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  • best storage option for recording live streaming video

    - by Alchemical
    We're creating a new web site that includes video chat capabilities. Wowza Server is being used for the streaming media. We would like the capability for users to record their video chats in certain circumstances. Is it feasible to record the videos to the same server running Wowza? Or performance-wise, would it make more sense to store them on another server? I understand SANs are popular as well, but I'm a little concerned about cost for those as we are on a fairly tight budget. Currently we have two servers with pretty decent RAID cnofigurations for storage, one has 14TB and the other 6TB. Mostly concerned if doing the recording on the same server as the streaming server (or web server), could significantly adversely affect that server's primary function.

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  • LIVE Video Streaming with Nginx + PHP-FPM / Process Timeout

    - by user3393046
    I have a live video streaming in my server using nginx + php. the php file reas a live streaming and it directly sends it to the client. I have only one problem. The problem is that i want each request to be in a new process of php-fpm. In a few words i don't want to have idle timeout for a process but instead i want them to close instant when a request is being closed. With idle timeout i have huge problems which are hard to explain at the moment but i'm really sure that if i disable the idle timeout everything will be perfect. Is there any way to do this? I'm using on demand php-fpm

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  • Watch 53rd Grammy Awards 2011 Live Stream & Follow The Event On Facebook & Twitter

    - by Gopinath
    Grammy Awards is the biggest music awards ceremony that honours music genres. The 53rd Grammy Awards function will be held on February 13, 2011 at Staples Center in Los Angeles. The star studded musical event will be telecasted live on CBS TV between 8pm ET until 11:30pm ET. Behind The Scenes Live Stream On YouTube and Grammy.com Grammy, YouTube and TurboTax has teamed up to provide live stream of behind the scene coverage of key events starting at 5pm Feb 11th and runs through February 13th. Note that this stream will not broadcast the 3 hour long award presentation. Kind of bad, except the award presentation rest of the ceremony is streamed here.  The video stream is embedded below Catch live behind-the-scenes coverage of key events during GRAMMY week. Before the show, you can watch the Nominees Reception, Clive Davis Pre-GRAMMY Gala, the red carpet and the pre-telecast ceremony. During the show, go backstage to see the winners after they accept their award! You can catch the same stream on Grammys’s YouTube channel as well as on Grammy.com website Follow Grammy Awards On Facebook & Twitter The Grammy Awards has an official Facebook and Twitter account and you can follow them for all the latest information The Grammy’s Facebook Page The Grammy’s Twitter Page Grammy Live Streaming on UStream & Justin.tv Even though there is no official source that stream live of The Grammy Awards presentation ceremony, there will be plenty of unofficial sources on UStream and Justin.tv. So search UStream and Justin.tv for live streaming of The Grammy’s. TV Channels Telecasting The Grammy Awards 2011 Here is the list of TV channels in various countries offering live telecast of The Grammy Awards 2011. We keep updating this section as and when we get more information USA – CBS Television Network  – February 13, 2011 India – VH1 TV – February 14  2011 , 6:30 AM United Kingdom – ITV2  – 16 February 2011, 10:00PM – 12:00AM This article titled,Watch 53rd Grammy Awards 2011 Live Stream & Follow The Event On Facebook & Twitter, was originally published at Tech Dreams. Grab our rss feed or fan us on Facebook to get updates from us.

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  • How can I force fullscreen on my right screen?

    - by Boris
    I have 2 screens: 17" on left and 22" screen on right side. When I watch a movie with VLC and want to watch on full-screen on my 22" screen (the right one), I move my VLC window in right screen and then click on full-screen = no problem. BUT when using Firefox and watching streaming video, watching on full-screen is always on 17" screen (left one). How can I force full-screen to be on my 22" screen (right one) ?

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  • Data structures for a 2D multi-layered and multi-region map?

