Search Results

Search found 581 results on 24 pages for 'codec'.

Page 8/24 | < Previous Page | 4 5 6 7 8 9 10 11 12 13 14 15  | Next Page >

• Convert png sequence to x264 with ffmpeg

  - by Thucydides411

  I am trying to convert a series of pngs into an mp4 video. I am using ffmpeg, and want to encode the video with the x264 codec. Using the command ffmpeg -y -r 30 -b 1800k -i _tmp%04d.png -vcodec libx264 out.mp4 I get the following warning message Incompatible pixel format 'bgra' for codec 'libx264', auto-selecting format 'yuv420p' My understanding is that there is an alpha channel in the pngs, which the x264 encoder cannot handle. Is there a way to get around this problem? Is there, for example, a way to get the encoder to ignore the alpha channel (my pngs don't actually have any transparent elements)? I'm aware that I could batch convert the pngs beforehand to strip the alpha channel, but the sequence of images is produced by another program, and having to preprocess the images each time I make a video would be less than optimal. Edit: After stripping the alpha channel from each frame using the command convert in.png -background white -flatten +matte out.png ffmpeg gives the warning message Incompatible pixel format 'pal8' for codec 'libx264', auto-selecting format 'yuv420p' so still no dice.

  Read the article

• Convert mkv to mp4 with ffmpeg

  - by JohnS

  When I try converting mkv to mp4 using ffmpeg, the following error occurs: [ipod @ 0x16fa0a0] Application provided invalid, non monotonically increasing dts to muxer in stream 0: -2 = -2 av_interleaved_write_frame(): Invalid argument I used this command to convert the file: ffmpeg -i input.mkv -vcodec copy -acodec copy -absf aac_adtstoasc output.m4v The input file has the following characteristics: mediainfo input.mkv General Unique ID : 200459305952356554213392832683163418790 (0x96CF0ED8DB5914CBB9E18163689280A6) Complete name : input.mkv Format : Matroska Format version : Version 2 File size : 1.46 GiB Duration : 1h 5mn Overall bit rate : 3 168 Kbps Encoded date : UTC 2010-09-26 21:44:02 Writing application : mkvmerge v2.9.5 ('Tu es le seul') built on Jun 17 2009 16:28:30 Writing library : libebml v0.7.8 + libmatroska v0.8.1 Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 4 frames Codec ID : V_MPEG4/ISO/AVC Duration : 1h 5mn Bit rate : 2 910 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 25.000 fps Color space : YUV Chroma subsampling : 4:2:0 Bit depth : 8 bits Scan type : Progressive Bits/(Pixel*Frame) : 0.126 Stream size : 1.31 GiB (90%) Writing library : x264 core 105 r1724 b02df7b Encoding settings : cabac=1 / ref=3 / deblock=1:0:0 / analyse=0x3:0x113 / me=hex / subme=6 / psy=1 / psy_rd=1.00:0.00 / mixed_ref=0 / me_range=16 / chroma_me=1 / trellis=1 / 8x8dct=1 / cqm=0 / deadzone=21,11 / fast_pskip=0 / chroma_qp_offset=-2 / threads=18 / sliced_threads=0 / nr=0 / decimate=1 / interlaced=0 / constrained_intra=0 / bframes=3 / b_pyramid=2 / b_adapt=1 / b_bias=0 / direct=3 / weightb=1 / open_gop=0 / weightp=0 / keyint=250 / keyint_min=25 / scenecut=40 / intra_refresh=0 / rc=2pass / mbtree=0 / bitrate=2910 / ratetol=1.0 / qcomp=0.60 / qpmin=10 / qpmax=51 / qpstep=4 / cplxblur=20.0 / qblur=0.5 / ip_ratio=1.40 / pb_ratio=1.30 / aq=1:1.00 Default : Yes Forced : No Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Mode extension : CM (complete main) Codec ID : A_AC3 Duration : 1h 5mn Bit rate mode : Constant Bit rate : 256 Kbps Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Bit depth : 16 bits Compression mode : Lossy Stream size : 121 MiB (8%) Language : English Default : Yes Forced : No Being new to ffmpeg, I'm not sure what the error means or how to correct it. Thanks!

