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  • Is there an equivalent of jsqlparser but for SPARQL instead of SQL?

    - by Programmer
    I'm trying to use Java to construct a SPARQL query, and then send it off to a remote database. However, I'm new to both Java and SPARQL, so I was wondering if anyone could explain how to do this, rather than just posting a link. I heard there is a tool called jsqlparser for the same task, except that it's for a SQL to SPARQL conversion using Java. Conversion nor parser won't be necessary, just a method for constructing a query and querying the database provided by the user.

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  • How to get sound on macbook pro 4,1

    - by Thomas
    I have just installed Xubuntu 12.04.2. My soundcard is detected: thomas@thomas-pc:~$ sudo aplay -l **** List of PLAYBACK Hardware Devices **** Home directory /home/thomas not ours. card 0: Intel [HDA Intel], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 Everything is put to max in alsamixer and nothing is muted (all the sliders are on OO. My speakers do not work, but when I plug in a headphone I hear it very soft. When I connect my stereo and put the sound VERY loud (3-blocks-of-complaining-neighbours loud) I hear it on a normal level but crackling. I added options snd-hda-intel model=mbp5 amixer set IEC958 off to at the end of /etc/modprobe.d/alsa-base.conf. When it's still not working I tried everything here: https://help.ubuntu.com/community/SoundTroubleshooting 1 >>> list-sinks 1 sink(s) available. * index: 0 name: <alsa_output.pci-0000_00_1b.0.analog-stereo> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE priority: 9959 volume: 0: 100% 1: 100% 0: 0.00 dB 1: 0.00 dB balance 0.00 base volume: 100% 0.00 dB volume steps: 65537 muted: no current latency: 0.00 ms max request: 0 KiB max rewind: 0 KiB monitor source: 0 sample spec: s16le 2ch 44100Hz channel map: front-left,front-right Stereo used by: 0 linked by: 0 configured latency: 0.00 ms; range is 0.50 .. 371.52 ms card: 0 <alsa_card.pci-0000_00_1b.0> module: 4 properties: alsa.resolution_bits = "16" device.api = "alsa" device.class = "sound" alsa.class = "generic" alsa.subclass = "generic-mix" alsa.name = "ALC889A Analog" alsa.id = "ALC889A Analog" alsa.subdevice = "0" alsa.subdevice_name = "subdevice #0" alsa.device = "0" alsa.card = "0" alsa.card_name = "HDA Intel" alsa.long_card_name = "HDA Intel at 0x9b500000 irq 46" alsa.driver_name = "snd_hda_intel" device.bus_path = "pci-0000:00:1b.0" sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0" device.bus = "pci" device.vendor.id = "8086" device.vendor.name = "Intel Corporation" device.product.name = "82801H (ICH8 Family) HD Audio Controller" device.form_factor = "internal" device.string = "front:0" device.buffering.buffer_size = "65536" device.buffering.fragment_size = "32768" device.access_mode = "mmap+timer" device.profile.name = "analog-stereo" device.profile.description = "Analog Stereo" device.description = "Built-in Audio Analog Stereo" alsa.mixer_name = "Realtek ALC889A" alsa.components = "HDA:10ec0885,106b3a00,00100103" module-udev-detect.discovered = "1" device.icon_name = "audio-card-pci" ports: analog-output-speaker: Speakers (priority 10000, available: unknown) properties: analog-output-headphones: Headphones (priority 9000, available: no) properties: active port: <analog-output-speaker> 2 and 3: Doesn't seem an permission issue, the sound is very far away (See opening paragraph). 4 thomas@thomas-pc:~$ sudo aplay -l **** List of PLAYBACK Hardware Devices **** Home directory /home/thomas not ours. card 0: Intel [HDA Intel], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 5 thomas@thomas-pc:~$ find /lib/modules/`uname -r` | grep snd /lib/modules/3.2.0-48-generic/kernel/sound/core/snd-hwdep.ko /lib/modules/3.2.0-48-generic/kernel/sound/core/snd-pcm.ko [.. huge lists continues ..] /lib/modules/3.2.0-48-generic/kernel/sound/pcmcia/pdaudiocf/snd-pdaudiocf.ko /lib/modules/3.2.0-48-generic/kernel/sound/pcmcia/vx/snd-vxpocket.ko thomas@thomas-pc:~$ 6 thomas@thomas-pc:~$ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Apple Inc. Device 00a4 Flags: bus master, fast devsel, latency 0, IRQ 46 Memory at 9b500000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel 7 I guess it's supported. Linux mint and Xubuntu 13.04 had no trouble with sounds. Everything worked out of the box Thanks in advance Edit: alsa-info.sh output: WARNING: /etc/modprobe.d/alsa-base.conf line 45: ignoring bad line starting with 'amixer' ALSA Information Script v 0.4.62 -------------------------------- This script visits the following commands/files to collect diagnostic information about your ALSA installation and sound related hardware. dmesg lspci lsmod aplay amixer alsactl /proc/asound/ /sys/class/sound/ ~/.asoundrc (etc.) See './alsa-info.sh --help' for command line options. WARNING: /etc/modprobe.d/alsa-base.conf line 45: ignoring bad line starting with 'amixer' Automatically upload ALSA information to www.alsa-project.org? [y/N] : y Uploading information to www.alsa-project.org ... Done! Your ALSA information is located at http://www.alsa-project.org/db/?f=6cffc584284d4c0b266eb53249824ef83d6c4e3e Please inform the person helping you. thomas@thomas-pc:~$

