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  • FFmpeg convert video w/ dropped frames, out of sync

    - by preahkumpii
    I recorded a video using Bandicam with the MJPEG encoder to get the least amount of lag. Now, I am trying to convert that massive file to a h264 avi using ffmpeg. I know there are dropped frames in the video stream...more than 100 in the first two minutes, which I assume is simply because Bandicam dropped some when it couldn't keep up. So, when I convert the file to h264, the video and audio are out of sync, and appear to be more and more out of sync as output video progresses. Here is my basic command in ffmpeg: ffmpeg -i "C:\...\input.avi" -vcodec libx264 -q 5 -acodec libmp3lame -ar 44100 -ac 2 -b:a 128k "C:\...\output.avi" I have tried EVERYTHING I can think of including: -itsoffset [-]00:00:01 Tried this before and after input file. This doesn't work because as the video progresses it becomes more and more out of sync. -async 1 Doesn't work. -vsync 1 Doesn't work, but it does show dropped frames being duplicated. Two inputs of same file with mapping using -map 0:0 -map 1:1. Doesn't work. The source plays just fine. Any ideas how to convert it with ffmpeg and keep the audio and video synced? Thanks.

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  • how to get adobe flash fullscreen video fluid with an atom processor?

    - by Antoine Rodriguez
    My system has an atom N270 + intel i915 graphic card. Under Windows I can enjoy 720p bigbuckbunny youtube video fullscreen without any trouble. Under Ubuntu 12.04 I have laggy and choppy fullscreen video and choppy video when not fullscreen. I've seen that under ubuntu the cpu is almost always at 100% use. What I must do in order to have videos playing well under ubuntu ? I've already tried the following : Force flash gpu detection : (no result) : mkdir /etc/adobe echo "OverrideGPUValidation = 1" | sudo tee -a /etc/adobe/mms.cfg grub options (had results but not enough) : i915_enable_rc6=1 i915_enable_fbc=1 i915_lvds_downclock=1 pcie_aspm=force updated intel drivers (glasen ppa) Using chrome instead of firefox (had impact but not enough)

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  • Proper 16:9 video size for non-HD 4:3 video (for youtube/vimeo)

    - by Xeoncross
    Since High Definition video came out on all the online sites it has changed the default aspect ratio of the player from 4:3 to 16:9. This means that for people posting SD video you have to resize some of your videos to get them to fit right. For example, NTSC DVD quality (aka 480i/p) is 720x480 pixels (width x height). However, low-end High Definition (720i/p) is 1280x720. Resolution Chart Anyway, now that the video players are built for HD you will find that uploading standard quality videos will result in videos that are "letter boxed" which means they have extra black bars on the top and bottom (or sides). Correct me if I'm wrong, but in order to get a 720x480 video to fit a box that is designed for HD the best practice would be to crop some of it off so that it fits as 720x404 since: 16/9 = 1.78 (1.7777777777778) 720/405 = 1.78 405x1.78 = 720.9 The same would stand for 640x480 (old TV quality) video that would need to be 640x360 correct? I'm asking because I'm not sure about all this and whether this is the proper way to fix these letter-boxing/display problems.

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  • ffserver - streaming problem transcodation for input

    - by zozo
    Good day to all. I have a little problem. I'm trying to stream something from a cam to a server and then forward to... somewhere (it will be a site or something). On the computer that I have the cam connected to I use vlc to stream it to the server and there I try to get the stream as an input for a ffserver. The problem is that ffserver doesn't detect the input (regardless of the protocol I use (udp, rtp, etc.)). I suspect a transcoding problem or something like that but I can't find any documentation about that so... Does any1 know what transcodation I should use? Thank you for help and have a great day.

