). Has the PC just forgotten that it has an optical? *I think I could relatively easily connect the 8-channel 3.5mm TRS jacks to my receiver 6-ch input jacks by way of TRS/RCA cables, but would that not stop me from being able to play music from media-player in multi-channel mode, as I'm not sure the motherboard can cope *Or do I need to bite the bullet and buy a sound-card? And if so, how can I be sure the one I get doesn't have the same problem? Any thoughts gratefully received, Cheers, simon

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  • Customizing Fantacy Remote .INI file

    - by karthik
    I am using Fantacy Remote to remote view other machines. I have attached the default .INI file that Fantacy Remote uses. When i connect to a machine, the client user should not have mouse and keyboard access of the Remote machine. It should be a View only remote connection. And i want to make the Remote viewer screen to be in full screen mode, because i dont want the user to do anything with menubars of Fatancy remote. Because this is kiosk application. What should i change in configuration file [.ini] inorder to achieve the above ? Anyone who have used this software before, kindly help.. [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=192.168.1.1 pcPort=7902 [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

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  • Convert mkv/h264 video so it can be played on a "mid-range" Sony Ericsson phone. (using Ubuntu).

    - by Johan
    Hi As a little experiment I thinking of converting some video/movies/tv-series into a format that could be playable on my K850, but to be a little bit more generic in this question let's say "mid range Sony Ericsson" phone since they all more or less behave the same and has the same screen resolution (240 x 320). I am looking for command line based tools (for Ubuntu), since I am thinking about writing a "convert and move" script later if it is successful. A lot of the video I have is encoded in mkv/h264, but since that is not supported by the phone I guess that I need to convert it into some mp4/mpeg4 low quality video. After some googling it seems like a good candidate for the job is ffmpeg, but that seems to be a very versatile tool with a lot of magic tricks. Am I on the right track? And if so how do I use ffmpeg to do this? Thanks Johan Update: After plating a little bit with ffmeg I noticed that it only uses 1 of my 4 cores, so the transcoding takes forever. I found a arg called -threads but that did not change much, maybe I got it wrong. I also found that something like this plays in the phone. ffmpeg -i Mythbusters\ S1D1_1.mkv -threads 4 -t 180 -vcodec mpeg4 -r 15 -s 320x240 Mythbusters\ S1D1_1_mini.mp4 It was possible to use 3gp/h263, but the quality was really useless. ffmpeg -i Mythbusters\ S1D1_1.mkv -t 180 -vcodec h263 -acodec libfaac -s cif Mythbusters\ S1D1_1_cif.3gp And it seems like mp4/h264 is also possible and the result is ok, thanks to this question, this one seem to use more than one core as well so it was a little bit faster for me. ffmpeg -i Mythbusters_S1D1_1.mkv -t 180 -acodec libfaac -ab 60k -s 320x240 -vcodec libx264 -b 500k -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -flags2 +mixed_refs -me_method umh -subq 6 -trellis 1 -refs 5 -coder 0 -me_range 16 -g 250 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -bt 500k -maxrate 768k -bufsize 2M -qcomp 0.6 -qmin 10 -qmax 51 -qdiff 4 -level 13 -threads 0 -f mp4 Mythbusters_S1D1_1_qvga.mp4 Update: I have tried to use HandBrakeCLI and it is no problem creating a new file that seem to be the same as the one created with ffmpeg with something like this. HandBrakeCLI -i Mythbusters_S1D1_1.mkv --size 100 -E faac -B 60 --maxHeight 240 -r 15 -e x264 -o Mythbusters_S1D1_1_hand.mp4 But that one did not play in the phone... I found this in the official manual: If you transfer video clips using another program than Media Go™, we recommend that you select H.264 Baseline profile video, up to QVGA at 30 fps, VBR 384 kbps (max 768 kps) with AAC+ audio at 128 kbps (max 255 kbps), 48 kHz and stereo audio in mp4 file format. So the idea to use H264 seems to be correct.