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  • What changed with timidity, alsa and jack in 11.10?

    - by Dave
    I (just) upgraded from 11.4 to 11.10 and noticed some differences in the behavior of timidity. I used to (11.4) exectute >timidity midifile.midi without running jackd, and thus using alsa (or pulseaudio?) to produce sound from midi files. Now having upgraded, this does not work -- currently this command just freezes if jack is not running. If jack is running, it does work but there is an initial audio glitch (noise burst at the start of playback, analogous to the sound of a plug being inserted) that I'd rather not have to deal with. All the indications that I have is that in 11.10 timidity will only work (albeit glitchy) with jack on, whereas in 11.4 it did not require this. Is there any way to restore timidity's non-jack operation in 11.10? Is there a way to get rid of the audio glitch in with jack operation? Overall, what underlying changes in these programs and the audio infrastructure are behind this?

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  • How do I know if my system is capable of playing 24bit/96kHz sound?

    - by Igor Zinov'yev
    Let me state for the record that I'm a total noob when it comes to Hi-Fi sound systems, but I am rather picky about the sound quality. Normally I listen to CD recordings ripped to FLAC in 16/44, but I have several albums that are also ripped from vinyls to FLAC in 24/96. But it seems that I can't tell the difference between 16-bit and 24-bit versions (except for some vinyl noises, of course). That can be due to several reasons: my equipment (onboard audio, monitor headphones) isn't good enough to make any difference, my system is not playing audio in 24-bit 96 kHz, I am physically unable to hear the difference. So here is my question, how do I tell if my system can play 24-bit sound with 96 or 192 kHz resolution? And if it can, how do I tell that it plays it instead of downsampling to 16-bit / 44 kHz? Also, what hardware (audio cards, amplifiers, etc.) would you recommend to play such recordings on Ubuntu?

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  • pulseaudio and alsa on ubuntu 12.04 server

    - by Dan
    I am running ubuntu 12.04server, and trying to get pulseaudio working. I followed the instructions at How do I run PulseAudio in a headless server installation? At the moment, pacmd list-cards is reporting 0 cards, aplay will only playing sound when I run it as sudo, and running alsamixer as sudo also works, but running it as my user produces "cannot open mixer: No such file or directory" As far as I can tell, this means the the kernel module for my sound card is in fact loaded. I have already tried adding my user to the "audio" group, but this does not help. The permissions on the devices in /dev/snd are all crw-rw---T 1 root audio 116 I noticed on an ubuntu 12.04 desktop, that the file permissions are slightly different. On the desktop, they are crw-rw---T+ 1 root audio 116 My questions are 1) How do I get aplay to work without running it as sudo on the server 2) Is there anything special I need to do to make pulseaudio work at this point.

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  • Firefox, Chrome, and Flash on Ubuntu

    - by Zimmer
    Ok I have recently run into some problems and was hoping you guys could help; 1) On Chrome sometimes when I play a video (even on Youtube) the audio won't work (yet other apps audio will work) but after pressing the play button (pausing and unpausing the video) it finally works but if I pause the video and click play it goes back to not working until I re-do that process. 2) When I go to play videos in firefox or go to grooveshark it says I don't have flash; but I do and when I go to install flash it says I have the LAST version for linux but flash works on Chrome fine (well except the audio problem above which annoys me to no end!)

