Search Results

Search found 5274 results on 211 pages for 'stream operators'.

Page 54/211 | < Previous Page | 50 51 52 53 54 55 56 57 58 59 60 61  | Next Page >

  • counting unique values based on multiple columns

    - by gooogalizer
    I am working in google spreadsheets and I am trying to do some counting that takes into consideration cell values across multiple cells in each row. Here's my table: |AUTHOR| |ARTICLE| |VERSION| |PRE-SELECTED| ANDREW GOLF STREAM 1 X ANDREW GOLF STREAM 2 X ANDREW HURRICANES 1 JOHN CAPE COD 1 X JOHN GOLF STREAM 1 (Google doc here) Each person can submit multiple articles as well as multiple versions of the same article. Sometimes different people submit different articles that happen to be identically named (Andrew and John both submitted different articles called "Golf Stream"). Multiple versions written by the same person do not count as unique, but articles with the same title written by different people do count as unique. So, I am looking to find a formula that Counts the number of unique articles that have been submitted [4] (without having to manually create extra columns for doing CONCATS, if possible) It would also be great to find formulas that: Count the number of unique articles that have been pre-selected (marked "X" in "PRE-SELECTED" column) [2] Count the number of unique articles that have only 1 version [4] Count the number of unique articles that have more than 1 of their versions pre-selected 1 Thank you so much! Nikita

    Read the article

  • How can I close a port that appears to be orphaned by Xvfb?

    - by Jim Fiorato
    I'm running Xvfb on a FC8 Amazon EC2 image. On occasion Xvfb will crash (unable at the moment to find out the reason for the crash), and after crashing the TCP port will appear to be orphaned. I'm unable to get a PID to kill any process that may be using it. I'm starting Xvfb with: Xvfb :7 -screen 0 1024x768x24 & Examples of what I'm working with are below, the Xvfb port is (was) 6007: # netstat -ap Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 0 0 *:ssh *:* LISTEN 1894/sshd tcp 0 0 *:6007 *:* LISTEN - tcp 0 352 ip-10-84-69-165.ec2.int:ssh c-71-194-253-238.hsd1:51689 ESTABLISHED 2981/0 udp 0 0 *:bootpc *:* 1817/dhclient udp 0 0 *:bootpc *:* 1463/dhclient Active UNIX domain sockets (servers and established) Proto RefCnt Flags Type State I-Node PID/Program name Path unix 2 [ ] DGRAM 871 668/udevd @/org/kernel/udev/udevd unix 2 [ ACC ] STREAM LISTENING 5385 1880/dbus-daemon /var/run/dbus/system_bus_socket unix 6 [ ] DGRAM 5353 1867/rsyslogd /dev/log unix 2 [ ] DGRAM 11861 2981/0 unix 2 [ ] DGRAM 5461 1974/crond unix 2 [ ] DGRAM 5451 1904/console-kit-da unix 3 [ ] STREAM CONNECTED 5438 1880/dbus-daemon /var/run/dbus/system_bus_socket unix 3 [ ] STREAM CONNECTED 5437 1904/console-kit-da unix 3 [ ] STREAM CONNECTED 5396 1880/dbus-daemon unix 3 [ ] STREAM CONNECTED 5395 1880/dbus-daemon unix 2 [ ] DGRAM 5361 1871/rklogd # lsof -i COMMAND PID USER FD TYPE DEVICE SIZE NODE NAME dhclient 1463 root 3u IPv4 4704 UDP *:bootpc dhclient 1817 root 4u IPv4 5173 UDP *:bootpc sshd 1894 root 3u IPv4 5414 TCP *:ssh (LISTEN) sshd 2981 root 3u IPv4 11825 TCP ip-10-84-69-165.ec2.internal:ssh->c-71-194-253-238.hsd1.il.comcast.net:51689 (ESTABLISHED) Attempting to force the port closed with iptables doesn't seem to work either. iptables -A INPUT -p tcp --dport 6007 -j DROP I'm at a loss as to how to reclaim/free the port. From what I can tell, this port will remain in this state until the EC2 instance is shut down. So, how can I close this port so I can restart Xvfb?

    Read the article

  • mplayer audio desync

    - by geek
    I have and avi file and an ac3 file that contains an alternate audio stream. I run mplayer like: mplayer -audiofile foo.ac3 bar.avi mplayer takes the audio stream from the ac3 file as expected, but when I try to scroll the video using arrows or pgup/pgdown keys, the audio gets desynced: mplayer just starts playing the audio stream from the beginning. Do I have to pass any additional command line arguments in order to make it scroll properly without desyncing audio?

    Read the article

  • How to open an iPhone compatible M3U file on Windows?

    - by user1158667
    This is how the M3U file looks like: #EXTM3U #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1400000 http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=900000 http://maskedip/http_livestr.str?r=true&id=test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=450000 http://maskedip/http_livestr.str?r=true&id=mobile-test&k=testkey #EXT-X-STREAM-INF:PROGRAM-ID=1,CODECS="mp4a.40.2",BANDWIDTH=64000 http://maskedup/http_livestr.str?r=true&id=test-audio&k=testkey Clicking on http://maskedip/http_livestr.str?r=true&id=mbit-test&k=testkey then returns another M3U file in this format: #EXTM3U #EXT-X-TARGETDURATION:10 #EXT-X-MEDIA-SEQUENCE:1361 #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1361.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1362.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1363.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1364.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1365.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1366.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1367.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1368.ts #EXTINF:10, http://maskedip/http_ls/testkey/mbit-test1/1369.ts Anyways, VLC won't recognize it. How can I play this on the PC?