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  • DVD playback with Windows Media Player 11 works fine, but when copied to HDD and then played back, t

    - by stakx
    I have several DVDs with short documentaries on it. Since the notebook I'm using (a Dell Latitude E6400) has only one DVD drive, and I might play back those short movies very often, I thought of copying them to the HDD and playing them back from there. However, I've run into a problem, namely stuttering audio. Problem description: When I play back these movies directly from DVD (with Windows Media Player 11 under Windows Vista), everything works fine. Smooth video, no significant audio problems (only the occasional click). But as soon as I copy any of these DVDs to the HDD and try to play them back from there (e.g. using the wmpdvd://drive/title/chapter?contentdir=path protocol, I get stuttering audio — audio playback sounds like a machine gun for a third of a second or so, approx. every 8 seconds. I have tried converting the VOB files from the DVD to another format (ie. ripping), but that resulted in a noticeable downgrade of picture quality. Therefore I thought it best to keep the files in their original format, if possible. Still, I suspect that the stuttering audio is due to some (de-)muxing problem, and that changing the file format might help. (After all, video playback is fine; therefore I don't think that the hardware is too slow for playback.) Only thing is, I don't know how to convert the VOB files to another Windows Media Player-compatible format without quality loss. I hope someone can help me, or give me further pointers on things I could try out to get HDD playback to work without the problem described. Some things I've tried so far, without any success: VOB2MPG, in order to convert the .vob file to a .mpg file. But that changes only the A/V container, not the content. No re-encoding takes place at all. Re-encoding with MPlayer/MEncoder. Lots of quality loss there, and I frankly haven't got the time to test all possible settings combinations available. Disabling all plug-ins, equalizers, etc. in Windows Media Player. Disabling all hardware acceleration on the audio playback device. Further info on the VOB files I'm trying to playback: The video format is MPEG ES, PAL 720x576 pixels @ 24/25 frames per second. The sound stream is uncompressed PCM, 16-bit stereo @ 48kHz. (Might it help if I somehow re-encoded the sound stream at a lower resolution, or as an MP3? If so, how would I do this without changing the video stream?) P.S.: I am limited to using Windows Media Player (11). (I previously tried MPlayer btw., but the video playback quality was surprisingly bad.)

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  • ffmpeg - join / merge on top of each other

    - by AisIceEyes
    I'm trying to join together two videos on top of each other. I already did these two ffmpeg commands ffmpeg -i 2_Out_of_Control.VOB -aspect 16:9 \ -vf "yadif=0:-1:0,crop=w=714:h=476:x=6:y=0,scale=1280:720,boxblur=lp=13" \ -c:v libx264 -preset medium \ -c:a copy \ '2(blurred)Out_of_Control.mp4' ffmpeg -i 2_Out_of_Control.VOB \ -vf "yadif=0:-1:0,crop=w=714:h=476:x=6:y=0,scale=1080:720" \ -c:v libx264 -preset medium \ -c:a copy \ '2(clear)Out_of_Control.mp4' I'm currently stuck on making the "clear" version on top of the "blurred" version. I'm not sure how to do that. Can anybody help please? Been googling for around 2 days already. Only achieved it by using OpenShot but yeah, would prefer if there is an ffmpeg command to merge the two videos on top of each other. Edit: I want the "clear" video to be at the center at the top of the "blurred" video Edit2: console output would be the same as above: ffmpeg -i 2(blurred)Out_of_Control.mp4 \ -i 2(clear)Out_of_Control.mp4 \ -aspect 16:9 \ -vf <just something that will join the two together: the blurred at the bottom, clear at top that is centered> \ -c:v libx264 -preset medium \ -c:a copy \ '2_Out_of_Control_VOB.mp4' Edit3: here is the output when I used ffmpeg -i 2_Out_of_Control.VOB $ ffmpeg -i 2_Out_of_Control.VOB ffmpeg version git-2013-10-03-c7fe2a3 Copyright (c) 2000-2013 the FFmpeg developers built on Oct 4 2013 05:22:06 with gcc 4.6 (Ubuntu/Linaro 4.6.3-1ubuntu5) configuration: --prefix=/home/username/ffmpeg_build --extra-cflags=-I/home/username/ffmpeg_build/include --extra-ldflags=-L/home/username/ffmpeg_build/lib --bindir=/home/username/bin --extra-libs=-ldl --enable-gpl --enable-libass --enable-libfdk-aac --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-nonfree --enable-x11grab libavutil 52. 46.100 / 52. 46.100 libavcodec 55. 34.100 / 55. 34.100 libavformat 55. 19.100 / 55. 19.100 libavdevice 55. 3.100 / 55. 3.100 libavfilter 3. 88.101 / 3. 88.101 libswscale 2. 5.100 / 2. 5.100 libswresample 0. 17.103 / 0. 17.103 libpostproc 52. 3.100 / 52. 3.100 Input #0, mpeg, from '2_Out_of_Control.VOB': Duration: 00:05:00.01, start: 0.500000, bitrate: 4574 kb/s Stream #0:0[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, smpte170m), 720x480 [SAR 8:9 DAR 4:3], max. 9334 kb/s, 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc Stream #0:1[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 384 kb/s At least one output file must be specified $