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  • No sound while playing multi-media in Ubuntu 12.04 for XPS15

    - by ved2254
    I have an XPS15 laptop, core i5, 8GB ram. Whenever I login my laptop I here the startup bongo sound. But my sound system just doesn't play anything, may it be a short audio clip or a movie. Output of lshw -c multimedia is : WARNING: you should run this program as super-user. *-multimedia description: Audio device product: 6 Series/C200 Series Chipset Family High Definition Audio Controller vendor: Intel Corporation physical id: 1b bus info: pci@0000:00:1b.0 version: 05 width: 64 bits clock: 33MHz capabilities: bus_master cap_list configuration: driver=snd_hda_intel latency=0 resources: irq:51 memory:f1c00000-f1c03fff WARNING: output may be incomplete or inaccurate, you should run this program as super-user. Headphones work just fine but there is no sound from the speakers. Is it a bug in multi-media players or ALSA?

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  • Networkmanager in systray gone and sound not working after update 13.10

    - by rubo77
    After upgrading my Xubuntu 13.04 to 13.10 I have no sound. I still have sound if I start VLC with sudo mpg123 test.mp3 So it seemd there is a right problem EDIT after adding myself to the group audio with adduser myself audio I could play sounds again from the desktop with VLC But one problem remaining: The systray, usually looking like this: is not working anymore. No audio-settings and no network-manager in the taskbar in XFCE: there is just one small box with nothing in it. When I install stalonetray, There I see the status of wicd and all the other statuses, so the systray seems to be broken.

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  • Which of VLC's dependencies causes sound device detection?

    - by Raphael
    I am setting up a headless music server based on the minimal Ubuntu image. After having installed the packages openssh-server,pulseaudio, libmad0,flac,liboff0,libid3tag0,libvorbis0a,ffmpeg, mpd,mpc,mpdscribble, paman,paprefs,pavumeter neither my internal soundcard nor the external DAC where detected by pulseaudio, that is pactl list did only list the dummy devices. Several reboots did not change that. The hardware devices are detected properly: ~$ lsusb | grep Texas Bus 002 Device 002: ID 08bb:2706 Texas Instruments Japan ~$ lspci | grep Audio 00:1b.0 Audio device: Intel Corporation N10/ICH 7 Family High Definition Audio Controller (rev 02) Following a hunch, I installed vlc with all dependencies. After a reboot, both devices are detected! ~$ pactl list | grep "Sink: alsa_output" Monitor of Sink: alsa_output.pci-0000_00_1b.0.analog-stereo Monitor of Sink: alsa_output.usb-Burr-Brown_from_TI_USB_Audio_DAC-00-DAC.analog-stereo Now I would like to remove VLC again but keep the devices. The question is: which of the many dependencies of VLC enables proper device detection? And why on earth is it not a dependency of pulseaudio?

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  • Can I get the Waves Maxx speaker effects to work in Ubuntu?

    - by Rafael
    I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

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  • Cheap sound on speakers - Dell XPS L502X

    - by Rafael
    I have a new machine that comes with JBL 2.1 Speakers with Waves Maxx Audio 3. On Windows it sounds perfect, though in Ubuntu 12.04 I get cheap/sound with simple mp3 files. I have tried a few things on different blogs but no luck so far. Any ideas? aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PCH [HDA Intel PCH], device 0: ALC665 Analog [ALC665 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 1: ALC665 Digital [ALC665 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: PCH [HDA Intel PCH], device 3: HDMI 0 [HDMI 0] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: N700 [Logitech Speaker Lapdesk N700], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0

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  • There's no Sound Mixer menu, missing menu option in Sound Recorder

    - by AlexN
    I am using: -Ubuntu 11.10 -Skype -PS3 Eye Toy camera to input video and sound This setup has been properly working in former Ubuntu releases. To use the mic already built in on the PS3 Eye Toy camera I open de Sound Recorder app (notice: not inside Skype, from inside Skype it is not possible to do this) that is included in Gnome and then I go to FileSound Mixer, from this menu I can choose Gnome to get the input audio from the PS3 Eye Toy, instead of from the Audio-In of the computer. Now in Ubuntu 11.10 this Sound Mixer menu inside Sound Recorder is missing, Gnome says something like this: gnome-volume-control is not installed in the proper directory Note: I have tried this on Unity, Unity 2D, Gnome Classic, Gnome Classic 2D and Gnome Shell. In all of them the problem is the same. What can I do? Basically what I want to do is to be able to tell the computer to get the audio in from the PS3 Camera. Thanks in advance.

