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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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  • 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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  • How can I stop Ubuntu from automatically unmounting Samba shares?

    - by Billy ONeal
    I have some music files I'd like to listen to sitting on a Samba share. I added this share via the Ubuntu GUI (Places - Connect to server...), and everything worked just fine. However, despite the fact that my music file is playing from this location, after I've not touched the location using the Nautilus GUI, Ubuntu/GNOME decides that I'm not using the share anymore and terminates the connection. Thus, my music stops playing and Rhythmbox is unhappy with me. Simply clicking on the new shortcut the "Connect to server..." bit created for me immediately makes the files come back again and allows me to restart the music playing. How can I have Ubuntu not automatically dismount samba shares?

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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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  • 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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  • 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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  • Victory rewards in digital CCG

    - by Nils Munch
    I am currently polishing a digital CCG where people can play against friend and random opponents in a classical Magic the Gathering-like duel CCG. I plan to award the players with 20 ingame currency units (lets call them gold) for each hour they are playing, 50 for each day they are playing and X for each victory. Now, the X is what I am trying to calculate here, since I would prefer keeping the currency to a certain value, but also with to entice the players to battle. I could go with a solid figure, say 25, for beating up an opponent. But that would result in experienced players only beating up newly started players, making the experience lame for both. I could also make a laddered tier, where you start at level 1, and raise in level as you defeat your opponents, where winning over a player awards you his level x 2 in gold. Which would you prefer if you were playing a game like this. There is no gold-based scoreboard, but the gold is used to purchase new cards along the way.

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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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  • Does anyone use 12.04 with HD 6950?

    - by Midori
    Back when I first tried to switch from Windows to Ubuntu (6-7 months ago) I had several graphics related problems. The VGA was overheating, graphic glitches, video playing issues, slow/choppy window movements etc. After 2-3 reinstalls, hours of playing with different drivers/settings I decided to return to Windows :\ Now I want to give it another shot. But before I start to format, partitioning, backup and other time consuming stuff I wanted to ask if anyone using the 12.04 with HD 6950. Can I utilize the full potential of it, or the drivers are still not good enough? I know that Linux isn't meant to play games in the first place, but the games I playing (SC2, BLC, HoN, DotA 2) are working with wine as far as I know (or aren't?) and I can't find any reason not to switch from Windows if I can utilize the full potential of my config in Ubuntu. So anyone who got experiences with this VGA in Ubuntu please reply. Thanks in advance :)

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  • Add New Features to WMP with Windows Media Player Plus

    - by DigitalGeekery
    Do you use Windows Media Player 11 or 12 as your default media player? Today, we’re going to show you how to add some handy new features and enhancements with the Windows Media Player Plus third party plug-in. Installation and Setup Download and install Media Player Plus! (link below). You’ll need to close out of Windows Media Player before you begin or you’ll receive the message below. The next time you open Media Player you’ll be presented with the Media Player Plus settings window. Some of the settings will be enabled by default, such as the Find as you type feature. Using Media Player Plus! Find as you type allows you to start typing a search term from anywhere in Media Player without having to be in the Search box. The search term will automatically fill in the search box and display the results.   You’ll also see Disable group headers in the Library Pane.   This setting will display library items in a continuous list similar to the functionality of Windows Media Player 10. Under User Interface you can enable displaying the currently playing artist and title in the title bar. This is enabled by default.   The Context Menu page allows you to enable context menu enhancements. The File menu enhancement allows you to add the Windows Context menu to Media Player on the library pane, list pane, or both. Right click on a Title, select File, and you’ll see the Windows Context Menu. Right-click on a title and select Tag Editor Plus. Tag Editor Plus allows you to quickly edit media tags.   The Advanced tab displays a number of tags that Media Player usually doesn’t show. Only the tags with the notepad and pencil icon are editable.   The Restore Plug-ins page allows you to configure which plug-ins should be automatically restored after a Media Player crash. The Restore Media at Startup page allows you to configure Media Player to resume playing the last playlist, track, and even whether it was playing or paused at the time the application was closed. So, if you close out in the middle of a song, it will begin playing from that point the next time you open Media Player. You can also set Media Player to rewind a certain number of seconds from where you left off. This is especially useful if you are in the middle of watching a movie. There’s also the option to have your currently playing song sent to Windows Live Messenger. You can access the settings at any time by going to Tools, Plug-in properties, and selecting Windows Media Player Plus. Windows Media Plus is a nice little free plug-in for WMP 11 and 12 that brings a lot of additional functionality to Windows Media Player. If you use Media Player 11 or WMP 12 in Windows 7 as your main player, you might want to give this a try. Download Windows Media Player Plus! Similar Articles Productive Geek Tips Install and Use the VLC Media Player on Ubuntu LinuxFixing When Windows Media Player Library Won’t Let You Add FilesMake VLC Player Look like Windows Media Player 10Make VLC Player Look like Windows Media Player 11Make Windows Media Player Automatically Open in Mini Player Mode 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 Xobni Plus for Outlook All My Movies 5.9 CloudBerry Online Backup 1.5 for Windows Home Server Snagit 10 Easily Create More Bookmark Toolbars in Firefox Filevo is a Cool File Hosting & Sharing Site Get a free copy of WinUtilities Pro 2010 World Cup Schedule Boot Snooze – Reboot and then Standby or Hibernate Customize Everything Related to Dates, Times, Currency and Measurement in Windows 7