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  • VLC tv card streaming

    - by Franco
    I'm trying to stream the output of my desktop's tv card to my laptop using vlc without success. I have on both pcs ArchLinux installed. I'm stuck here: $ cvlc v4l2:///dev/video0:norm=pal-nc:frequency=543250:size=640x480:channel=0:input-slave=alsa:///dev/dsp:audio=0 --sout '#transcode{vcodec=mp4v,acodec=mpga,vb=3000,ab=256,vt=800000,keyint=80,deinterlace}:standard{access=http,mux=ogg,dst=192.168.0.2:8080}' --ttl 12 VLC media player 1.1.4 The Luggage (revision exported) Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS") Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE") [0x1c9e480] inhibit interface error: Failed to connect to the D-Bus session daemon: /usr/bin/dbus-launch terminated abnormally with the following error: Autolaunch error: X11 initialization failed. [0x1c9e480] main interface error: no suitable interface module [0x1ca1500] main interface error: no suitable interface module [0x1bb3120] main libvlc error: interface "globalhotkeys,none" initialization failed [0x1c9f940] dummy interface: using the dummy interface module... [0x1ca4850] main access out: creating httpd [0x1ebb340] mux_ogg mux: Open And on my laptop: $ vlc http://192.168.0.2:8080 VLC media player 1.1.4.1 The Luggage (revision exported) Blocked: call to unsetenv("DBUS_ACTIVATION_ADDRESS") Blocked: call to unsetenv("DBUS_ACTIVATION_BUS_TYPE") Blocked: call to setlocale(6, "") Blocked: call to sigaction(17, 0xb25c7058, 0xb25c70e4) Warning: call to signal(13, 0x1) Warning: call to signal(13, 0x1) Blocked: call to setenv("ORBIT_SOCKETDIR", "/tmp/orbit-zf", 1) Warning: call to srand(1287690122) Warning: call to rand() Blocked: call to setlocale(6, "") (process:17933): Gtk-WARNING **: Locale not supported by C library. Using the fallback 'C' locale. Warning: call to signal(13, 0x1) Blocked: call to setlocale(6, "") [0x8af5f04] main stream error: cannot pre fill buffer Any idea why this isn't working?

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  • Streaming Flash Video - getting my feet wet

    - by Travis
    I'm working on a project that will involve having a couple thousand short videos online. I haven't done anything with online video before and this is all a bit new to me, so I am looking for some general advice... I would like to use Flowplayer, and I would like to encode the videos as H264s. I am enamoured with Flowplayer's slow motion feature, which if I understand correctly, is only available using a Wowza server. I'm wondering: Is it advisable to use a delivery network of some sort? (Flowplayer seems to have a partnership with HDDN, and recommends them. http://www.hddn.com/) Or would I be better off purchasing Wowza and installing it on our own server? (At first glance, it looks as though signing up with a network like HDDN is much simpler, but perhaps there are problems that come along with this...?) Any tips / warnings of imminent peril would be much appreciated. Thanks in advance.

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  • Streaming music throughout the house on a budget?

    - by greggannicott
    I was wondering whether anyone knew of a way to stream music throughout a house on a budget? I want to avoid spending any money on this (eg. I don't want to buy a d-link style device). It would be ideal if I could use my existing hardware and some open source software. I have three old(ish) PCs knocking around. I'm happy to stick either Windows or Linux on them. They can all be hooked up to speakers. The ideal solution would result in: the same audio being heard from every device (eg. when you hear a beat on one device, you'd hear it at the exact same time on another (so you don't get any echo)) I'd be able to control the source application (eg. the songs lined up) with my iPhone. I realise I'm being cheeky with those two wishes - but you never know your luck. Am I asking for too much, or is there a piece of software/protocol out there with this purpose in mind? I've been searching for sometime now, but haven't had any joy. Thanks in advance.

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  • Trancoding a video with vlc and the MPEG-4 codec with a video bit rate

    - by Nicolas
    When I try to trancode a video with vlc using the MPEG-4 codec and a video bit rate, I get the following error : Either both buffer size and max rate or neither must be specified Here is the command line I used : vlc -vvv myvideo.m2ts --sout-ffmpeg-strict=-2 --sout '#transcode{vcodec=mp4v,vb=2000,acodec=mp4a,ab=96}:standard{access=file,mux=mp4,dst=/path/file.mp4}' It seems that I can't specify a video bit rate with the MPEG-4 codec. (If I remove the vb option, it works). So how can I specify a video bit rate with the MPEG-4 codec ?