    Read the article

  • Having trouble Getting "RTSP over HTTP"

    - by Muhammad Adeel Zahid
    There is an axis camera that is connected to our site (camba.tv) through axis one click connection component (which acts as proxy). We can communicate with this camera only through http by setting the proxy to our OCCC server's address. If we want to get RTSP streams (h.264) we are only left with "RTSP over HTTP" option. For this I have followed axis VAPIX 3 documentation section 3.3. I issue requests through fiddler but don't get any response. But when i put the URL (axrtsphttp://1.00408CBEA38B/axis-media/media.amp) in windows media player (with proxy set to OCCC server 212.78.237.156:3128) the player is able to get RTSP stream over HTTP after logging in. I have created a request trace of communication between camera and windows media player through wireshark and the request that brings the stream looks like http://1.00408cbea38b/axis-media/media.amp HTTP/1.1 x-sessioncookie: 619 User-Agent: Axis AMC Host: 1.00408CBEA38B Proxy-Connection: Keep-Alive Pragma: no-cache Authorization: Digest username="root",realm="AXIS_00408CBEA38B",nonce="000a8b40Y0100409c13ac7e6cceb069289041d8feb1691",uri="/axis-media/media.amp",cnonce="9946e2582bd590418c9b70e1b17956c7",nc=00000001,response="f3cab86fc84bfe33719675848e7fdc0a",qop="auth" HTTP/1.0 200 OK Content-Type: application/x-rtsp-tunnelled Date: Tue, 02 Nov 2010 11:45:23 GMT RTSP/1.0 200 OK CSeq: 1 Content-Type: application/sdp Content-Base: rtsp://1.00408CBEA38B/axis-media/media.amp/ Date: Tue, 02 Nov 2010 11:45:23 GMT Content-Length: 410 v=0 o=- 1288698323798001 1288698323798001 IN IP4 1.00408CBEA38B s=Media Presentation e=NONE c=IN IP4 0.0.0.0 b=AS:50000 t=0 0 a=control:* a=range:npt=0.000000- m=video 0 RTP/AVP 96 b=AS:50000 a=framerate:30.0 a=transform:1,0,0;0,1,0;0,0,1 a=control:trackID=1 a=rtpmap:96 H264/90000 a=fmtp:96 packetization-mode=1; profile-level-id=420029; sprop-parameter-sets=Z0IAKeNQFAe2AtwEBAaQeJEV,aM48gA== RTSP/1.0 200 OK CSeq: 2 Session: 3F4763D8; timeout=60 Transport: RTP/AVP/TCP;unicast;interleaved=0-1;ssrc=060922C6;mode="PLAY" Date: Tue, 02 Nov 2010 11:45:24 GMT RTSP/1.0 200 OK CSeq: 3 Session: 3F4763D8 Range: npt=0- RTP-Info: url=rtsp://1.00408CBEA38B/axis-media/media.amp/trackID=1;seq=7392;rtptime=4190934902 Date: Tue, 02 Nov 2010 11:45:24 GMT [Binary Stream Content] But when i copy this request to fiddler, I only get 200 status code with content-type set to application/x-rtsp-tunneled and there is no stream data. The only thing i do different with stream is to use Basic in authorization header instead of Digest and I do not get 401 (Un authorized) status code. Can anyone explain what's happening here? How can I write request sequences to get stream in fiddler? If it is needed, I can upload the wireshark request dump somewhere.