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  • Red Meat's Music is Rare - and Well Done

    - by Oracle OpenWorld Blog Team
    By Karen Shamban The blogger has questions; San Francisco-based country band Red Meat has answers. Although we forgot to ask how they got their band name, dang it. Read on and enjoy the honesty and insight. Q. What do you like best about performing in front of a live audience?A. Probably just having fun and entertaining the audience. We've been together for almost two decades, and in that time we've played for crowds of five people, and for crowds of more than 15,000. Both are equally important to us, and just as fun. We turn Jill and Smelley loose on the between-songs repartee, and let the songs shine through. On the best night, we feed on the audience's love and vice-versa. It's emotional vampirism of the best sort. [Blogger's note: now that whole "red meat" thing is starting to make sense ...] Q. Do you prefer smaller, intimate venues or larger, louder ones? Why?A. We love both. Whether it's a chance to connect with a small room or huge audience, we always try to hit 'em between the eyes! Q. What about your fans surprises you?A. Since we've been together for so long, we're pretty much on our third generation of fans now. We're excited that the Bakersfield sound has that same effect on the new, younger fans as it did on the punk rockers that we played to 20 years ago. And we still see them at our shows too! Q. What about your live act surprises your fans?A. For people who haven't seen Red Meat before, they may be dragged to a show thinking they don't like country music. But they're surprised to hear it done in a way that excites them so much. We get a lot of first-timers coming up to us after a performance and asking, "Wait, THAT'S what country music can sound like?" Q. There are going to be a lot of technical people (you could call them geeks) in the Oracle crowd - what are they going to love about your performance?A. Just what everyone loves about a Red Meat show - the chance to drink beer, dance, get rowdy, and have a great time. Q. Have you been on tour recently? If so, what do you like about touring, and what do you dislike?A. Actually, we're going to be coming off the road immediately into the Oracle OpenWorld Music Festival, having just played some Texas dates. On tour, we love playing for fans who don't get to see us as often as our California fans do. And food. Most of our conversations in the van center around food. Q. Ever think about playing another kind of music? If so, what, and why?A. Our tastes and influences in the band run all over the place. Obviously we love the Bakersfield artists - Buck Owens, Merle Haggard, Dwight Yoakam - but we love other types of roots music as well, along with the Beatles, NRBQ, MC5, punk/new wave, and countless bar bands that we've had the privilege of playing with through the years. But as far as playing a different kind of music as Red Meat? Nah. We love what we're doing. Q. What are the top three things people should know about your music?A1. Country music, done right, has unlimited soul.A2. Red Meat is a modern band, playing original material, with a great debt to the Bakersfield sound of Buck Owens and Merle Haggard.A3. It's FUN. More details on the Festival and the band: Oracle OpenWorld Music Festival Red Meat

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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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  • problem in AudioStreaming in Iphone Sdk?

    - by senthilmuthu
    I am using sample code of AudioStream.zip,but when i use this to play Mp3 file , it gives wrong total amount of playing time(after played completely through streaming).... i checked through downloading that Mp3 file into Document Directory and played in Itune it exactly is played for 2.10 seconds.but in streaming through that code(- (double)progress method) gives total playing time only 2.3 sec, is there any sample code for AudioStreaming except that one to give right Total playing Time?

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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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  • How do I force one method to be executed before another method?

    - by RexOnRoids
    I've got 2 methods. One method starts playing an audio file (.mp3), the other method updates a UIToolBar to show a button (PLAY or PAUSE). These two methods are called in the following order: //Adds some UIBarButtonItems to a UIToolBar [self togglePlayer]; //Uses AVAudioPlayer [audioPlayer play]; I call the methods in the above order so that the (pause) button will be shown at the time the song starts playing. But, the problem is that the song starts playing first, and the UIToolBar remains unchanged for quite a while (from 2 to 5 secs) until the button is added and shown. What I want is for the button to be shown at the same time the song starts playing (i.e. NO DELAY). Is there any way to do this?

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  • Sound File editing in Objective C

    - by Biranchi
    Hi All, I am able to record and create audio files using AudioFileCreateWithURL in the AudioToolbox Framework. I want to figure out if there is any way to edit the .caf sound files. I want to insert another recoreded audio inside the main audio file. Any thoughts or suggestions how to proceed ?? Thanks.

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