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  • How to convert a video to 4:3 or 16:9 aspect ratio without loss of video quality

    - by Linux Jedi
    I uploaded my video to youtube and the highest resolution it appears as is 360p. This is much lower than what I uploaded. I believe that youtube isn't making higher resolutions available for my video because of its aspect ratio. The video is 720x400. How can I convert this to a different aspect ratio without losing any of the picture or picture quality? I don't care if blank space appears around the video so long as the picture doesn't get stretched horizontally or vertically.

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  • custom video icon for a single video file in windows 7 file explorer

    - by MrBrody
    recently I found a video on the net ( a .mp4 file), and when I had it on my computer with Windows7, I noticed its thumbnail was not the average windows 7 video thumbnail (which looks like a piece of video film with a random picture from the movie), but a custom thumbnail! Looking in the file properties did not help find the correct button to change the thumbnail...so I just wonder how he did it! Here is a picture: left: the custom thumbnail, right: the average thumbnail...

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  • Resizing video best practices (frame size)

    - by undefined
    I have read the following which is from Best Practices for Encoding Video with the VP6 Codec on the Adobe website here - http://www.adobe.com/devnet/flash/articles/encoding_video_print.html. It is talking about common video ratios (320x240, 640x480) Although these ratios are standard, and should be used to avoid distorting the video, the size of the encoded video is not set in stone. The original web video sizes used heights and widths that were evenly divisible by 16. This was mandatory for many early codecs. Although this is not necessary for modern codecs, you should stick to even heights and widths. What do they mean by 'even heights and widths'. I am thinking about encoding my video at 400x300 to make it slightly bigger, this is still 4x3 format but should I just stick at 320x240 and resize it on the screen? Clearly there are benefits to this in terms of storage size and delivery costs. In some places on my site I want to show the video at 400x300 but in others I want it to play full screen so this is why I am wondering if a larger original size (400x300) will give better results when blown up. Any thoughts?

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  • Audio/video streaming on Windows platform

    - by bushtucker
    I'm building an interactive language learning application to be used in a classroom environment. The idea is that a teacher should be able to talk to the students (=audio stream to all students), let students talk to each other (= audio P2P) in groups of two or more, let students watch a video coming from a the DVD player or coming from a media server. It should be possible to save the audio/video streams. The teacher should also be able to monitor, take-over or block the desktop of the students. The platform is Windows and it's a desktop application, no web application. The audio delay should be as minimal as poosible. Optionally a student sitting at home should be supported, but it's not a high priority. I am now finished with the classroom control part of the application (login, monitor, block, ...) and want to start the audio and video part. I've been evaluating several options like DirectX, GStreamer and SIP but now I have to make a decision. DirectX seems an obvious choice for the Windows platform, but it only lets me capture and playback audio and video. The encoding/decoding/network part I should do myself. GStreamer contains all kinds of options to capture/encode/stream/save audio and video streams. I've experimented a bit with it (ossbuild) and it does seem to involve a lot of trial and error to make something work: - microphone capture (via directsoundsrc) produces cracking noises on some computers - rtpL16 payloader didn't work well - streaming raw audio over the network only working at a sampling rate of 8000, no higher - there are a lot of errors when receiving mpeg4 video (bad I-frame), on some computers worse than others It is my impression that gstreamer is primary targetted at linux platforms. Development and support for the Windows platform seems to be a little behind. Nevertheless it's a powerful framework that could save me months and years of work. SIP seems to be able to do everything I want, but it is targeted towards telephony and IM. I don't know how flexible SIP is. It seems to me that the SIP layer would just be overhead as I already have a central (teacher) application that can control and setup all the streams. The interesting parts of frameworks like opalvoip and freeswitch are the actual audio/video capture, the encoding and transmission. Does anyone know how these interesting parts relate a framework like gstreamer? Are they easy to integrate into a custom application? Are they flexible enough? Does anyone have experience with all or one of these technologies? Maybe there are even other options I can look at? Many thanks for your advice

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  • Streaming audio - where to start?

    - by Adam Davis
    I need to develop an embedded audio streaming server. Requirements: Voice quality or better Intended for low power wifi transmission Broad support in existing software and devices (ie, windows media player, quicktime, vlc, iPhone, Android, etc). Royalty/patent free, or cheap to license Preferences: Low overhead TCP/IP based streaming protocol Voice grade codec (easy to implement in software, no DSP, 32bit CPU if needed) Would be nice if it supported HTML5 browsers, but is there any codec (such as raw) that is supported by the latest browsers that is lower overhead than MP3? Therefore: What are the relevant streaming protocols I should be looking at? What are the relevant codecs I should be looking at? What transport streams should I be looking at? What am I missing, or where else should I be looking for this type of need?