    - by DevilWithin
    I am working on a 2D world editor and a world format subsequently. If I were to handle the game "world" being created just as a layered set of structures, either in top or side views, it would be considerably simple to do most things. But, since this editor is meant for 3rd parties, I have no clue how big worlds one will want to make and I need to keep in mind that eventually it will become simply too much to check, handling and comparing stuff that are happening completely away from the player position. I know the solution for this is to subdivide my world into sub regions and stream them on the fly, loading and unloading resources and other data. This way I know a virtually infinite game area is achievable. But, while I know theoretically what to do, I really have a few questions I'd hoped to get answered for some hints about the topic. The logic way to handle the regions is some kind of grid, would you pick evenly distributed blocks with equal sizes or would you let the user subdivide areas by taste with irregular sized rectangles? In case of even grids, would you use some kind of block/chunk neighbouring system to check when the player transposes the limit or just put all those in a simple array? Being a region a different data structure than its owner "game world", when streaming a region, would you deliver the objects to the parent structures and track them for unloading later, or retain the objects in each region for a more "hard-limit" approach? Introducing the subdivision approach to the project, and already having a multi layered scene graph structure on place, how would i make it support the new concept? Would you have the parent node have the layers as children, and replicate in each layer node, a node per region? Or the opposite, parent node owns all the regions possible, and each region has multiple layers as children? Or would you just put the region logic outside the graph completely(compatible with the first suggestion in Q.3) When I say virtually infinite worlds, I mean it of course under the contraints of the variable sizes and so on. Using float positions, a HUGE world can already be made. Do you think its sane to think beyond that? Because I think its ok to stick to this limit since it will never be reached so easily.. As for when to stream a region, I'm implementing it as a collection of watcher cameras, which the streaming system works with to know what to load/unload. The problem here is, i will be needing some kind of warps/teleports built in for my game, and there is a chance i will be teleporting a player to a unloaded region far away. How would you approach something like this? Is it sane to load any region to memory which can be teleported to by a warp within a radius from the player? Sorry for the huge question, any answers are helpful!

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  • Is it possible to transcode audio in C# using DirectSound?

    - by Robert Davis
    I want to transcode a lot of audio from its source format to PCM without resampling or messing with the sample size. I figure if Windows Media Player can play the file and it doesn't use a legacy ACM codecs it must be using DirectSound to do so (this is on Windows XP and Windows Server 2k3). So is it possible to access DirectSound from C# and do so? I've tried searching the web but all the examples have been about playback which I have no interest in doing.

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  • What's the best way of playing media files (esp. audio) with Mono/C#?

    - by supercheetah
    I'm trying to create something that will be playing some sound and music for some things in Mono+C#, but I'm not sure what the best thing will be for that. I'm trying to make it usable with things like Ogg Vorbis, MP3s, and wave files. My primary platform will be Linux, although a cross platform solution would be nice. Anyone have any suggestions for libraries for playing audio files?

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  • Where to start learning about audio or video codecs ?

    - by Vamsi
    Hi, I am very much confused to know what happens inside the codecs. I want to learn about the elements inside audio encoders and decoders. Would be very happy if you can provide me some links where i can find some good study material. Thanks precisely i would like to know how the codec parses the a media file.

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  • Compiling a list of audio + video players (flash / javascript / otehr) that I can embed into a websi

    - by FiveTools
    I'm compiling a list of audio + video players (flash / javascript / other) that I can embed into a website. flowplayer: http://flowplayer.org/ jw player: http://www.longtailvideo.com/players/ premium beat: http://www.premiumbeat.com/flash_resources/free_flash_music_player/ xspf web player: http://musicplayer.sourceforge.net/ yahoo media player: http://mediaplayer.yahoo.com/ any popular ones I'm missing? (anyone know if I can skin / customize any of them to operate similar to the Windows vista volume control?)

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  • Where can I get a splitter to connect a device with a single 3.5 mm plug into the audio input/output jacks on my laptop?

    - by XinJeisan
    I recently bought the :Hype Retro Handset for Mobile Phone" -- its just a device that looks like a handset to use when chatting on a computer or mobile phone that plugs into the phone/computer with a single 3.5 mm plug. I was hoping to use it on my windows 7 Toshiba laptop. I can hear audio fine through the handset but what I'm saying is not being picked up on the handset. On the box it says "some phones and computers may need additional adapters," so I'm hoping it is possible to get a splitter or something for this to work properly. I did email the parent company (http://dglusa.com/) but I haven't heard from them, and, looking over their website, I doubt I will. I also went to the local radio shack, and the guy said I needed a splitter, but he didn't know where to get one. I can find the kind of splitter I think I need online, but I'm unsure whether they are just for output or can also do input/output.