  Read the article

• MPlayer does not work

  - by Soham Pal

  Using the xubuntu desktop, on Ubuntu Raring updated from Quantal. MPlayer never really worked. No video, no audio, nothing. I really can't be any more helpful, so here's the log: petey@home-pc:~$ mplayer "/home/petey/Downloads/Polar Bear Cafe (480p)HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv" MPlayer SVN-r35984-4.7 (C) 2000-2013 MPlayer Team Playing /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/[HorribleSubs] Polar Bear Cafe - 01 [480p].mkv. libavformat version 55.0.100 (internal) libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0 [lavf] stream 2: subtitle (ass), -sid 0 VIDEO: [H264] 848x480 0bpp 23.810 fps 0.0 kbps ( 0.0 kbyte/s) Clip info: creation_time: 2012-04-05 21:36:10 Load subtitles in /home/petey/Downloads/Polar Bear Cafe (480p)[HorribleSubs]/ Can't open /dev/fb0: Permission denied [fbdev2] Can't open /dev/fb0: Permission denied VO: [v4l2] No such file or directory vo_cvidix: No vidix driver name provided, probing available ones (-v option for details)! [cyberblade] Error occurred during pci scan: Operation not permitted [mach64] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [mga] Error occurred during pci scan: Operation not permitted [nvidia_vid] Error occurred during pci scan: Operation not permitted [pm3] Error occurred during pci scan: Operation not permitted [radeon] Error occurred during pci scan: Operation not permitted [rage128] Error occurred during pci scan: Operation not permitted [s3_vid] Error occurred during pci scan: Operation not permitted [SiS] Error occurred during pci scan: Operation not permitted [unichrome] Error occurred during pci scan: Operation not permitted [VO_SUB_VIDIX] Couldn't find working VIDIX driver. ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family libavcodec version 55.0.100 (internal) Selected video codec: [ffh264] vfm: ffmpeg (FFmpeg H.264) ========================================================================== ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders AUDIO: 44100 Hz, 2 ch, floatle, 0.0 kbit/0.00% (ratio: 0->352800) Selected audio codec: [ffaac] afm: ffmpeg (FFmpeg AAC (MPEG-2/MPEG-4 Audio)) ========================================================================== [AO OSS] audio_setup: Can't open audio device /dev/dsp: No such file or directory DVB card number must be between 1 and 4 AO: [null] 44100Hz 2ch floatle (4 bytes per sample) Starting playback... Movie-Aspect is 1.78:1 - prescaling to correct movie aspect. VO: [null] 848x480 = 854x480 Planar YV12 A: 4.7 V: 4.7 A-V: 0.002 ct: 0.083 0/ 0 22% 0% 0.5% 0 0 MPlayer interrupted by signal 2 in module: sleep_timer A: 4.7 V: 4.7 A-V: 0.001 ct: 0.083 0/ 0 21% 0% 0.5% 0 0 Exiting... (Quit)

  Read the article

• Unable To Get Sound Working to External Speaker on HP TouchSmart 320 on 11.04 or 11.10