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  • Streaming desktop with avconv - severe sound issues

    - by Tommy Brunn
    I'm trying to do some live streaming in Ubuntu 12.10, but I'm having some problems with audio. More specifically, the quality is complete garbage and it's at least 10 seconds out of sync with the video. I'm using an excellent guide found here to set up my loopback devices so that I can combine the desktop audio with the microphone input. It seems to work, as I'm able to stream both audio and video to Twitch.tv. But, as I said, the audio quality is terrible. The microphone audio is very, very low, but if I increase it, I get a horrible garbled sound that is absolutely unbearable. Nothing like that is present during VoIP calls or when recording sound alone with the sound recorder, so it's not an issue with the microphone itself. The entire audio stream is also delayed about 10-15 seconds compared to the video stream. I put together an imgur album of my settings. Here is some example output from when I'm streaming: avconv version 0.8.4-6:0.8.4-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Nov 6 2012 16:51:11 with gcc 4.7.2 [x11grab @ 0x162fd80] device: :0.0+570,262 -> display: :0.0 x: 570 y: 262 width: 1280 height: 720 [x11grab @ 0x162fd80] shared memory extension found [x11grab @ 0x162fd80] Estimating duration from bitrate, this may be inaccurate Input #0, x11grab, from ':0.0+570,262': Duration: N/A, start: 1353181686.735113, bitrate: 884736 kb/s Stream #0.0: Video: rawvideo, bgra, 1280x720, 884736 kb/s, 30 tbr, 1000k tbn, 30 tbc [alsa @ 0x163fce0] capture with some ALSA plugins, especially dsnoop, may hang. [alsa @ 0x163fce0] Estimating duration from bitrate, this may be inaccurate Input #1, alsa, from 'pulse': Duration: N/A, start: 1353181686.773841, bitrate: N/A Stream #1.0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' [buffer @ 0x1641ec0] w:1280 h:720 pixfmt:bgra [scale @ 0x1642480] w:1280 h:720 fmt:bgra -> w:852 h:480 fmt:yuv420p flags:0x4 [libx264 @ 0x165ae80] VBV maxrate unspecified, assuming CBR [libx264 @ 0x165ae80] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle SSE4.2 [libx264 @ 0x165ae80] profile Main, level 3.1 [libx264 @ 0x165ae80] 264 - core 123 r2189 35cf912 - H.264/MPEG-4 AVC codec - Copyleft 2003-2012 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x1:0x111 me=hex subme=6 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=4 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=3 b_pyramid=0 b_adapt=1 b_bias=0 direct=1 weightb=0 open_gop=1 weightp=1 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=30 rc=cbr mbtree=1 bitrate=712 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=712 