    Read the article

  • set proxy in apache for XMPP chat

    - by Hunt
    I want to setup a proxy settings in Apache to use Facebook XMPP Chat So far I have setup ejabber server and I am able to access xmpp service using http://mydomain.com:5280/xmpp-http-bind I am able to create Jabber Account too. Now as I want to integrate Facebook XMPP chat , I want my server to sit in between client and chat.facebook.com because I want to implement Facebook chat and custom chat too. So I have read this article and come to know that I need to serve BOSH Service as a proxy in apache to access Facebook Chat service. So I don't know how to set up a proxy in a apache httpd.conf as I have tried following <Proxy *> Order deny,allow Allow from all </Proxy> ProxyPass /xmpp-httpbind http://www.mydomain.com:5280/xmpp-http-bind ProxyPassReverse /xmpp-httpbind http://www.mydomain.com:5280/xmpp-http-bind But whenever I request http://www.mydomain.com:5280/xmpp-http-bind from strophe.js I am getting following response from server <body type='terminate' condition='internal-server-error' xmlns='http://jabber.org/protocol/httpbind'> BOSH module not started </body> and server log says following E(<0.567.0:ejabberd_http_bind:1239) : You are trying to use BOSH (HTTP Bind) in host "chat.facebook.com", but the module mod_http_bind is not started in that host. Configure your BOSH client to connect to the correct host, or add your desired host to the configuration, or check your 'modules' section in your ejabberd configuration file. here is my existing settings of ejabberd.cfg , but still no luck {5280, ejabberd_http, [ {access,all}, {request_handlers, [ {["pub", "archive"], mod_http_fileserver}, {["xmpp-http-bind"], mod_http_bind} ]}, captcha, http_bind, http_poll, register, web_admin ]} ]}. in a module section {mod_http_bind, [{max_inactivity, 120}]}, and whenever i fire http://www.mydomain.com:5280/xmpp-http-bind url independently am getting following message ejabberd mod_http_bind An implementation of XMPP over BOSH (XEP-0206) This web page is only informative. To use HTTP-Bind you need a Jabber/XMPP client that supports it. I have added chat.facebook.com in a list of host in ejabber.cfg as follows {hosts, ["localhost","mydomain.com","chat.facebook.com"]} and now i am getting following response <body xmlns='http://jabber.org/protocol/httpbind' sid='710da2568460512eeb546545a65980c2704d9a27' wait='300' requests='2' inactivity='120' maxpause='120' polling='2' ver='1.8' from='chat.facebook.com' secure='true' authid='1917430584' xmlns:xmpp='urn:xmpp:xbosh' xmlns:stream='http://etherx.jabber.org/streams' xmpp:version='1.0'> <stream:features xmlns:stream='http://etherx.jabber.org/streams'> <mechanisms xmlns='urn:ietf:params:xml:ns:xmpp-sasl'> <mechanism>DIGEST-MD5</mechanism> <mechanism>PLAIN</mechanism> </mechanisms> <c xmlns='http://jabber.org/protocol/caps' hash='sha-1' node='http://www.process-one.net/en/ejabberd/' ver='yy7di5kE0syuCXOQTXNBTclpNTo='/> <register xmlns='http://jabber.org/features/iq-register'/> </stream:features> </body> if i use valid BOSH service created my jack moffit http://bosh.metajack.im:5280/xmpp-httpbind then i am getting following valid XML from facebook , but from my server i am not getting this <body xmlns='http://jabber.org/protocol/httpbind' inactivity='60' secure='true' authid='B8732AA1' content='text/xml; charset=utf-8' window='3' polling='15' sid='928073b02da55d34eb3c3464b4a40a37' requests='2' wait='300'> <stream:features xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:client'> <mechanisms xmlns='urn:ietf:params:xml:ns:xmpp-sasl'> <mechanism>X-FACEBOOK-PLATFORM</mechanism> <mechanism>DIGEST-MD5</mechanism> </mechanisms> </stream:features> </body> Can anyone please help me to resolve the issue

    Read the article

  • Convert mp4 video to a format xbox 360 can play

    - by Björn Lindqvist
    Here is a video file my Xbox 360 refuses to play: $ MP4Box -info video.mp4 * Movie Info * Timescale 90000 - Duration 02:18:33.365 Fragmented File no - 2 track(s) File Brand mp42 - version 0 Created: GMT Sat Jul 21 07:08:55 2012 File has root IOD (9 bytes) Scene PL 0xff - Graphics PL 0xff - OD PL 0xff Visual PL: ISO Reserved Profile (0x7f) Audio PL: High Quality Audio Profile @ Level 2 (0x0f) No streams included in root OD iTunes Info: Encoder Software: HandBrake 0.9.6 2012022800 Track # 1 Info - TrackID 1 - TimeScale 90000 - Duration 02:18:33.235 Media Info: Language "Undetermined" - Type "vide:avc1" - 199318 samples Visual Track layout: x=0 y=0 width=1280 height=688 MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 1280 x 688 AVC Info: 1 SPS - 1 PPS - Profile High @ Level 4.1 NAL Unit length bits: 32 Self-synchronized Track # 2 Info - TrackID 2 - TimeScale 48000 - Duration 02:18:33.365 Media Info: Language "English" - Type "soun:mp4a" - 389689 samples MPEG-4 Config: Audio Stream - ObjectTypeIndication 0x40 MPEG-4 Audio MPEG-4 Audio AAC LC - 6 Channel(s) - SampleRate 48000 Synchronized on stream 1 $ avconv -i video.mp4 avconv version 0.8.4-4:0.8.4-0ubuntu0.12.04.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:33 with gcc 4.6.3 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: mp42isomavc1 creation_time : 2012-07-21 07:08:55 encoder : HandBrake 0.9.6 2012022800 Duration: 02:18:33.36, start: 0.000000, bitrate: 2299 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x688, 1973 kb/s, 23.98 fps, 90k tbr, 90k tbn, 180k tbc Metadata: creation_time : 2012-07-21 07:08:55 Stream #0.1(eng): Audio: aac, 48000 Hz, 5.1, s16, 319 kb/s Metadata: creation_time : 2012-07-21 07:08:55 At least one output file must be specified What tool, such as ffmpeg or mencoder, and what magic command line incantation should I use to transcode this file into a format Xbox 360 can play? I want the transcode process to retain as good video quality as possible.