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  • Is there now any way to convert mp3 files to m4a or aac 192kbit?

    - by piedro
    Since about two years now I am trying to find a way to convert high quality mp3 files to m4a or aac files with a fixed bitrate 192k. Please don't suggest using another format - i thought this through as far as it goes. The problem here is: ffmpeg obvioulsy can't convert to a higher bitrate than 152k. Even when it says it does so the resulting files still have 152k instead of 192k. ffmpeg also has/had a bug not writing the bitrate into the audio file tags which means when testing you have to calculate the bitrate manually by dividing the filesize by the length of the audio in seconds (resulting in 152k - see above) choosing faac as converter gets me the same results other programs don't work reliably (see this thread Howto convert audio files to *.m4a? I know that this is not an original new problem but I am wondering if there is still no way to convert with ubuntu/kubuntu 12.04 after a lot time passed and I can't find some of the bug issues mentioned in the other thread anymore. So: Is there a solution after all?

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  • Problem with sound in Kubuntu 12.10

    - by Mihkel
    I'm really enjoying Kubuntu 12.10 experience, but the problem starts with sound. It wasn't here before, but today sound sounds garbled and echoed and wrong. It happens in Audacity and VLC. It doesn't happen when I test the sound devices nor when I use Amarok to play the music files (but come on, who uses Amarok to listen to a random music file, it's much more natural to use VLC for that ;-) ) Kubuntu/Phonon recognizes 2 sound devices: 1) RV770 HDMI Audio [Radeon HD 4850/4870] Digital Stereo [HDMI] 2) Built-in Audio Analog Stereo I know it has to use the second option, and it probably does, but that's not the case. What I did find out was that I had to rescan for audio devices in Audacity (and probably select "sysdefault") for it to sound normal. Why does it happen? I've tried following some other questions, but well.

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  • How to Install Linux on my PC

    - by Holic
    Hi i need some help to install the drivers from my pc, on Ubuntu 10.10 i just installed it, and i a newbie on Ubuntu, but i understand a bit of Windows...but i want to try ubuntu and then Maybe change to UBUNTU!!! My hardware: QuadCore Intel Core i7-870, 3266 MHz (24 x 136) Asus P7P55D-E (2 PCI, 3 PCI-E x1, 2 PCI-E x16, 4 DDR3 DIMM, Audio, Gigabit LAN, IEEE-1394) NVIDIA GeForce GTX 480 (1536 MB) nVIDIA HDMI @ nVIDIA GF100 - High Definition Audio Controller VIA VT1828S @ Intel Ibex Peak PCH - High Definition Audio Controller [B-3] DIMM1: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) DIMM3: G Skill F3-12800CL9-2GBRL 2 GB DDR3-1333 DDR3 SDRAM (8-8-8-22 @ 609 MHz) (7-7-7-20 @ 533 MHz) (6-6-6-17 @ 457 MHz) my pc is not connected to the internet with a wire(RJ45) but with a wireless LAn Asus WL-167G-V3(wich i also whant to install if possible) Anything would've help me :) Cheers & Thank you!

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  • 12.04 sound keeps auto-muting when idle

    - by fali
    I just installed 12.04 on an HP8510W. Everything works fine except for one weird behavior which I have noticed. When ever there is no audio playing, the audio mute indicator on the laptop is on. As soon as I start playing a you tube video the mute indicator turns off and I get sound. Here is my pulse audio output which says that the sink is suspended because it is idle: Welcome to PulseAudio! Use "help" for usage information. list-sinks 1 sink(s) available. index: 0 name: <alsa_output.pci-0000_00_1b.0.analog-stereo> driver: <module-alsa-card.c> flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY state: SUSPENDED suspend cause: IDLE I tried running alsamixer, but I don't see the auto-mute option.