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  • Flash in browsers does not play sound accurately using Pulse network audio

    - by Dave M G
    I use PulseAudio to send sound over the LAN to an audio server. When playing any Flash media in Firefox or Chrome, the sound flutters, as if the volume were going up and down every second. The problem does not exhibit with any other software, and I think it's specific to how Flash interacts with my sound set up. How do I get Flash to play nice with the PulseAudio network sound server? Update I have discovered that I can stop the sound fluttering if I follow these steps: Start a Flash video Run pulseaudio --kill on the server Wait about 7 seconds After this, the PulseAudio server automatically respawns, and the sound in the Flash video is perfect. The problem now, though, is that I have to do this every time I start a Flash video. This is obviously not desireable. So, the question is, how do I make whatever it is that makes the sound work when I go through these steps stick so that I don't have to do them? Also, I've uploaded some PulseAudio log output to Pastebin, taken while attempting to play a Flash video, if that helps. I've tried to get logging details from Flash, but despite installing and enabling Flash for debugging, it has not generated any ouput at all. Details I have uploaded an example video of the problem onto Youtube. In the video you can see the opening of a Ted Talk video, and the sound flutters as it plays. The video also stutters while playing back. Here are my sound device output settings:

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  • Jack Audio ubuntu 12.10

    - by Shaneo1
    I used to have Jack Server working with 10.10, 11.04, 11.10 but not 12.04 and now 12.10. I have installed jackd jackd2 qjackctl surfed many forums and even given advice of how to get jack working, but now I am stuck. Tue Nov 27 22:30:46 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:19.960 D-BUS: JACK server could not be started. Sorry Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Tue Nov 27 22:31:19 2012: Starting jack server... Tue Nov 27 22:31:19 2012: JACK server starting in realtime mode with priority 10 Tue Nov 27 22:31:19 2012: [1m[31mERROR: cannot register object path "/org/freedesktop/ReserveDevice1/Audio0": A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to acquire device name : Audio0 error : A handler is already registered for /org/freedesktop/ReserveDevice1/Audio0[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Audio device hw:0,0 cannot be acquired...[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Cannot initialize driver[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: JackServer::Open failed with -1[0m Tue Nov 27 22:31:19 2012: [1m[31mERROR: Failed to open server[0m Tue Nov 27 22:31:21 2012: Saving settings to "/home/shane/.config/jack/conf.xml" ... 22:31:22.047 Could not connect to JACK server as client. - Overall operation failed. - Unable to connect to server. Please check the messages window for more info. Cannot connect to server socket err = No such file or directory Cannot connect to server request channel jack server is not running or cannot be started Can anyone assist?

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  • Double audio cd ripping weirdness

    - by jqno
    Since I installed Ubuntu 12.04, Rhythmbox, Banshee and Sound Juicer have started acting weird around double cd's, and specifically, cd #2 of said double cd. Sometimes, they will show the information of cd #1. Track names, durations, and even count are incorrect. Sometimes, they will first show the tracks for cd #1, then continue onto cd #2 if cd #2 has more tracks than #1. Sound Juicer seems to be unable to find any track durations at all, even for single cd's. Obviously, this is a pain when I'm trying to rip double cd's. And I have a fair number of them, which I want to rip. This happens on both my machines (a slightly aging iMac, and a 1-year-old Sony Vaio). However, on previous versions of Ubuntu, this never happened. All on the same machines. So I suspect 12.04 is using a different lib for extracting audio cd data. Just for kicks, I tried with Linux Mint 13, and there it works correctly, even though it claims to be based on Ubuntu 12.04 and therefore should be using (partially) the same software. So if the Mint guys can fix it, I should be able to do it too, right? So, my question: what changed in 12.04 that could cause this? And more importantly: what can I do to fix it?