    Read the article

  • Convert swf file to mp4 file using FFMPEG

    - by user1624004
    I now want to show an html5 video on a html page. Now I have an sample.swf file, I want to convert it to .mp4 or .ogg or .webm file. I have tried: ffmpeg -i sample.swf sample.mp4 But I got this error: [swf @ 0000000001feef40] Could not find codec parameters for stream 0 (Audio: pcm_s16le, 5512 Hz, 1 channels, 88 kb/s): unspecified sample format Consider increasing the value for the 'analyzeduration' and 'probesize' options [swf @ 0000000001feef40] Estimating duration from bitrate, this may be inaccurate Guessed Channel Layout for Input Stream #0.0 : mono Input #0, swf, from 'sample.swf': Duration: N/A, bitrate: N/A Stream #0:0: Audio: pcm_s16le, 5512 Hz, mono, 88 kb/s Stream #0:1: Video: mjpeg, yuvj444p, 1024x768 [SAR 100:100 DAR 4:3], 16 fps, 16 tbr, 16 tbn File 'sample.mp4' already exists. Overwrite ? [y/N] y Invalid sample format '(null)' Error opening filters!

    Read the article

  • Batch Convert .mkv to .mp4

    - by IamHere
    I want to batch convert all .mkv files in a folder into .mp4 with VLC. It should use the original video-/audio stream and if possible the .ass subtitle of the .mkv. It's not really a conversion, it's more like changing the container – my player can't read the MKV videos. If I do this conversion by hand (manually) it works, but I have a lot of MKV files to convert, so it would take a lot of time. I have searched the internet for a batch file to do this and I found a few. I tried to modify them to my wish, but all attempts I tried just created a .mp4 file that doesn't contain the audio stream and the video stream also cannot be rendered by all my media players on the PC. So could someone tell me how the batch has to look like, so it works with the original video and audio stream (and maybe .ass subtitles)?

    Read the article

  • How do I convert a video to GIF using ffmpeg, with reasonable quality?

    - by Kamil Hismatullin
    I'm converting .flv movie to .gif file with ffmpeg. ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif It works great, but output gif file has a very law quality. Any ideas how can I improve quality of converted gif? Output of command: $ ffmpeg -i input.flv -ss 00:00:00.000 -pix_fmt rgb24 -r 10 -s 320x240 -t 00:00:10.000 output.gif ffmpeg version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:52:53 with gcc 4.7.2 *** THIS PROGRAM IS DEPRECATED *** This program is only provided for compatibility and will be removed in a future release. Please use avconv instead. Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'input.flv': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 Duration: 00:00:18.85, start: 0.000000, bitrate: 3098 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 1280x720, 2905 kb/s, 25 fps, 25 tbr, 50 tbn, 50 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(und): Audio: aac, 44100 Hz, stereo, s16, 192 kb/s Metadata: creation_time : 2013-02-14 04:00:07 [buffer @ 0x92a8ea0] w:1280 h:720 pixfmt:yuv420p [scale @ 0x9215100] w:1280 h:720 fmt:yuv420p -> w:320 h:240 fmt:rgb24 flags:0x4 Output #0, gif, to 'output.gif': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2013-02-14 04:00:07 encoder : Lavf53.21.1 Stream #0.0(und): Video: rawvideo, rgb24, 320x240, q=2-31, 200 kb/s, 90k tbn, 10 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream mapping: Stream #0.0 -> #0.0 Press ctrl-c to stop encoding frame= 101 fps= 32 q=0.0 Lsize= 8686kB time=10.10 bitrate=7045.0kbits/s dup=0 drop=149 video:22725kB audio:0kB global headers:0kB muxing overhead -61.778676% Thanks.

    Read the article

  • Possible to capture and re-broadcast as RTMP?

    - by Jeremy White
    I may have a need coming up soon to capture a live video broadcast stream and re-broadcast it as an RTMP stream for playback in Flash Player. Is this possible? I'm seeing posts online from 2005 to 2009 claiming that RTMP either isn't or is poorly supported in VLC. I do not currently know what format the incoming video stream will be -- will update when I get that information.

    Read the article

  • how to use time out in mplayer?

    - by manoj
    I am trying to save audio using mplayer from a live http stream. saving audio is successful. If there is no live stream playing it does not exit automatically. Is there any way to set timeout if there is no live stream? code : mplayer -i -t 00:00:10 -acodec libmp3lame -ab 24 -ar 8000 audio.mp3 Thanks in advance.

    Read the article

  • converting to MXF using ffmpeg

    - by Prakash
    I have been trying to use FFmpeg utility to convert a avi file using DNxHD to mxf format. I am using "FFmpeg" with params as following: ffmpeg -i ccvt_box.avi -vcodec dnxhd -video_size 1920x1080 -r 24 -b:v 115m ex.mxf The error it is giving : ffmpeg version N-43737-g76c3fff Copyright (c) 2000-2012 the FFmpeg developers built on Aug 20 2012 18:50:42 with llvm-gcc 4.2.1 (LLVM build 2336.11.00) configuration: libavutil 51. 70.100 / 51. 70.100 libavcodec 54. 53.100 / 54. 53.100 libavformat 54. 25.104 / 54. 25.104 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 11.101 / 3. 11.101 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 Input #0, avi, from 'ccvt_box.avi': Duration: 00:00:10.00, start: 0.000000, bitrate: 691 kb/s Stream #0:0: Video: indeo5 (IV50 / 0x30355649), yuv410p, 340x344, 10 tbr, 10 tbn, 10 tbc Metadata: title : bob.avi [dnxhd @ 0x7fcd60818e00] video parameters incompatible with DNxHD Output #0, mxf, to 'ex.mxf': Stream #0:0: Video: dnxhd, yuv422p, 340x344, q=2-1024, 90k tbn, 24 tbc Metadata: title : bob.avi Stream mapping: Stream #0:0 -> #0:0 (indeo5 -> dnxhd) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height