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  • Sound issue in Lubuntu

    - by jvsa90
    I'm recently having a problem in my Lubuntu deskptop: sound through the speakers doesn't seem to work. The funny thing is: it works when I plug in my earphones. I've tried to unmute everything with pavucontrol and alsamixer, but everything seems to be OK. $ sudo aplay -l **** Liste der Hardware-Geräte (PLAYBACK) **** Karte 0: Intel [HDA Intel], Gerät 0: HDA Generic [HDA Generic] Sub-Geräte: 0/1 Sub-Gerät #0: subdevice #0 $ lspci -v | grep -A7 -i "audio" 00:1b.0 Audio device: Intel Corporation NM10/ICH7 Family High Definition Audio Controller (rev 02) Subsystem: Acer Incorporated [ALI] Device 034a Flags: bus master, fast devsel, latency 0, IRQ 44 Memory at 58200000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> Kernel driver in use: snd_hda_intel Kernel modules: snd-hda-intel Can anyone guess what's happening? It has worked until recently and it definitely works in my Windows partition.

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  • Generic Content Player?

    - by Jantire
    The general idea on the web appears to be that video/audio are to be separated with plain text. By separated, I mean you have a place that plays video/audio and a place that you read text. This is because it is widely understood that they are vastly different. However, audio and video are just another way of communication, just like text. So why do we separate the two even if they are nearly the same thing? Correct me if I'm wrong but, most tutorials are either plain text how-to's (wiki-style) or visual/auditory instructional videos (YouTube). Why aren't the two combined? Or, if it's already been done can someone reply with the link? This might be bordering off-topic and if it is off-topic then please point me to the right place so it won't be. This might also appear to be an obvious question, however I'm not sure if this subject has really been deeply thought-out by more than a few individuals.

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  • 12.04 - sound is laggy when running games through Wine

    - by orzechowskid
    Lenovo U400 Wine 1.5.5 Ubuntu 12.04 with all updates applied I'm experiencing severe (~500ms) audio lag in all games run in Wine. Portal 2, Half-Life, World of Goo, and Fallout are all exhibiting this problem. When I run winecfg though and click the "Test Sound" button at the bottom of the Audio tab, the sound effect appears to play immediately. So I'm not sure what's going on. I don't think it's a problem with PulseAudio by itself since totem videos and Youtube clips both play in perfect sync. Anyone have any ideas on where to start fixing this? thanks! (edit: I thought this was limited to Steam games but I installed a non-Steam game and I now see that's not the case. I get audio lag in other apps too.)

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  • Compress Large Video Files with DivX / Xvid and AutoGK

    - by DigitalGeekery
    Have you ever recorded home video on a camcorder only to find the video size is enormous? What if you wanted to share a video clip on YouTube or another video sharing site, but the file size was bigger than the maximum upload size? Today we’ll look at a way to compress certain video files, such as MPEG and AVI, with Auto Gordian Knot (AutoGK). AutoGK is a free application that runs on Windows. It supports Mpeg1, Mpeg2, Transport Streams, Vobs, and virtually any codec used for an .AVI file. AutoGK will accept as input the following file types: MPG, MPEG, VOB, VRO, M2V, DAT, IFO, TS, TP, TRP, M2T, and AVI. Files are output as .AVI files and are converted using the DivX or XviD codecs. Installing and Using AutoGK Download and install AutoGK (link below) Open the AutoGK. You’ll need to navigate a few wizard screens, but you can just accept the defaults.   Choose your video file by clicking on the folder to the right of the Input file text box.   Browse for and select your video file and click “Open.”   For this example, we’ll be working with an .AVI file that’s 167MB in size.   The output file is copied into the same directory as the input file by default, but you can change this if you choose. If the input file is also .AVI, AutoGK will append an _agk to the output file so that the original is not overwritten. Next, you’ll see any audio tracks listed. You can unselect the check box if you’d like to remove the audio track. You can choose one of the Predefined size options… Or, select a Custom size in MB or Target Quality in percentage. For our example, we’ll be compressing our 167MB file to 35MB. Click on Advanced Settings. Here you can choose your codec, if you have a preference, as well as output resolution and output audio. If you’d like to use the DivX codec, you’ll need to download and install it separately. (See link below) Typically you’ll want to keep the defaults. Click “OK.” Now you’re ready to add your file conversion job to the Job queue. Click Add Job to add it to the queue. You can add multiple files conversions to the job queue and  convert them in one batch. Click Start to begin the conversion process. The process will begin. You’ll be able to see the progress in the Log window on the bottom left. When the conversion is complete you’ll see a “Job finished” and the total time in the log window.   Check your output file to see it’s compressed size. Test your video just to make sure the output quality is satisfactory.   Note:  Conversion times can vary greatly depending on the size of the file and your computer hardware. Files that are several GBs in size may take several hours to compress. AutoGK is no longer being actively developed but is still a wonderful DivX/XviD conversion tool. It can also be used to compress and convert non-copy protected DVDs. Downloads AutoGordianKnot DivX (optional) Similar Articles Productive Geek Tips Use Your Mac Mini as a Media Server Part 2Make Disk Cleanup Compress Older(or Newer) Files on XPMysticgeek Blog: Exclusive Look Inside Vreel – Including Interview With Vreel Founder!Friday Fun: Watch HD Video Content with MeevidConvert a DVD Movie Directly to AVI with FairUse Wizard 2.9 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 DVDFab 6 Revo Uninstaller Pro Registry Mechanic 9 for Windows PC Tools Internet Security Suite 2010 Penolo Lets You Share Sketches On Twitter Visit Woolyss.com for Old School Games, Music and Videos Add a Custom Title in IE using Spybot or Spyware Blaster When You Need to Hail a Taxi in NYC Live Map of Marine Traffic NoSquint Remembers Site Specific Zoom Levels (Firefox)