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  • How to make pulseaudio and ubuntu detect the same audio device as alsa driver

    - by Kiwy
    I use Ubuntu 14.04 x64 and I use gnome-shell on my laptop. I have a Bose companion 5 (which is basically a USB sound system) and a HDMI port, both does work perfectly when I just boot with the cable plugin. However, when my laptop go to sleep or get unplugged from those two outputs, if I plug back the device, I end up without any hardware detection (only the built-in speakers) from pulse and gnome-shell sound output selector while if I use alsamixer, the device look up and ready. gstreamer-properties allow me to select and test effectively any device but while alsa recognize any device on the run, pulse is not capable of handling things correctly, my question is then: How can I make pulse detect and use the same hardware as alsa, or how to remove completely and gracefully pulseaudio (meaning volume applet running in gnome shell) I don't mind if the project implies to recompile half gnome shell if it implies those audio outputs work all the time. Pulse does not list my soundcard when I use command pactl list cards while the module plug&play for sound card is loaded in pactl list modules. I really don't know what to do, the behavior seems pretty random.

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  • 12.10 no audio via hdmi and video speeds up

    - by jackson
    I have a laptop with an ati radeon 4200, on 12.04 everything worked fine, since upgrading to 12.10 I cannot get sound over the hdmi. When I switch to hdmi audio the video speeds up to about 2x. I can use the speakers in my laptop and watch video via hdmi with no problems. Things I have tried: Various tutorials to install the AMD/ATI drivers, all of which resulted in low graphics mode. Checked that everything is properly set in alsamixer, the sound utility and - installed pavucontrol and checked everything in there. Verified the output from cat /proc/asound/cards looks normal When I initially upgraded there was a plethora of problems which I believe were due to the old proprietary driver still being used but not compatible, after a few hours trying to fix that I decided just to back up and do a fresh install which works great except for the above stated problem. Any help would be greatly appreciated!! Finally hopefully this hasn't already been answered, I have tried a few different searches on the boards and haven't come up with anything. $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC269VB Analog [ALC269VB Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: HDMI 0 [HDMI 0] Subdevices: 0/1 Subdevice #0: subdevice #0

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  • No audio in Google Chrome

    - by Z9iT
    I started with Ubuntu 12.04 Minimal. Then installed only 3 utils sudo apt-get install xorg xinit google-chrome-stable alsa-base alsa-utils alsa-oss I have added google-chrome to .xinitrc file. Used sudo alsamixer to unmute everything using M. Also I am able to hear sound when I run this independently in a terminal sudo aplay /usr/share/sounds/alsa/Front_Center.wav However Google Chrome is not giving any sound output be it on youtube or the same file (/usr/share/sounds/alsa/Front_Center.wav) opened by browsing in chrome. UPDATE : the moment i install some Desktop (display) Manager like gnome or lxde and launch chrome then, the audio is perfect success. However if i kill the xsession and the desktop manager (lxde) AND then start with loading only the chrome (without DM) then again i loose the sound. This makes me wonder that there is something which is not allowing the sound to be loaded into chrome directly, but once the session like lxde loads, then it works flawless. I am thinking that i should rather ask, how to authorize google-chrome to use sound software? Miscellaneous : I am surprised to know that I cannot start google-chrome by sudo command (it asks to be a normal user) && that i cannot start alsamixer as a normal user (i must use sudo alsamixer ) May someone please help what i need to do so that google chrome speaks????

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  • My sound stopped working today, how can I fix it?