vbv_bufsize=512 nal_hrd=none ip_ratio=1.25 aq=1:1.00 Output #0, flv, to 'rtmp://live.justin.tv/app/live_23011330_Pt1plSRM0z5WVNJ0QmCHvTPmpUnfC4': Metadata: encoder : Lavf53.21.0 Stream #0.0: Video: libx264, yuv420p, 852x480, q=-1--1, 712 kb/s, 1k tbn, 30 tbc Stream #0.1: Audio: libmp3lame, 44100 Hz, 2 channels, s16, 712 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #1:0 -> #0:1 (pcm_s16le -> libmp3lame) Press ctrl-c to stop encoding frame= 17 fps= 0 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 32 fps= 31 q=0.0 size= 0kB time=10000000000.00 bitrate= 0.0kbitframe= 40 fps= 23 q=29.0 size= 44kB time=0.03 bitrate=13786.2kbits/s dup=frame= 47 fps= 21 q=31.0 size= 93kB time=2.73 bitrate= 277.7kbits/s dup=0frame= 62 fps= 23 q=29.0 size= 160kB time=3.23 bitrate= 406.2kbits/s dup=0frame= 77 fps= 24 q=23.0 size= 209kB time=3.71 bitrate= 462.5kbits/s dup=0frame= 92 fps= 25 q=20.0 size= 267kB time=4.91 bitrate= 445.2kbits/s dup=0frame= 107 fps= 25 q=20.0 size= 318kB time=5.41 bitrate= 482.1kbits/s dup=0frame= 123 fps= 26 q=18.0 size= 368kB time=5.96 bitrate= 505.7kbits/s dup=0frame= 139 fps= 26 q=16.0 size= 419kB time=6.48 bitrate= 529.7kbits/s dup=0frame= 155 fps= 27 q=15.0 size= 473kB time=7.00 bitrate= 553.6kbits/s dup=0frame= 170 fps= 27 q=14.0 size= 525kB time=7.52 bitrate= 571.7kbits/s dup=0 frame= 180 fps= 25 q=-1.0 Lsize= 652kB time=7.97 bitrate= 670.0kbits/s dup=0 drop=32 //Here I stop the streaming video:531kB audio:112kB global headers:0kB muxing overhead 1.345945% [libx264 @ 0x165ae80] frame I:1 Avg QP:30.43 size: 39748 [libx264 @ 0x165ae80] frame P:45 Avg QP:11.37 size: 11110 [libx264 @ 0x165ae80] frame B:134 Avg QP:15.93 size: 27 [libx264 @ 0x165ae80] consecutive B-frames: 0.6% 0.0% 1.7% 97.8% [libx264 @ 0x165ae80] mb I I16..4: 7.3% 0.0% 92.7% [libx264 @ 0x165ae80] mb P I16..4: 0.1% 0.0% 0.1% P16..4: 49.1% 1.2% 2.1% 0.0% 0.0% skip:47.4% [libx264 @ 0x165ae80] mb B I16..4: 0.0% 0.0% 0.0% B16..8: 0.1% 0.0% 0.0% direct: 0.0% skip:99.9% L0:42.5% L1:56.9% BI: 0.6% [libx264 @ 0x165ae80] coded y,uvDC,uvAC intra: 82.3% 87.4% 71.9% inter: 7.1% 8.4% 7.0% [libx264 @ 0x165ae80] i16 v,h,dc,p: 27% 29% 16% 28% [libx264 @ 0x165ae80] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 22% 21% 14% 8% 8% 8% 7% 5% 7% [libx264 @ 0x165ae80] i8c dc,h,v,p: 47% 22% 20% 11% [libx264 @ 0x165ae80] Weighted P-Frames: Y:0.0% UV:0.0% [libx264 @ 0x165ae80] ref P L0: 96.4% 3.6% [libx264 @ 0x165ae80] kb/s:474.19 Received signal 2: terminating. Any ideas on how I can resolve this? The video delay is perfectly acceptable, so I wouldn't think that it's a network issue that's causing the delay in the audio. Any help would be appreciated.