    Read the article

  • Debugging an IP Camera

    - by Kevin Boyd
    Further to my previous question on ServerFault here, I finally can view the stream on RTSP however I still cannot view the camera stream in a web browser. The IP camera uses an activeX control in Internet Explorer. And although I can configure the camera settings from IE, I cannot view the stream it shows connecting for a few sec and shows disconnecting. I have forwarded the HTTP, RTSP and Stream ports of the IP camera. the public port is 7071 and private port is 7070. When I try to see the connections in TCPView it shows that the ActiveX control in IE is trying to connect to port 7070 which is quite unusual since it should connect to 7071 Also the state shows SYN_SENT for sometime and then disconnects. I have really no clue what's going on and why?

    Read the article

  • Need advice on how to set up live video streaming to web/mobile devices

    - by jasondewitt
    I have a bunch of live udp video streams that currently are viewed by set top boxes in my network. I would like to pick this video up (I can do this with vlc now) and stream it out to other non-STB endpoints (webpage or a phone/tablet of some sort). Right now I am able to pick up the udp stream with vlc and convert it to an http stream on port 8080 of my vlc box. Then I can use the vlc client to pick up and watch that video stream. This is where I'm not sure where to go with it. I really doubt I would want everyone who is watching the video to make a connection back to my vlc server that is doing the encoding, so how do I distribute this live video to the people who want to see it?

    Read the article

  • Batch Convert .mkv to .mp4 with VLC

    - by IamHere
    i want to batch convert all .mkv file in a folder into .mp4 with vlc. it should use the original video-/audio-stream and if possible the .ass subtitle of the .mkv. so its not really a convert, its more changing the container. (my player cant read the mkv) if i do this conversion by hand (manually) it works, but i have a lot of mkv to convert, so it would take a lot of time. i have searched the internet for a batch file to do this and i found a few and tried to modify them to my wish, but all attempts i tried just created a .mp4 file that doesn't contain the audio-stream and the video-stream also cannot be rendered by all my mediaplayers on the pc. so i would be very thankful, if you could tell me how the batch have to look like, so it works with original video-/audio-stream (maybe .ass subtitle)

    Read the article

  • Does Apache 2.2 (windows) have any default bandwidth limit?

    - by igino manfre'
    I'm running Apache on a server in cloud (Windows server 2008 R2 on VMware, 1 Gbps of BW, http://95.110.164.61 ). I'm streaming many live DVB MPEG Transport Stream, precompressed in loop, (not flash) generated by VLC on port 640xx and then reverse proxied by Apache on port 80. The server's firewall is open for VLC and Apache on all ports. Above 1.5 Mbps the reproduction is affected by continous stop & go. Please note that if you request a stream generated by VLC directly at http://95.110.164.61:64087/mpg2_6.4 you see a correct stream, while if you request http://95.110.164.61/mpg2_6.4 you do not. I know that Flash streaming Server uses Apache to stream on port 80 (and it works). I'm not an expert with Apache, can anyone tell me if any "special" module is required to increase the bandwidth?

    Read the article

  • Lots of artifacts while streaming HD content with VLC 0.9.9 on CentOS

    - by Zsub
    I'm trying to stream (multicast) a x264 encoded file using VLC. This in itself succeeds, but the stream has a huge lot of artifacts. This seems to suggest that the data cannot be transported fast enough. If I check network usage, though, it's only using about 15 mbit. I have a similar SD stream which functions perfectly. I think I could improve stream performance by not streaming the raw data, but I cannot seem to get this working. It seems that on keyframes all artifacts are removed for a short while (less than a second). This is the command I use: vlc -vv hdtest.mkv --sout '#duplicate{dst=rtp{dst=ff02::1%eth1,mux=ts,port=5678,sap,group="Testgroup",name="TeststreamHD"}}' --loop Which is all one long line.

    Read the article

  • perl - universal operator overload

    - by Todd Freed
    I have an idea for perl, and I'm trying to figure out the best way to implement it. The idea is to have new versions of every operator which consider the undefined value as the identity of that operation. For example: $a = undef + 5; # undef treated as 0, so $a = 5 $a = undef . "foo"; # undef treated as '', so $a = foo $a = undef && 1; # undef treated as false, $a = true and so forth. ideally, this would be in the language as a pragma, or something. use operators::awesome; However, I would be satisfied if I could implement this special logic myself, and then invoke it where needed: use My::Operators; The problem is that if I say "use overload" inside My::Operators only affects objects blessed into My::Operators. So the question is: is there a way (with "use overoad" or otherwise) to do a "universal operator overload" - which would be called for all operations, not just operations on blessed scalars. If not - who thinks this would be a great idea !? It would save me a TON of this kind of code if($object && $object{value} && $object{value} == 15) replace with if($object{value} == 15) ## the special "is-equal-to" operator