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  • openGL migration from SFML to glut, vertices arrays or display lists are not displayed

    - by user3714670
    Due to using quad buffered stereo 3D (which i have not included yet), i need to migrate my openGL program from a SFML window to a glut window. With SFML my vertices and display list were properly displayed, now with glut my window is blank white (or another color depending on the way i clear it). Here is the code to initialise the window : int type; int stereoMode = 0; if ( stereoMode == 0 ) type = GLUT_DOUBLE | GLUT_RGB | GLUT_DEPTH; else type = GLUT_DOUBLE | GLUT_RGB | GLUT_DEPTH | GLUT_STEREO; glutInitDisplayMode(type); int argc = 0; char *argv = ""; glewExperimental = GL_TRUE; glutInit(&argc, &argv); bool fullscreen = false; glutInitWindowSize(width,height); int win = glutCreateWindow(title.c_str()); glutSetWindow(win); assert(win != 0); if ( fullscreen ) { glutFullScreen(); width = glutGet(GLUT_SCREEN_WIDTH); height = glutGet(GLUT_SCREEN_HEIGHT); } GLenum err = glewInit(); if (GLEW_OK != err) { fprintf(stderr, "Error: %s\n", glewGetErrorString(err)); } glutDisplayFunc(loop_function); This is the only code i had to change for now, but here is the code i used with sfml and displayed my objects in the loop, if i change the value of glClearColor, the window's background does change color : glClear(GL_COLOR_BUFFER_BIT | GL_DEPTH_BUFFER_BIT); glClearColor(255.0f, 255.0f, 255.0f, 0.0f); glLoadIdentity(); sf::Time elapsed_time = clock.getElapsedTime(); clock.restart(); camera->animate(elapsed_time.asMilliseconds()); camera->look(); for (auto i = objects->cbegin(); i != objects->cend(); ++i) (*i)->draw(camera); glutSwapBuffers(); Is there any other changes i should have done switching to glut ? that would be great if someone could enlighten me on the subject. In addition to that, i found out that adding too many objects (that were well handled before with SFML), openGL gives error 1285: out of memory. Maybe this is related. EDIT : Here is the code i use to draw each object, maybe it is the problem : GLuint LightID = glGetUniformLocation(this->shaderProgram, "LightPosition_worldspace"); if(LightID ==-1) cout << "LightID not found ..." << endl; GLuint MaterialAmbientID = glGetUniformLocation(this->shaderProgram, "MaterialAmbient"); if(LightID ==-1) cout << "LightID not found ..." << endl; GLuint MaterialSpecularID = glGetUniformLocation(this->shaderProgram, "MaterialSpecular"); if(LightID ==-1) cout << "LightID not found ..." << endl; glm::vec3 lightPos = glm::vec3(0,150,150); glUniform3f(LightID, lightPos.x, lightPos.y, lightPos.z); glUniform3f(MaterialAmbientID, MaterialAmbient.x, MaterialAmbient.y, MaterialAmbient.z); glUniform3f(MaterialSpecularID, MaterialSpecular.x, MaterialSpecular.y, MaterialSpecular.z); // Get a handle for our "myTextureSampler" uniform GLuint TextureID = glGetUniformLocation(shaderProgram, "myTextureSampler"); if(!TextureID) cout << "TextureID not found ..." << endl; glActiveTexture(GL_TEXTURE0); sf::Texture::bind(texture); glUniform1i(TextureID, 0); // 2nd attribute buffer : UVs GLuint vertexUVID = glGetAttribLocation(shaderProgram, "color"); if(vertexUVID==-1) cout << "vertexUVID not found ..." << endl; glEnableVertexAttribArray(vertexUVID); glBindBuffer(GL_ARRAY_BUFFER, color_array_buffer); glVertexAttribPointer(vertexUVID, 2, GL_FLOAT, GL_FALSE, 0, 0); GLuint vertexNormal_modelspaceID = glGetAttribLocation(shaderProgram, "normal"); if(!vertexNormal_modelspaceID) cout << "vertexNormal_modelspaceID not found ..." << endl; glEnableVertexAttribArray(vertexNormal_modelspaceID); glBindBuffer(GL_ARRAY_BUFFER, normal_array_buffer); glVertexAttribPointer(vertexNormal_modelspaceID, 3, GL_FLOAT, GL_FALSE, 0, 0 ); GLint posAttrib; posAttrib = glGetAttribLocation(shaderProgram, "position"); if(!posAttrib) cout << "posAttrib not found ..." << endl; glEnableVertexAttribArray(posAttrib); glBindBuffer(GL_ARRAY_BUFFER, position_array_buffer); glVertexAttribPointer(posAttrib, 3, GL_FLOAT, GL_FALSE, 0, 0); glBindBuffer(GL_ELEMENT_ARRAY_BUFFER, elements_array_buffer); glDrawElements(GL_TRIANGLES, indices.size(), GL_UNSIGNED_INT, 0); GLuint error; while ((error = glGetError()) != GL_NO_ERROR) { cerr << "OpenGL error: " << error << endl; } disableShaders();