  - by Schof

  This is an HP TouchSmart 320, model number 320-1200m. I'm using Ubuntu 11.04. Hardware information: root@hp320:/home/mpower# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Generic [HD-Audio Generic], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 root@hp320:~$ cat /proc/asound/card0/codec#0 | grep Codec Codec: IDT 92HD91BXX Sound to headphone jack works properly, but sound to built-in speakers does not work. I have installed Windows, and with Windows 7 installed, all audio hardware works properly, so this isn't a hardware fault. I've looked at https://help.ubuntu.com/community/HdaIntelSoundHowto and have been unable to find my card in http://www.kernel.org/doc/Documentation/sound/alsa/HD-Audio-Models.txt . I have tried adding almost every conceivable model in the line "options snd-hda-intel model=MODEL" line I added to /etc/modprobe.d/alsa-base.conf. Update 2011-11-09 2:31 PM PST: I've gone to Control Center - Sound Preferences to attempt settings that make sound work. The "Hardware" tab shows one device: "Internal Audio 1 Output / 1 Input Analog Stereo Duplex." There are two output profiles listed in the selection box at the bottom of the tag: Analog Stereo Duplex and Analog Stereo Output. Neither cause sound to emit from the speakers. I've also run alsamixer on the command-line and ensured that everything is set to maximum and nothing is muted. Update 2011-11-09 5:15 PM PST: I've replicated the exact same symptoms in 11.10. Update 2011-11-10 11:31 AM PST: I've filed a bug in launchpad: https://launchpad.net/ubuntu/+source/alsa-driver/+bug/888703

  Read the article

• Bsplayer - load audio tracks from external files

  - by torran

  I have a movie file: Video ID : 1 Format : AVC Format/Info : Advanced Video Codec Format profile : [email protected] Format settings, CABAC : Yes Format settings, ReFrames : 5 frames Muxing mode : Container [email protected] Codec ID : V_MPEG4/ISO/AVC Duration : 54mn 13s Bit rate : 3 380 Kbps Nominal bit rate : 3 459 Kbps Width : 1 280 pixels Height : 720 pixels Display aspect ratio : 16:9 Frame rate : 23.976 fps Resolution : 8 bits Colorimetry : 4:2:0 Scan type : Progressive Bits/(Pixel*Frame) : 0.153 Stream size : 1.28 GiB (88%) Writing library : x264 core 88 r1471 1144615 Audio ID : 2 Format : AC-3 Format/Info : Audio Coding 3 Codec ID : A_AC3 Duration : 54mn 16s Bit rate mode : Constant Bit rate : 384 Kbps Channel(s) : 6 channels Channel positions : Front: L C R, Side: L R, LFE Sampling rate : 48.0 KHz Stream size : 149 MiB (10%) and additional audio files in same folder: .mp3 and .ac3. How can I load them with bsplayer? Right click-audio-audio streams is empty. If i open the movie with media players classic I can switch audio files.