    Read the article

  • Make conversion to a native type explicit in C++

    - by Tal Pressman
    I'm trying to write a class that implements 64-bit ints for a compiler that doesn't support long long, to be used in existing code. Basically, I should be able to have a typedef somewhere that selects whether I want to use long long or my class, and everything else should compile and work. So, I obviously need conversion constructors from int, long, etc., and the respective conversion operators (casts) to those types. This seems to cause errors with arithmetic operators. With native types, the compiler "knows" that when operator*(int, char) is called, it should promote the char to int and call operator*(int, int) (rather than casting the int to char, for example). In my case it gets confused between the various built-in operators and the ones I created. It seems to me like if I could flag the conversion operators as explicit somehow, that it would solve the issue, but as far as I can tell the explicit keyword is only for constructors (and I can't make constructors for built-in types). So is there any way of marking the casts as explicit? Or am I barking up the wrong tree here and there's another way of solving this? Or maybe I'm just doing something else wrong...

    Read the article

  • C++0x Smart Pointer Comparisons: Inconsistent, what's the rationale?

    - by GManNickG
    In C++0x (n3126), smart pointers can be compared, both relationally and for equality. However, the way this is done seems inconsistent to me. For example, shared_ptr defines operator< be equivalent to: template <typename T, typename U> bool operator<(const shared_ptr<T>& a, const shared_ptr<T>& b) { return std::less<void*>()(a.get(), b.get()); } Using std::less provides total ordering with respect to pointer values, unlike a vanilla relational pointer comparison, which is unspecified. However, unique_ptr defines the same operator as: template <typename T1, typename D1, typename T2, typename D2> bool operator<(const unique_ptr<T1, D1>& a, const unique_ptr<T2, D2>& b) { return a.get() < b.get(); } It also defined the other relational operators in similar fashion. Why the change in method and "completeness"? That is, why does shared_ptr use std::less while unique_ptr uses the built-in operator<? And why doesn't shared_ptr also provide the other relational operators, like unique_ptr? I can understand the rationale behind either choice: with respect to method: it represents a pointer so just use the built-in pointer operators, versus it needs to be usable within an associative container so provide total ordering (like a vanilla pointer would get with the default std::less predicate template argument) with respect to completeness: it represents a pointer so provide all the same comparisons as a pointer, versus it is a class type and only needs to be less-than comparable to be used in an associative container, so only provide that requirement But I don't see why the choice changes depending on the smart pointer type. What am I missing? Bonus/related: std::shared_ptr seems to have followed from boost::shared_ptr, and the latter omits the other relational operators "by design" (and so std::shared_ptr does too). Why is this?

    Read the article

  • The fastest way to resize images from ASP.NET. And it’s (more) supported-ish.