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  • Multiple Audio Issues

    - by Lerp
    I am having issues with my audio on Ubuntu 12.04, I will try and give as much detail as possible so sorry if there's too much detail. The Problem Audio plays from both speakers and headphone regardless of what connector I choose and regardless of the profile I use. The microphone is constantly being played through headphones & speakers. The headphone audio is extremely quiet but plays from both ears when I select "Headphones" for the connector in Sound Settings. The headphone audio only plays from one ear and is quiet (but not as quiet as above) when I select "Analogue Output" for the connector in Sound Settings. I can only select "Headphones" as the connector in Sound Settings if I set the profile to either "Analogue Stereo Output/Duplex", all others only allow me to choose "Analogue Output" for the connector. Despite the headphone sound issues, the speaker sound is fine apart from the fact that I am not able to select which output is used, they just both play. My headphone and microphone are plugged into the front and my speakers are plugged into the back. What I have tried I have put everything in alsamixer to 100 apart from "Front Mic Boost" which I have set to 0. Command Output aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 2: AD198x Headphone [AD198x Headphone] Subdevices: 1/1 Subdevice #0: subdevice #0 arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 2/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf7ff8000 irq 70 cat /proc/asound/modules 0 snd_hda_intel cat /proc/asound/card*/codec* | grep "Codec" Codec: Analog Devices AD1989B cat /etc/modprobe.d/alsa-base.conf # autoloader aliases install sound-slot-0 /sbin/modprobe snd-card-0 install sound-slot-1 /sbin/modprobe snd-card-1 install sound-slot-2 /sbin/modprobe snd-card-2 install sound-slot-3 /sbin/modprobe snd-card-3 install sound-slot-4 /sbin/modprobe snd-card-4 install sound-slot-5 /sbin/modprobe snd-card-5 install sound-slot-6 /sbin/modprobe snd-card-6 install sound-slot-7 /sbin/modprobe snd-card-7 # Cause optional modules to be loaded above generic modules install snd /sbin/modprobe --ignore-install snd $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-ioctl32 ; /sbin/modprobe --quiet --use-blacklist snd-seq ; } # # Workaround at bug #499695 (reverted in Ubuntu see LP #319505) install snd-pcm /sbin/modprobe --ignore-install snd-pcm $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-pcm-oss ; : ; } install snd-mixer /sbin/modprobe --ignore-install snd-mixer $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-mixer-oss ; : ; } install snd-seq /sbin/modprobe --ignore-install snd-seq $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; /sbin/modprobe --quiet --use-blacklist snd-seq-oss ; : ; } # install snd-rawmidi /sbin/modprobe --ignore-install snd-rawmidi $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq-midi ; : ; } # Cause optional modules to be loaded above sound card driver modules install snd-emu10k1 /sbin/modprobe --ignore-install snd-emu10k1 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-emu10k1-synth ; } install snd-via82xx /sbin/modprobe --ignore-install snd-via82xx $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist snd-seq ; } # Load saa7134-alsa instead of saa7134 (which gets dragged in by it anyway) install saa7134 /sbin/modprobe --ignore-install saa7134 $CMDLINE_OPTS && { /sbin/modprobe --quiet --use-blacklist saa7134-alsa ; : ; } # Prevent abnormal drivers from grabbing index 0 options bt87x index=-2 options cx88_alsa index=-2 options saa7134-alsa index=-2 options snd-atiixp-modem index=-2 options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-usb-audio index=-2 options snd-usb-caiaq index=-2 options snd-usb-ua101 index=-2 options snd-usb-us122l index=-2 options snd-usb-usx2y index=-2 # Ubuntu #62691, enable MPU for snd-cmipci options snd-cmipci mpu_port=0x330 fm_port=0x388 # Keep snd-pcsp from being loaded as first soundcard options snd-pcsp index=-2 # Keep snd-usb-audio from beeing loaded as first soundcard options snd-usb-audio index=-2 Hopefully I have provided enough information, I will happily provide anymore information needed. Thank you. Update Reinstalling alsa-base and pulseaudio fixed the headphone issues I was having.

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  • Beat Detection on iPhone with wav files and openal

    - by Dmacpro
    Using this website i have tried to make a beat detection engine. http://www.gamedev.net/reference/articles/article1952.asp { ALfloat energy = 0; ALfloat aEnergy = 0; ALint beats = 0; bool init = false; ALfloat Ei[42]; ALfloat V = 0; ALfloat C = 0; ALshort *hold; hold = new ALshort[[myDat length]/2]; [myDat getBytes:hold length:[myDat length]]; ALuint uiNumSamples; uiNumSamples = [myDat length]/4; if(alDatal == NULL) alDatal = (ALshort *) malloc(uiNumSamples*2); if(alDatar == NULL) alDatar = (ALshort *) malloc(uiNumSamples*2); for (int i = 0; i < uiNumSamples; i++) { alDatal[i] = hold[i*2]; alDatar[i] = hold[i*2+1]; } energy = 0; for(int start = 0; start<(22050*10); start+=512){ //detect for 10 seconds of data for(int i = start; i<(start+512); i++){ energy+= fabs(alDatal[i]) + fabs(alDatar[i]); } aEnergy = 0; for(int i = 41; i>=0; i--){ if(i ==0){ Ei[0] = energy; } else { Ei[i] = Ei[i-1]; } if(start >= 21504){ aEnergy+=Ei[i]; } } aEnergy = aEnergy/43.f; if (start >= 21504) { for(int i = 0; i<42; i++){ V += (Ei[i]-aEnergy); } V = V/43.f; C = (-0.0025714*V)+1.5142857; init = true; if(energy >(C*aEnergy)) beats++; } } } alDatal and alDatar are (short*) type; myDat is NSdata that holds the actual audio data of a wav file formatted to 22050 khz and 16 bit stereo. This doesn't seem to work correctly. If anyone could help me out that would be amazing. I've been stuck on this for 3 days.

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