  Read the article

• Motion - can't get streaming working from a webcam

  - by Emmanuel Brunet

  I'm trying to record a video stream from my Tenvis IP camera with motion 3.2.12 on Debian 7.5. I used the standard debian package with sudo apt-get install motion Assume my DNS IP cam is webcam, user : admin, password : password /etc/motion/motion.conf Bellow are my configuration file settings : netcam_url http://webcam/videostream.cgi netcam_userpass admin:password target_dir /media/videos/log/motion # The mini-http server listens to this port for requests (default: 0 = disabled) webcam_port 1234 ffmpeg_cap_new on ffmpeg_video_codec mpeg4 output_motion off snapshot_interval 0 # Quality of the jpeg (in percent) images produced (default: 50) webcam_quality 50 # Output frames at 1 fps when no motion is detected and increase to the # rate given by webcam_maxrate when motion is detected (default: off) webcam_motion on # Maximum framerate for webcam streams (default: 1) webcam_maxrate 15 # Restrict webcam connections to localhost only (default: on) webcam_localhost on # Limits the number of images per connection (default: 0 = unlimited) # Number can be defined by multiplying actual webcam rate by desired number of seconds # Actual webcam rate is the smallest of the numbers framerate and webcam_maxrate webcam_limit 0 control_port 8080 control_authentication admin:password Issue #1 when I try display http:/localhost:1234 the browser looks frozen, no HTTP 404 received but it stills waiting for data it seems .. Issue #2 in the output directory motion writes a lot of jpeg snapshots althought I just want to have several video sequenced files. Log I run motion in interactive mode in a terminal, here is the ouput root@mercure:/etc/motion# motion -c motion-1.0.conf [0] Processing thread 0 - config file motion-1.0.conf [0] Motion 3.2.12 Started [0] ffmpeg LIBAVCODEC_BUILD 3482368 LIBAVFORMAT_BUILD 3478785 [0] Thread 1 is from motion-1.0.conf [0] motion-httpd/3.2.12 running, accepting connections [0] motion-httpd: waiting for data on port TCP 8080 [1] Thread 1 started [1] Resizing pre_capture buffer to 1 items [1] Started stream webcam server in port 1234 [1] avcodec_open - could not open codec: Operation now in progress [1] ffopen_open error creating (new) file [~/tmp/motion/01-20140603165303.avi]: Operation now in progress [1] File of type 1 saved to: ~/tmp/motion/01-20140603165303-01.jpg [1] Thread exiting [1] Calling vid_close() from motion_cleanup [1] vid_close: calling netcam_cleanup [1] netcam camera handler: finish set, exiting [0] Motion thread 1 restart [1] Thread 1 started [1] Resizing pre_capture buffer to 1 items [1] Started stream webcam server in port 1234 [1] avcodec_open - could not open codec: Resource temporarily unavailable [1] ffopen_open error creating (new) file [~/tmp/motion/01-20140603165329.avi]: Resource temporarily unavailable [1] File of type 1 saved to: ~/tmp/motion/01-20140603165329-00.jpg [1] Thread exiting [1] Calling vid_close() from motion_cleanup [1] vid_close: calling netcam_cleanup [1] netcam camera handler: finish set, exiting [0] Motion thread 1 restart [1] Thread 1 started [1] Resizing pre_capture buffer to 1 items [1] Started stream webcam server in port 1234 [1] avcodec_open - could not open codec: Connection reset by peer [1] ffopen_open error creating (new) file [~/tmp/motion/01-20140603165355.avi]: Connection reset by peer [1] File of type 1 saved to: ~/tmp/motion/01-20140603165355-06.jpg [1] Thread exiting [1] Calling vid_close() from motion_cleanup [1] vid_close: calling netcam_cleanup [0] httpd - Finishing [0] httpd Closing [0] httpd thread exit [1] netcam camera handler: finish set, exiting [0] Motion thread 1 restart [1] Thread 1 started [1] Resizing pre_capture buffer to 1 items [1] Started stream webcam server in port 1234 It doesn't find the codec ... avcodec_open - could not open codec: Operation now in progress I've changed fmpeg_video_codec from mpeg4 to swf the result is the same. When using flv format motion writes a lot of .jpg image but I can't see anything at http://localhost:1234 [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-00.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-01.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-02.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-03.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-04.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-05.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171035-06.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171036-00.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171036-01.jpg [1] File of type 1 saved to: ~/tmp/motion/01-20140603171036-02.jpg Any idea just to get the video stream recoded on my local disk ?