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  • Use DivX settings to encode to mp4 with ffmpeg

    - by sjngm
    I'm used to use VirtualDub to encode a video to AVI container with DivX-codec (and MP3 for audio). Now I'm planning to use ffmpeg to encode videos to MP4 container with h264-codec. What I've figured out is that I need to use libx264 and one of those presets to make anything work. However, I'm amazed about the video bitrate ffmpeg uses for encoding. What I currently have is this little batch file: @ECHO OFF SETLOCAL SET IN=source.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-fpre "%FFMPEG_PATH%\presets\libx264-lossless_slow.ffpreset" SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %VIDEO% %PRESET% test.mp4 ENDLOCAL With this I tell ffmpeg to use 1978k as the bitrate, but ffmpeg uses 15000k+! I tried other presets, but they don't use my specified bitrate. Here are the presets I have: libx264-baseline.ffpreset libx264-ipod320.ffpreset libx264-ipod640.ffpreset libx264-lossless_fast.ffpreset libx264-lossless_max.ffpreset libx264-lossless_medium.ffpreset libx264-lossless_slow.ffpreset libx264-lossless_slower.ffpreset libx264-lossless_ultrafast.ffpreset ffmpeg version: FFmpeg git-N-29181-ga304071 libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.120. 0 / 52.120. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 79. 0 / 1. 79. 0 libswscale 0. 13. 0 / 0. 13. 0 Note that I don't use the latest version as it has problems with spaces in filenames. Here's what seems to be the full parameter list DivX 6.9.2 uses: -bvnn 1978000 -vbv 218691200,100663296,100663296 -dir "C:\Users\sjngm\AppData\Roaming\DivX\DivX Codec" -w -b 1 -use_presets=1 -preset=10 -windowed_fullsearch=2 -thread_delay=1 What command line parameters would that be for ffmpeg? EDIT: Going with slhck's suggestion I tried a new 32-bit version. I have no idea if that is 0.9 or newer, I can't find that info. ffmpeg version N-36890-g67f5650 libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 I reworked my batch file to look like this (interestingly enough I can't find parameter -vprofile in the documentation): @ECHO OFF SETLOCAL SET IN=VTS_01_1.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-vprofile high -preset veryslow SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %PRESET% %VIDEO% test.mp4 ENDLOCAL I see that it now uses the bitrate properly (thanks to LongNeckbeard for pointing out that the lossless-stuff ignores the bitrate!). Just in case you wonder how I came up with the 1978000, I'm using this formula which I found valid for DivX-files (I'm guessing the bitrate won't change that much for h264): width * height * 25 * 0.22 / 1000 I'm not sure if the 0.22 correlates with the CRF somehow. Overall I forgot to say the I will use a two-pass scenario, which is why I don't use the CRF here. I will try to read more about this. Currently I'm just trying to get something running that shows me that I'm doing something right (ffmpeg isn't the easiest tool to understand ;)). C:\Program Files (x86)\ffmpeg\ffmpeg.exe" -i VTS_01_1.avs -acodec libmp3lame -ab 128000 -vcodec libx264 -vb 1978000 -vprofile high -preset veryslow test.mp4 The output is now: ffmpeg version N-36890-g67f5650 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 16 2012 21:57:13 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, avs, from 'VTS_01_1.avs': Duration: 00:58:46.12, start: 0.000000, bitrate: 0 kb/s Stream #0:0: Video: rawvideo (YV12 / 0x32315659), yuv420p, 576x448, 77414 kb/s, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s File 'test.mp4' already exists. Overwrite ? [y/N] y w:576 h:448 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 05A2C400] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 05A2C400] profile High, level 3.1 [libx264 @ 05A2C400] 264 - core 120 r2120 0c7dab9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1978 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf53.30.100 Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 576x448, q=-1--1, 1978 kb/s, 25 tbn, 25 tbc Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 48000 Hz, 2 channels, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:1 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 3 fps= 1 q=22.0 size= 39kB time=00:00:00.04 bitrate=8063.8kbits/ frame= 8 fps= 2 q=22.0 size= 82kB time=00:00:00.24 bitrate=2801.3kbits/ frame= 13 fps= 3 q=23.0 size= 120kB time=00:00:00.44 bitrate=2229.5kbits/ frame= 16 fps= 4 q=23.0 size= 147kB time=00:00:00.56 bitrate=2156.7kbits/ frame= 20 fps= 4 q=22.0 size= 175kB time=00:00:00.72 bitrate=1987.4kbits/ : video:4387kB audio:273kB global headers:0kB muxing overhead 0.260038% [libx264 @ 05A2C400] frame I:2 Avg QP:19.53 size: 29850 [libx264 @ 05A2C400] frame P:76 Avg QP:22.24 size: 19541 [libx264 @ 05A2C400] frame B:359 Avg QP:25.93 size: 8210 [libx264 @ 05A2C400] consecutive B-frames: 0.5% 0.5% 0.0% 8.2% 17.2% 52.2% 16.0% 5.5% 0.0% [libx264 @ 05A2C400] mb I I16..4: 5.4% 75.3% 19.3% [libx264 @ 05A2C400] mb P I16..4: 1.3% 16.5% 2.2% P16..4: 36.3% 28.6% 12.7% 1.8% 0.2% skip: 0.4% [libx264 @ 05A2C400] mb B I16..4: 0.4% 3.8% 0.3% B16..8: 40.0% 18.4% 4.7% direct:18.5% skip:13.9% L0:45.4% L1:38.1% BI:16.5% [libx264 @ 05A2C400] final ratefactor: 20.35 [libx264 @ 05A2C400] 8x8 transform intra:83.1% inter:68.5% [libx264 @ 05A2C400] direct mvs spatial:99.2% temporal:0.8% [libx264 @ 05A2C400] coded y,uvDC,uvAC intra: 64.9% 83.4% 49.2% inter: 49.0% 50.4% 4.4% [libx264 @ 05A2C400] i16 v,h,dc,p: 25% 22% 27% 26% [libx264 @ 05A2C400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 10% 7% 23% 9% 10% 10% 10%10% 13% [libx264 @ 05A2C400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 11% 13% 9% 12% 11% 10% 9% 12% [libx264 @ 05A2C400] i8c dc,h,v,p: 42% 28% 16% 14% [libx264 @ 05A2C400] Weighted P-Frames: Y:18.4% UV:7.9% [libx264 @ 05A2C400] ref P L0: 29.1% 11.3% 15.7% 7.3% 6.9% 4.9% 5.1% 3.4%3.9% 2.7% 2.8% 1.8% 1.7% 1.2% 1.4% 0.9% [libx264 @ 05A2C400] ref B L0: 68.8% 11.4% 5.5% 2.9% 2.3% 1.9% 1.5% 1.1%1.1% 1.0% 0.9% 0.7% 0.5% 0.3% 0.1% [libx264 @ 05A2C400] ref B L1: 91.9% 8.1% [libx264 @ 05A2C400] kb/s:2055.88 As far as I'm concerned it doesn't look that bad to me.