    - by Oli
    This seems to be a problem with pulseaudio. I was logged in over VNC on my phone and started playing a video this caused X to crash (as sometimes happens). I restarted and suddenly the sound doesn't work. I have a Intel HDA/Realtek ALC889 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller alsamixer is detecting this just fine. PulseAudio doesn't detect this alsa device so is using auto_null as the default sink (logs below). When I properly kill PulseAudio (tell it not to auto-start) direct ALSA communication with the sound card works just fine. speaker-test, for example, works. So the hardware and ALSA layers are fine IMO. In the logs, it seems that the card might be "busy" but I really don't know how or why it would be now (and never before). Is there an ALSA lock file somewhere that it still there because of my crash? I just ran sudo fuser /dev/snd/* and saw this: oli@bert:~$ sudo fuser /dev/snd/* /dev/snd/controlC0: 1884 /dev/snd/pcmC0D0c: 1884m /dev/snd/timer: 1884 A look at the process list (ps aux | grep 1884) tells me process 1884 is arecord -c 1 -f S16_LE -r 8000 -t raw. No idea what this is or why it's running. When I try and kill arecord (as root), it just respawns and rebinds on the hardware. I'm in a very annoying situation where I don't know what is going on and don't know how to find out. I'm open to all suggestions to get this working again. Fire away. And here's what I get when I stop PA auto-loading, kill it and then start it with -vvvv. oli@bert:~$ pulseaudio -vvvvv I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted D: core-rtclock.c: Timer slack is set to 50 us. D: core-util.c: RealtimeKit worked. I: core-util.c: Successfully gained nice level -11. I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: main.c: Running on host: Linux x86_64 2.6.38-rc3 #1 SMP Tue Feb 1 10:53:04 GMT 2011 D: main.c: Found 8 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimised build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 8310740c4729ef474fe5ecec4bbf5a6b. I: main.c: Session ID is 8310740c4729ef474fe5ecec4bbf5a6b-1297338553.571075-1050119523. I: main.c: Using runtime directory /home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-runtime. I: main.c: Using state directory /home/oli/.pulse. I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Enjoy ol' chap! I: cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes.tdb' I: module-device-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes'. I: module.c: Loaded "module-device-restore" (index: #0; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes.tdb' I: module-stream-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes'. I: module.c: Loaded "module-stream-restore" (index: #1; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database.tdb' I: module-card-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database'. I: module.c: Loaded "module-card-restore" (index: #2; argument: ""). I: module.c: Loaded "module-augment-properties" (index: #3; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards. I: module.c: Loaded "module-udev-detect" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': success D: dbus-util.c: Successfully connected to D-Bus system bus ba7c9a1f90b3d49d930bca2100000015 as :1.62 D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: bluetooth-util.c: Bluetooth daemon is apparently not available. I: module.c: Loaded "module-bluetooth-discover" (index: #5; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-gconf.so': success I: module.c: Loaded "module-gconf" (index: #8; argument: ""). I: module-default-device-restore.c: Saved default sink 'auto_null' not existant, not restoring default sink setting. I: module-default-device-restore.c: Saved default source 'auto_null.monitor' not existant, not restoring default source setting. I: module.c: Loaded "module-default-device-restore" (index: #9; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #10; argument: ""). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. I: sink.c: Created sink 0 "auto_null" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: sink.c: device.description = "Dummy Output" I: sink.c: device.class = "abstract" I: sink.c: device.icon_name = "audio-card" D: core-subscribe.c: Dropped redundant event due to change event. I: source.c: Created source 0 "auto_null.monitor" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: source.c: device.description = "Monitor of Dummy Output" I: source.c: device.class = "monitor" I: source.c: device.icon_name = "audio-input-microphone" D: module-null-sink.c: Thread starting up I: module.c: Loaded "module-null-sink" (index: #11; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: module.c: Loaded "module-intended-roles" (index: #13; argument: ""). D: module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1 I: module.c: Loaded "module-console-kit" (index: #15; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). D: dbus-util.c: Successfully connected to D-Bus session bus efbffc6788fad56cfd64d40c00000018 as :1.182 D: main.c: Got org.pulseaudio.Server! I: main.c: Daemon startup complete. I: client.c: Created 1 "Native client (UNIX socket client)" I: client.c: Created 2 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon D: core-subscribe.c: Dropped redundant event due to change event. I: module-suspend-on-idle.c: Sink auto_null idle for too long, suspending ... D: sink.c: Suspend cause of sink auto_null is 0x0004, suspending Note the one section that seems to find the hardware but says it's busy (no idea if this is relevant). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards.