  Read the article

• Convert old AVI files to a modern format

  - by iWerner

  Hi, we have a collection of old home videos that were saved in AVI format a long time ago. I want to convert these files to a more modern format because the Totem Movie Player that comes with Ubuntu 10.4 seems to be the only program capable of playing them. The files seem to be encoded with a MJPEG codec, and playing them in VLC or Windows Media Player plays only the sound but there is no video. Avidemux was able to open the files, but the quality of the video is severely degraded: The video skips frames and is interlaced (it's not interlaced when playing it in Totem). Neither ffmpeg nor mencoder seems to be able to read the video stream. mencoder reports that it is using ffmpeg's codec. Here's a section from its output: ========================================================================== Opening video decoder: [ffmpeg] FFmpeg's libavcodec codec family [mjpeg @ 0x92a7260]mjpeg: using external huffman table [mjpeg @ 0x92a7260]mjpeg: error using external huffman table, switching back to internal Unsupported PixelFormat -1 Selected video codec: [ffmjpeg] vfm: ffmpeg (FFmpeg MJPEG) while running ffmpeg produces the following: $ ffmpeg -i input.avi output.avi FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --extra-version=4:0.5.1-1ubuntu1 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-vdpau --enable-bzlib --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --enable-zlib --disable-stripping --disable-vhook --enable-runtime-cpudetect --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --enable-libdc1394 --enable-shared --disable-static libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 1 / 52.20. 1 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 0. 4. 0 / 0. 4. 0 libswscale 0. 7. 1 / 0. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Mar 4 2010 12:35:30, gcc: 4.4.3 [avi @ 0x87952c0]non-interleaved AVI Input #0, avi, from 'input.avi': Duration: 00:00:15.24, start: 0.000000, bitrate: 22447 kb/s Stream #0.0: Video: mjpeg, yuvj422p, 720x544, 25 tbr, 25 tbn, 25 tbc Stream #0.1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s Output #0, avi, to 'output.avi': Stream #0.0: Video: mpeg4, yuv420p, 720x544, q=2-31, 200 kb/s, 90k tbn, 25 tbc Stream #0.1: Audio: mp2, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 0 fps= 0 q=0.0 Lsize= 143kB time=15.23 bitrate= 76.9kbits/s video:0kB audio:119kB global headers:0kB muxing overhead 20.101777% So the problem is that output does not contain any video, as evidenced by the video:0kB at the end. In all of the above cases the audio comes out fine. So my question is: What can I do to convert these files to a more modern format with more modern codecs?

  Read the article

• Protocol specific channel handlers

  - by Mickael Marrache

  I'm writing an application server that will receive SIP and DNS messages from the network. When I receive a message from the network, I understand from the documentation that at first, I get a ChannelBuffer. I would like to determine which kind of message has been received (SIP or DNS) and to decode it. To determine the message type, I can dedicate port to each type of message, but I would be interested to know if there exist another solution for that. My question is more about how to decode the ChannelBuffer. Is there a ChannelHandler provided by Netty to decode SIP or DNS messages? If not, what would be the right place in the type hierarchy to write my custom ChannelHandler? To illustrate my question, let's take as example the HttpRequestDecoder, the hierarchy is: java.lang.Object org.jboss.netty.channel.SimpleChannelUpstreamHandler org.jboss.netty.handler.codec.frame.FrameDecoder org.jboss.netty.handler.codec.replay.ReplayingDecoder<HttpMessageDecoder.State> org.jboss.netty.handler.codec.http.HttpMessageDecoder org.jboss.netty.handler.codec.http.HttpRequestDecoder Also, do I need to use two different ChannelHandler for decoding and encoding, or is there a possibility to use a single ChannelHandler for both? Thanks

  Read the article

• Thumbnail image saved with worse quality on Windows Server 2003

  - by Angelo

  Hello, In asp.net 2.0 application I am trying to create thumbnails from uploaded images. However when I test the application on my PC under Windows7 it works fine, but on the real Windows 2003 Server the resized image has worse quality. Where this difference could come from? Different JPEG codec or what, if Yes how it can be updated on Win 2003 Server? Thanks! Here is the code: Resize of the Image: Bitmap newBmp = new Bitmap(imgWidth, imgHeight, PixelFormat.Format24bppRgb); newBmp.SetResolution(inputBmp.HorizontalResolution, inputBmp.VerticalResolution); //Create a graphics object attached to the new bitmap Graphics newBmpGraphics = Graphics.FromImage(newBmp); newBmpGraphics.InterpolationMode = InterpolationMode.HighQualityBicubic; newBmpGraphics.SmoothingMode = SmoothingMode.HighQuality; newBmpGraphics.PixelOffsetMode = PixelOffsetMode.HighQuality; newBmpGraphics.DrawImage(inputBmp, new Rectangle(0, 0, imgWidth, imgHeight), new Rectangle(0, 0, inputBmp.Width, inputBmp.Height), GraphicsUnit.Pixel); Save of the Image: System.IO.Stream imgStream = new System.IO.MemoryStream(); //Get the ImageCodecInfo for the desired target format ImageCodecInfo destCodec = FindCodecForType(ImageMimeTypes.JPEG); if (destCodec == null) { //No codec available for that format throw new ArgumentException("The requested format image/jpeg does not have an available codec installed", "destFormat"); } //Create an EncoderParameters collection to contain the //parameters that control the dest format's encoder EncoderParameters destEncParams = new EncoderParameters(1); EncoderParameter qualityParam = new EncoderParameter(System.Drawing.Imaging.Encoder.Quality,(long)quality); destEncParams.Param[0] = qualityParam; //Save w/ the selected codec and encoder parameters inputBmp.Save(imgStream, destCodec, destEncParams); Bitmap destBitmap = new Bitmap(imgStream);