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  • friending istream operator with class

    - by user1388172
    hello i'm trying to overload my operator >> to my class but i ecnouter an error in eclipse. code: friend istream& operator>>(const istream& is, const RAngle& ra){ return is >> ra.x >> ra.y; } code2: friend istream& operator>>(const istream& is, const RAngle& ra) { is >> ra.x; is >> ra.y; return is } Both crash and i don't know why, please help. EDIT: ra.x & ra.y are both 2 private ints of my class; Full error: error: ..\/rightangle.h: In function 'std::istream& operator>>(std::istream&, const RAngle&)': ..\/rightangle.h:65:12: error: ambiguous overload for 'operator>>' in 'is >> ra.RAngle::x' ..\/rightangle.h:65:12: note: candidates are: c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:122:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__istream_type& (*)(std::basic_istream<_CharT, _Traits>::__istream_type&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:122:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__istream_type& (*)(std::basic_istream<char>::__istream_type&) {aka std::basic_istream<char>& (*)(std::basic_istream<char>&)}' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:126:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__ios_type& (*)(std::basic_istream<_CharT, _Traits>::__ios_type&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>, std::basic_istream<_CharT, _Traits>::__ios_type = std::basic_ios<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:126:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__ios_type& (*)(std::basic_istream<char>::__ios_type&) {aka std::basic_ios<char>& (*)(std::basic_ios<char>&)}' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:133:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::ios_base& (*)(std::ios_base&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:133:7: note: no known conversion for argument 1 from 'const int' to 'std::ios_base& (*)(std::ios_base&)' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:241:7: note: std::basic_istream<_CharT, _Traits>& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__streambuf_type*) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__streambuf_type = std::basic_streambuf<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:241:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__streambuf_type* {aka std::basic_streambuf<char>*}' ..\/rightangle.h:66:12: error: ambiguous overload for 'operator>>' in 'is >> ra.RAngle::y' ..\/rightangle.h:66:12: note: candidates are: c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:122:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__istream_type& (*)(std::basic_istream<_CharT, _Traits>::__istream_type&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:122:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__istream_type& (*)(std::basic_istream<char>::__istream_type&) {aka std::basic_istream<char>& (*)(std::basic_istream<char>&)}' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:126:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__ios_type& (*)(std::basic_istream<_CharT, _Traits>::__ios_type&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>, std::basic_istream<_CharT, _Traits>::__ios_type = std::basic_ios<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:126:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__ios_type& (*)(std::basic_istream<char>::__ios_type&) {aka std::basic_ios<char>& (*)(std::basic_ios<char>&)}' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:133:7: note: std::basic_istream<_CharT, _Traits>::__istream_type& std::basic_istream<_CharT, _Traits>::operator>>(std::ios_base& (*)(std::ios_base&)) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__istream_type = std::basic_istream<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:133:7: note: no known conversion for argument 1 from 'const int' to 'std::ios_base& (*)(std::ios_base&)' c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:241:7: note: std::basic_istream<_CharT, _Traits>& std::basic_istream<_CharT, _Traits>::operator>>(std::basic_istream<_CharT, _Traits>::__streambuf_type*) [with _CharT = char, _Traits = std::char_traits<char>, std::basic_istream<_CharT, _Traits>::__streambuf_type = std::basic_streambuf<char>] <near match> c:\mingw\bin\../lib/gcc/mingw32/4.6.1/include/c++/istream:241:7: note: no known conversion for argument 1 from 'const int' to 'std::basic_istream<char>::__streambuf_type* {aka std::basic_streambuf<char>*}''

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  • Sound Effects/Manipulation?

    - by Adam
    Hello, I am creating an android app that basically records an applies an "Effect" on the audio track then plays it back. I got my app to record an play back but I am stuck an not sure where do go from here. I have been Googling for days now trying to find a open source audio library or some way to change the audio after I record it. I currently have it setup to play back using SoundPool an I't lets me speed up an slow down the audio. I would like to do things like change pitch an add echo etc. I will appreciate any responses because I am totally stumped right now. Thanks Adam

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  • VU meter implementaion in iphone

    - by Sreelal
    Hi, I am developing an aplication for iphone which records audio and save that audio file .I need to create a UI similar to that in Voice Memo app with VU meter .I implemented codes to record audio,but i have no idea about VU meter implementation.Looking forward for a reply ......Thanks in advance

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  • avaudioplayer interferes with mpmovieplayer on ipad

    - by user175826
    my app plays video and audio. however, i have a problem where once i play an audio file using avaudioplayer, the video refuses to play. when i play the video first, everything is fine. but if the audio is played first, any time i try to play the video it simply pops up the video player but will not play the actual video (you can use the scroller to go to any point in the video, but no playback will happen). this issue does not come up on the iphone, nor on the ipad simulator. clearly there is some resource conflict here, probably related to the audio, and i'd welcome some input on how to address it.

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