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  • Group Video Chat On iPad With Fring App

    - by Gopinath
    Apple’s Facetime is the simplest and most easy to use video chat application available for iOS devices and Apple Macs. Facetime lets you have a one-to-one video chat, but what about having a group chat on your iPad with a bunch of friends? Here comes Fring App for iPad that lets you group chat up to 4 members at a time. Unlike Facetime Fring does not impose any restriction on the network connectivity for video chatting. You can initiate a group video chatting on any network (3G, Wi-Fi, 4G, etc.) but Wi-Fi is the most preferred option for smooth video streaming. Also Fring is a cross platform application(runs on iOS, Android & Nokia), so your group video chat session can have a mix of devices – iPads, Android smartphones/tablets and Nokia mobiles. Anyone mobile device with a front facing  cam and Fring app is allowed to join the party. Here is the promotional ad of Fring’s group video chatting application By the way did I say that Fring is a free app? Group video on iPad at no cost!!! Download Fring from Apple’s AppStore This article titled,Group Video Chat On iPad With Fring App, was originally published at Tech Dreams. Grab our rss feed or fan us on Facebook to get updates from us.

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  • Streaming encoded video in Adobe AIR Application

    - by RIA Developer
    Hi, I am developing a desktop application in Adobe AIR that will be used to stream the user's camera video to a wowza media server. I want to encode the video on the fly, means transmit the H.264 encoded video instead of the default flash player encoded video for quality purpose. Is there any way around for this? Waiting for the help from people around, Rick

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  • HD Video Capture Card w/ Good API?

    - by Sheep Slapper
    Does anyone here know of a good HD video capture card that has a good (comprehensive) API? I administer a few servers that do some video encoding right now, but when we make the switch to HD cameras, they won't be sufficient. In addition to this, the servers we have now are black boxes, closed to me except to start/stop the video capture device. I'd like to be able to roll my own, so we can better integrate it with our existing systems, but I know almost nothing about what kind of HD capture cards are out there, and if I can avoid spending money just to test their APIs that would rock. So does anyone have any experience with this? All our other software is in C#, and I'd like to set up the new servers with web interfaces to start/stop the capture (also in C#, using .NET 3.5 probably). I'm not sure how language specific these APIs would be, but that's what I'm working with just as a reference point. I appreciate any help the community can give!

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  • Video from vhs via USB capture device on Linux

    - by Juanlu001
    I want to transfer video from a VHS tape to my computer with Arch Linux, using a USB capture device that comes with Roxio Easy VHS to DVD, which I recently bought. I tried to plug in the device, and it was properly recognized. From /var/log/messages.log, em28xx #0: Identified as Pinnacle Dazzle DVC 90/100/101/107 / Kaiser Baas Video to DVD maker / Kworld DVD Maker 2 (card=9) saa7115 3-0025: saa7113 found (1f7113d0e100000) @ 0x4a (em28xx #0) em28xx #0: Config register raw data: 0x50 em28xx #0: AC97 vendor ID = 0xffffffff em28xx #0: AC97 features = 0x6a90 em28xx #0: Empia 202 AC97 audio processor detected em28xx #0: v4l2 driver version 0.1.2 em28xx #0: V4L2 video device registered as video1 em28xx #0: V4L2 VBI device registered as vbi0 Is there any software that enables me to capture video coming from this device? I don't mind if it is console or GUI based.

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  • Local Live Quicktime Video Broadcast, latency?

    - by Snowwire
    I'm looking into the feasibility of using a local server to distribute live video of a conference to delegates in the same room. They would still hear the live audio coming from the speaker, so only the video would be streamed. I was considering a Darwin Steaming Server (a lot of iPhone users to support) and encoding with H.264. My main concern is latency across the network. Even with everything running locally, would there be lip sync issues between the live audio and the 'live' video stream? It feels like there will be problems given the encoding, broadcasting, decoding to be completed, but I've never done any like this before so thought I would check. Thanks

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  • 500 MB Avi video from CD lagging/glitching when reproduced

    - by Caki Esther
    I created an extra cd with Nero Burning that contains both audio tracks (I can hear them correctly) and a .avi presentation (pretty big one, 540 MB on a 700 MB cd). The audio is fine but the problem is that when the video is played from the cd (with whatever media player: Windows Media Player, VLC, etc..) it lags/glitches/stutters. I'd like the video to be smooth, how should I burn the video to reduce this effect? I mean: what kind of compression/format and why?

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  • Skype Video not working 13.04 & 13.10?

    - by MrA
    I am trying to fix the issue with the video calls. I can check the video from the option menu, it works fine. But it does not work while calling someone. It shows video disabled. How can i fix this issue? I am using Ubuntu 13.10 and skype version 4.2.0.11 I checked all sites including below links but could not find solution: Skype no video in 13.04 Skype Video Call Not Working Ubuntu 11.10 No video skype found Video doesn't work well in skype http://www.youtube.com/watch?v=WeaLCzB4qy8

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  • Video files from another partition or volume does not play

    - by user49523
    Some video files from another partition or volume does not display in some video displays but display on another and crashes on others. Seem to play on these video players: vlc smplayer gnome-mplayer Seem to have some issues in these video players: totem -some videos are play while others not banshee -some videos are play while others not bangarang -few videos are play while others not and it crashes and happens also in home folder of filesystem dragon player -crashes and seem not to display any video and happens also in home folder of filesystem Why this happens? Is this a bug? On ubuntu 11.10, every video display play all the video from another partition or from home filesystem.

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