  Read the article

• Totem not recognising realmedia cook decoder

  - by ageis23

  Hi I'm having trouble with totem not being able to play a rmvb file with cook encoded sound. It says the codec is missing but gst-inspect says there is an FFMPEG based codec installed.

  Read the article

• If I use a video format converter to change a movie from AVI to MKV, will quality stay the same?

  - by Matt

  I know they're both container formats and what matters is the actual codec used, but what I don't know is if video converting software will do anything to change the codec, or if it just repackages. The reason I need to know is that I have several .avi files with subtitle files, and I'm wanting to turn them into .mkv so I can attach the subs and not need a second subtitle file anymore. Will my new .mkv files be identical in video and audio quality to the original .avi?

  Read the article

• Youtube "unable to convert video file"

  - by Alexandra

  I encoded some videos from a dvd format (mpeg 2 I think) to h246 (using a .mkv container). When I upload them to youtube, most of them work, but there are a few that don't. After I upload it, I get the message "Failed (unable to convert video file)" What could be the problem because all the videos are the same format, and only a few of them fail. When I click upload details, while uploading, the file seems to be recognized by youtube: Format: MATROSKA Dimensions: 704 x 480 px Video codec: H264 Audio codec: AAC

  Read the article

• How do I write raw binary data in Python?

  - by Chris B.

  I've got a Python program that stores and writes data to a file. The data is raw binary data, stored internally as str. I'm writing it out through a utf-8 codec. However, I get UnicodeDecodeError: 'charmap' codec can't decode byte 0x8d in position 25: character maps to <undefined> in the cp1252.py file. This looks to me like Python is trying to interpret the data using the default code page. But it doesn't have a default code page. That's why I'm using str, not unicode. I guess my questions are: How do I represent raw binary data in memory, in Python? When I'm writing raw binary data out through a codec, how do I encode/unencode it?