    - by Bertrand Le Roy
    I’ve shown before how to resize images using GDI, which is fairly common but is explicitly unsupported because we know of very real problems that this can cause. Still, many sites still use that method because those problems are fairly rare, and because most people assume it’s the only way to get the job done. Plus, it works in medium trust. More recently, I’ve shown how you can use WPF APIs to do the same thing and get JPEG thumbnails, only 2.5 times faster than GDI (even now that GDI really ultimately uses WIC to read and write images). The boost in performance is great, but it comes at a cost, that you may or may not care about: it won’t work in medium trust. It’s also just as unsupported as the GDI option. What I want to show today is how to use the Windows Imaging Components from ASP.NET APIs directly, without going through WPF. The approach has the great advantage that it’s been tested and proven to scale very well. The WIC team tells me you should be able to call support and get answers if you hit problems. Caveats exist though. First, this is using interop, so until a signed wrapper sits in the GAC, it will require full trust. Second, the APIs have a very strong smell of native code and are definitely not .NET-friendly. And finally, the most serious problem is that older versions of Windows don’t offer MTA support for image decoding. MTA support is only available on Windows 7, Vista and Windows Server 2008. But on 2003 and XP, you’ll only get STA support. that means that the thread safety that we so badly need for server applications is not guaranteed on those operating systems. To make it work, you’d have to spin specialized threads yourself and manage the lifetime of your objects, which is outside the scope of this article. We’ll assume that we’re fine with al this and that we’re running on 7 or 2008 under full trust. Be warned that the code that follows is not simple or very readable. This is definitely not the easiest way to resize an image in .NET. Wrapping native APIs such as WIC in a managed wrapper is never easy, but fortunately we won’t have to: the WIC team already did it for us and released the results under MS-PL. The InteropServices folder, which contains the wrappers we need, is in the WicCop project but I’ve also included it in the sample that you can download from the link at the end of the article. In order to produce a thumbnail, we first have to obtain a decoding frame object that WIC can use. Like with WPF, that object will contain the command to decode a frame from the source image but won’t do the actual decoding until necessary. Getting the frame is done by reading the image bytes through a special WIC stream that you can obtain from a factory object that we’re going to reuse for lots of other tasks: var photo = File.ReadAllBytes(photoPath); var factory = (IWICComponentFactory)new WICImagingFactory(); var inputStream = factory.CreateStream(); inputStream.InitializeFromMemory(photo, (uint)photo.Length); var decoder = factory.CreateDecoderFromStream( inputStream, null, WICDecodeOptions.WICDecodeMetadataCacheOnLoad); var frame = decoder.GetFrame(0); We can read the dimensions of the frame using the following (somewhat ugly) code: uint width, height; frame.GetSize(out width, out height); This enables us to compute the dimensions of the thumbnail, as I’ve shown in previous articles. We now need to prepare the output stream for the thumbnail. WIC requires a special kind of stream, IStream (not implemented by System.IO.Stream) and doesn’t directlyunderstand .NET streams. It does provide a number of implementations but not exactly what we need here. We need to output to memory because we’ll want to persist the same bytes to the response stream and to a local file for caching. The memory-bound version of IStream requires a fixed-length buffer but we won’t know the length of the buffer before we resize. To solve that problem, I’ve built a derived class from MemoryStream that also implements IStream. The implementation is not very complicated, it just delegates the IStream methods to the base class, but it involves some native pointer manipulation. Once we have a stream, we need to build the encoder for the output format, which could be anything that WIC supports. For web thumbnails, our only reasonable options are PNG and JPEG. I explored PNG because it’s a lossless format, and because WIC does support PNG compression. That compression is not very efficient though and JPEG offers good quality with much smaller file sizes. On the web, it matters. I found the best PNG compression option (adaptive) to give files that are about twice as big as 100%-quality JPEG (an absurd setting), 4.5 times bigger than 95%-quality JPEG and 7 times larger than 85%-quality JPEG, which is more than acceptable quality. As a consequence, we’ll use JPEG. The JPEG encoder can be prepared as follows: var encoder = factory.CreateEncoder( Consts.GUID_ContainerFormatJpeg, null); encoder.Initialize(outputStream, WICBitmapEncoderCacheOption.WICBitmapEncoderNoCache); The next operation is to create the output frame: IWICBitmapFrameEncode outputFrame; var arg = new IPropertyBag2[1]; encoder.CreateNewFrame(out outputFrame, arg); Notice that we are passing in a property bag. This is where we’re going to specify our only parameter for encoding, the JPEG quality setting: var propBag = arg[0]; var propertyBagOption = new PROPBAG2[1]; propertyBagOption[0].pstrName = "ImageQuality"; propBag.Write(1, propertyBagOption, new object[] { 0.85F }); outputFrame.Initialize(propBag); We can then set the resolution for the thumbnail to be 96, something we weren’t able to do with WPF and had to hack around: outputFrame.SetResolution(96, 96); Next, we set the size of the output frame and create a scaler from the input frame and the computed dimensions of the target thumbnail: outputFrame.SetSize(thumbWidth, thumbHeight); var scaler = factory.CreateBitmapScaler(); scaler.Initialize(frame, thumbWidth, thumbHeight, WICBitmapInterpolationMode.WICBitmapInterpolationModeFant); The scaler is using the Fant method, which I think is the best looking one even if it seems a little softer than cubic (zoomed here to better show the defects): Cubic Fant Linear Nearest neighbor We can write the source image to the output frame through the scaler: outputFrame.WriteSource(scaler, new WICRect { X = 0, Y = 0, Width = (int)thumbWidth, Height = (int)thumbHeight }); And finally we commit the pipeline that we built and get the byte array for the thumbnail out of our memory stream: outputFrame.Commit(); encoder.Commit(); var outputArray = outputStream.ToArray(); outputStream.Close(); That byte array can then be sent to the output stream and to the cache file. Once we’ve gone through this exercise, it’s only natural to wonder whether it was worth the trouble. I ran this method, as well as GDI and WPF resizing over thirty twelve megapixel images for JPEG qualities between 70% and 100% and measured the file size and time to resize. Here are the results: Size of resized images   Time to resize thirty 12 megapixel images Not much to see on the size graph: sizes from WPF and WIC are equivalent, which is hardly surprising as WPF calls into WIC. There is just an anomaly for 75% for WPF that I noted in my previous article and that disappears when using WIC directly. But overall, using WPF or WIC over GDI represents a slight win in file size. The time to resize is more interesting. WPF and WIC get similar times although WIC seems to always be a little faster. Not surprising considering WPF is using WIC. The margin of error on this results is probably fairly close to the time difference. As we already knew, the time to resize does not depend on the quality level, only the size does. This means that the only decision you have to make here is size versus visual quality. This third approach to server-side image resizing on ASP.NET seems to converge on the fastest possible one. We have marginally better performance than WPF, but with some additional peace of mind that this approach is sanctioned for server-side usage by the Windows Imaging team. It still doesn’t work in medium trust. That is a problem and shows the way for future server-friendly managed wrappers around WIC. The sample code for this article can be downloaded from: http://weblogs.asp.net/blogs/bleroy/Samples/WicResize.zip The benchmark code can be found here (you’ll need to add your own images to the Images directory and then add those to the project, with content and copy if newer in the properties of the files in the solution explorer): http://weblogs.asp.net/blogs/bleroy/Samples/WicWpfGdiImageResizeBenchmark.zip WIC tools can be downloaded from: http://code.msdn.microsoft.com/wictools To conclude, here are some of the resized thumbnails at 85% fant:

    Read the article

  • How do I set up live audio streams to a DLNA compliant device?