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  • Where to find xmoov port to C#? (to make Http Pseudo Streaming from c# app)

    - by Ole Jak
    So I found this beautifull script for FLV video format Http Pseudo Streaming but in is in PHP ( found on http://stream.xmoov.com/ ) So does any one know opensource translations or can translate such PHP code into C#? <?php /* xmoov-php 1.0 Development version 0.9.3 beta by: Eric Lorenzo Benjamin jr. webmaster (AT) xmoov (DOT) com originally inspired by Stefan Richter at flashcomguru.com bandwidth limiting by Terry streamingflvcom (AT) dedicatedmanagers (DOT) com This work is licensed under the Creative Commons Attribution-NonCommercial-ShareAlike 3.0 License. For more information, visit http://creativecommons.org/licenses/by-nc-sa/3.0/ For the full license, visit http://creativecommons.org/licenses/by-nc-sa/3.0/legalcode or send a letter to Creative Commons, 543 Howard Street, 5th Floor, San Francisco, California, 94105, USA. */ // SCRIPT CONFIGURATION //------------------------------------------------------------------------------------------ // MEDIA PATH // // you can configure these settings to point to video files outside the public html folder. //------------------------------------------------------------------------------------------ // points to server root define('XMOOV_PATH_ROOT', ''); // points to the folder containing the video files. define('XMOOV_PATH_FILES', 'video/'); //------------------------------------------------------------------------------------------ // SCRIPT BEHAVIOR //------------------------------------------------------------------------------------------ //set to TRUE to use bandwidth limiting. define('XMOOV_CONF_LIMIT_BANDWIDTH', TRUE); //set to FALSE to prohibit caching of video files. define('XMOOV_CONF_ALLOW_FILE_CACHE', FALSE); //------------------------------------------------------------------------------------------ // BANDWIDTH SETTINGS // // these settings are only needed when using bandwidth limiting. // // bandwidth is limited my sending a limited amount of video data(XMOOV_BW_PACKET_SIZE), // in specified time intervals(XMOOV_BW_PACKET_INTERVAL). // avoid time intervals over 1.5 seconds for best results. // // you can also control bandwidth limiting via http command using your video player. // the function getBandwidthLimit($part) holds three preconfigured presets(low, mid, high), // which can be changed to meet your needs //------------------------------------------------------------------------------------------ //set how many kilobytes will be sent per time interval define('XMOOV_BW_PACKET_SIZE', 90); //set the time interval in which data packets will be sent in seconds. define('XMOOV_BW_PACKET_INTERVAL', 0.3); //set to TRUE to control bandwidth externally via http. define('XMOOV_CONF_ALLOW_DYNAMIC_BANDWIDTH', TRUE); //------------------------------------------------------------------------------------------ // DYNAMIC BANDWIDTH CONTROL //------------------------------------------------------------------------------------------ function getBandwidthLimit($part) { switch($part) { case 'interval' : switch($_GET[XMOOV_GET_BANDWIDTH]) { case 'low' : return 1; break; case 'mid' : return 0.5; break; case 'high' : return 0.3; break; default : return XMOOV_BW_PACKET_INTERVAL; break; } break; case 'size' : switch($_GET[XMOOV_GET_BANDWIDTH]) { case 'low' : return 10; break; case 'mid' : return 40; break; case 'high' : return 90; break; default : return XMOOV_BW_PACKET_SIZE; break; } break; } } //------------------------------------------------------------------------------------------ // INCOMING GET VARIABLES CONFIGURATION // // use these settings to configure how video files, seek position and bandwidth settings are accessed by your player //------------------------------------------------------------------------------------------ define('XMOOV_GET_FILE', 'file'); define('XMOOV_GET_POSITION', 'position'); define('XMOOV_GET_AUTHENTICATION', 'key'); define('XMOOV_GET_BANDWIDTH', 'bw'); // END SCRIPT CONFIGURATION - do not change anything beyond this point if you do not know what you are doing //------------------------------------------------------------------------------------------ // PROCESS FILE REQUEST //------------------------------------------------------------------------------------------ if(isset($_GET[XMOOV_GET_FILE]) && isset($_GET[XMOOV_GET_POSITION])) { // PROCESS VARIABLES # get seek position $seekPos = intval($_GET[XMOOV_GET_POSITION]); # get file name $fileName = htmlspecialchars($_GET[XMOOV_GET_FILE]); # assemble file path $file = XMOOV_PATH_ROOT . XMOOV_PATH_FILES . $fileName; # assemble packet interval $packet_interval = (XMOOV_CONF_ALLOW_DYNAMIC_BANDWIDTH && isset($_GET[XMOOV_GET_BANDWIDTH])) ? getBandwidthLimit('interval') : XMOOV_BW_PACKET_INTERVAL; # assemble packet size $packet_size = ((XMOOV_CONF_ALLOW_DYNAMIC_BANDWIDTH && isset($_GET[XMOOV_GET_BANDWIDTH])) ? getBandwidthLimit('size') : XMOOV_BW_PACKET_SIZE) * 1042; # security improved by by TRUI www.trui.net if (!file_exists($file)) { print('<b>ERROR:</b> xmoov-php could not find (' . $fileName . ') please check your settings.'); exit(); } if(file_exists($file) && strrchr($fileName, '.') == '.flv' && strlen($fileName) > 2 && !eregi(basename($_SERVER['PHP_SELF']), $fileName) && ereg('^[^./][^/]*$', $fileName)) { # stay clean @ob_end_clean(); @set_time_limit(0); # keep binary data safe set_magic_quotes_runtime(0); $fh = fopen($file, 'rb') or die ('<b>ERROR:</b> xmoov-php could not open (' . $fileName . ')'); $fileSize = filesize($file) - (($seekPos > 0) ? $seekPos + 1 : 0); // SEND HEADERS if(!XMOOV_CONF_ALLOW_FILE_CACHE) { # prohibit caching (different methods for different clients) session_cache_limiter("nocache"); header("Expires: Thu, 19 Nov 1981 08:52:00 GMT"); header("Last-Modified: " . gmdate("D, d M Y H:i:s") . " GMT"); header("Cache-Control: no-store, no-cache, must-revalidate, post-check=0, pre-check=0"); header("Pragma: no-cache"); } # content headers header("Content-Type: video/x-flv"); header("Content-Disposition: attachment; filename=\"" . $fileName . "\""); header("Content-Length: " . $fileSize); # FLV file format header if($seekPos != 0) { print('FLV'); print(pack('C', 1)); print(pack('C', 1)); print(pack('N', 9)); print(pack('N', 9)); } # seek to requested file position fseek($fh, $seekPos); # output file while(!feof($fh)) { # use bandwidth limiting - by Terry if(XMOOV_CONF_LIMIT_BANDWIDTH) { # get start time list($usec, $sec) = explode(' ', microtime()); $time_start = ((float)$usec + (float)$sec); # output packet print(fread($fh, $packet_size)); # get end time list($usec, $sec) = explode(' ', microtime()); $time_stop = ((float)$usec + (float)$sec); # wait if output is slower than $packet_interval $time_difference = $time_stop - $time_start; # clean up @flush(); @ob_flush(); if($time_difference < (float)$packet_interval) { usleep((float)$packet_interval * 1000000 - (float)$time_difference * 1000000); } } else { # output file without bandwidth limiting print(fread($fh, filesize($file))); } } } } ?>

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  • Dealing with numerous, simultaneous sounds in unity

    - by luxchar
    I've written a custom class that creates a fixed number of audio sources. When a new sound is played, it goes through the class, which creates a queue of sounds that will be played during that frame. The sounds that are closer to the camera are given preference. If new sounds arrive in the next frame, I have a complex set of rules that determines how to replace the old ones. Ideally, "big" or "important" sounds should not be replaced by small ones. Sound replacement is necessary since the game can be fast-paced at times, and should try to play new sounds by replacing old ones. Otherwise, there can be "silent" moments when an old sound is about to stop playing and isn't replaced right away by a new sound. The drawback of replacing old sounds right away is that there is a harsh transition from the old sound clip to the new one. But I wonder if I could just remove that management logic altogether, and create audio sources on the fly for new sounds. I could give "important" sounds more priority (closer to 0 in the corresponding property) as opposed to less important ones, and let Unity take care of culling out sound effects that exceed the channel limit. The only drawback is that it requires many heap allocations. I wonder what strategy people use here?

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