    - by Takkat
    Is there a way to stream the live output of the soundcard from our 12.04.1 LTS amd64 desktop to a DLNA-compliant external device in our network? Selecting media content in shared directories using Rygel, miniDLNA, and uShare is always fine - but so far we completely failed to get a live audio stream to a client via DLNA. Pulseaudio claims to have a DLNA/UPnP media server that together with Rygel is supposed to do just this. But we were unable to get it running. We followed the steps outlined in live.gnome.org, this answer here, and also in another similar guide. As soon as we select the local audio device, or our GST-Launch stream in the DLNA client Rygel displays the following message and the client states it reached the end of the playlist: (rygel:7380): Rygel-WARNING **: rygel-http-request.vala:97: Invalid seek request This is how we configured GST-Launch in rygel.conf: [GstLaunch] enabled=true launch-items=mypulseaudiosink mypulseaudiosink-title=Audio on @HOSTNAME@ mypulseaudiosink-mime=audio/x-wav mypulseaudiosink-launch=pulsesrc device=<device> ! wavpackenc For <device> we tried with the default sink name, this name appended with .monitor, and in addition with upnp-sink and upnp.monitor that was created when we selected DLNA media server from paprefs. We also tried to encode using lamemp3enc with no luck. These are our pulseaudio modules: http://paste.ubuntu.com/1202913/ These are our sinks: http://paste.ubuntu.com/1202916/ Did we miss any other additional configuration needed to get this running? Are there any other alternatives for sending the audio of our soundcard as live stream to a DLNA client?

    Read the article

  • When selecting Bookmarks in Chromium causes force close

    - by pst007x
    Chromium or Google Chrome in Ubuntu 12.04, selecting Bookmarks from the drop down menu causes Chromium/Chrome to close. Terminal: pst007x@pst007x-Serval-Professional:~$ google-chrome ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream [466:466:24874766730:ERROR:gles2_cmd_decoder.cc(4932)] GL ERROR :GL_INVALID_VALUE : glDeleteProgram: unknown program ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream ALSA lib pcm_dmix.c:957:(snd_pcm_dmix_open) The dmix plugin supports only playback stream [466:466:24877302452:ERROR:gles2_cmd_decoder.cc(4932)] GL ERROR :GL_INVALID_VALUE : glDeleteProgram: unknown program [466:466:24878889327:ERROR:gl_context_glx.cc(163)] Couldn't make context current with X drawable. [466:466:24878999015:ERROR:x11_util.cc(1273)] X Error detected: serial 150, error_code 146 (GLXBadDrawable), request_code 135, minor_code 5 (Unknown) Aborted I have deleted my profile, I have installed the latest Chromium/Chrome and tried BETA and UNSTABLE releases. It seems to be an issue with my Ubuntu installation. Any ideas? Thanks Current Chromium Version: 23.01246.0r153452-Oubuntu1 (precise) Current Chrome Version: 21.0.1180.89-r154005 (Stable) pst007x@pst007x-Serval-Professional:~$ lsb_release -a LSB Version: core-2.0-amd64:core-2.0-noarch:core-3.0-amd64:core-3.0-noarch:core-3.1-amd64:core-3.1-noarch:core-3.2-amd64:core-3.2-noarch:core-4.0-amd64:core-4.0-noarch Distributor ID: Ubuntu Description: Ubuntu 12.04.1 LTS Release: 12.04 Codename: precise pst007x@pst007x-Serval-Professional:~$

    Read the article

  • Apache DVB http video Streaming bandwidth or priority problem

    - by igino manfre'
    I'm streaming few precompressed DVB videos from cloud. The streams are generated from VLC on "impossible" ports (such as 64085, 64086 etc) reverse proxed by Apache on port 80 and 8080. All the generated streams are listed in "http://95.110.164.61/indexv.html". From an ADSL connection with enough downlink bandwidth, recalling the stream generated by VLC (such as "http://95.110.164.61:64087/mpg2_6.4") it flows fluently. Recalling the same stream proxed by Apache ("http://95.110.164.61/mpg2_6.4") the stream stops and goes. The only situation in which the Apache proxed streams flow regularly is from a site connected through 64 Mbps warranted bandwith with RTT to the server less than 10 mseconds. Please note that streams below 2 Mbps are fluently proxed. The system is a single core xeon with windows 2008 R2 on 4 GB of RAM with 1 Gbps of network bandwidth. The drain of computational and bandwidth resources is negligeable, the RAM usage always lower than 50%. On the system I run many VLC streamers. Any of them drains a variable amount of RAM (from about 25 to 70 MB). On the contrary the couple of httpd.exe processes drain no more than 7 MB. Using Wireshark (on the server) I see that VLC directy send to the client much more packets than Apache, and the stream is framgmented on many frames. I'm not a programmer, a newby of Apache. Can anyone please address me to a specific portion of the Apache's huge documentation? Thank you. igino

    Read the article

< Previous Page | 50 51 52 53 54 55 56 57 58 59 60 61  | Next Page >