  Read the article

• Record Screen Activity with CamStudio

  - by Asian Angel

  Sometimes a visual demonstration works much better than a list of instructions. If you need to make a demo video for family and/or friends then you might want to have a look at CamStudio. Using CamStudio To get properly set up you will need to install two different files (the main program followed by the codec). Once that is done you are ready to get started. When you start the program you will see a surprisingly small window. Notice the highlighted Record to text…it serves as a visual indicator for the video type selected for recording. Before you start creating a video it would be a good idea to look through some of the settings. The first one to look at is the region or area that you want to record. Next you will want to look through the video options since these will affect the quality and final size of your video files. The default setting for quality is 70…adjust that to the level that best suits your needs. Note: For our example we maxed out the various video settings for best quality. On our system Microsoft Video 1 was listed as the default compressor but as you can see there were other options available. You can configure the settings for the compressor you want to use if desired. Keep in mind that each compressor will have unique settings of their own, so if you change it, be certain to go back and check. We decided to use the CamStudio Lossless Codec for our example (it gave the best results while trying the software). Going back to the main window you can toggle back and forth between .avi and .swf output using the last button. Once you are satisfied with the settings click on the red record button to start. If you need to pause while recording or stop recording click on the system tray icon and select the appropriate command. When you are finished recording, you will be presented with the save file window. Browse for the desired save location and name your new file. Once you have saved the file the movie player window will automatically open so that you view your new video. Our sample video shown here is at 50% of original size so may look slightly “gritty”. The detail was much better at 100%. If you decide to record and save as .swf the process will be identical to recording in .avi format until the movie player window opens. At that time the conversion process from .avi to .swf will begin. When complete you will have a new flash video and html file that goes with it. Depending on which browser you have set as default, you may run into a small problem when the preview for your new .swf file tries to open. There is a small bug in the generated html file. You can use this work-around or… Just open the .swf file directly in your favorite browser. Conclusion CamStudio may not produce the highest quality videos, but it’s free and does a very nice job nonetheless. If you are working on a tight budget or only need to make an occasional video then CamStudio is a very sensible choice. Links Download CamStudio Stable Version & CamStudio Codec *Download links are approximately half-way down the page. Download CamStudio Stable Version & CamStudio Codec at SourceForge *Beta version also available here. Similar Articles Productive Geek Tips Get the Classic Style Network Activity Indicator Back in Windows 7How To Copy a DVD with VLC 1.0ALLCapture 3.0 [Review]Listen and Record Over 12,000 Online Radio Stations with RadioSureGeek Reviews: Play And Record Internet Radio With Screamer Radio TouchFreeze Alternative in AutoHotkey The Icy Undertow Desktop Windows Home Server – Backup to LAN The Clear & Clean Desktop Use This Bookmarklet to Easily Get Albums Use AutoHotkey to Assign a Hotkey to a Specific Window Latest Software Reviews Tinyhacker Random Tips Xobni Plus for Outlook All My Movies 5.9 CloudBerry Online Backup 1.5 for Windows Home Server Snagit 10 TimeToMeet is a Simple Online Meeting Planning Tool Easily Create More Bookmark Toolbars in Firefox Filevo is a Cool File Hosting & Sharing Site Get a free copy of WinUtilities Pro 2010 World Cup Schedule Boot Snooze – Reboot and then Standby or Hibernate

  Read the article

• Converting a video file in arbitrary file format into MPEG4/H.264?

  - by knorv

  I want to convert a large number of video files in various formats into .mp4 files (container MPEG-4, codec H.264). I want to do this on an Ubuntu machine, using only command-line tools and I'm willing to install packages from main, restricted, universe and multiverse. Ideally I'd like to be able to do ... for VIDEO_FILE in *; do some_conversion_program $VIDEO_FILE $VIDEO_FILE.mp4 done ... and have all my video files in .mp4 format with container MPEG-4 and codec H.264. How would you tackle this problem on an Ubuntu machine? What packages do I need to install?

  Read the article

• Google funds OGG Theora

  <b>Lnux Pro Magazine:</b> "Grant goes toward development of ARM-optimized video codec"

  Read the article

• How do i tell ubuntu to only install is asked for and do not bring other dependencies which will break the whole system?

  - by YumYumYum

  How can i only install python-webkit but not other packages? which is showing to install? (no gstreamer*.*, i do not want to have any single files installed in my distro because of GPL license and it slows my machine a lot) $ sudo apt-get install libwebkitgtk-1.0-0 python-webkit Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libgstreamer-plugins-base0.10-0 libgstreamer0.10-0 Suggested packages: gstreamer-codec-install gnome-codec-install gstreamer0.10-tools gstreamer0.10-plugins-base The following NEW packages will be installed: libgstreamer-plugins-base0.10-0 libgstreamer0.10-0 libwebkitgtk-1.0-0 0 upgraded, 3 newly installed, 0 to remove and 333 not upgraded. Need to get 8,231 kB of archives. After this operation, 28.2 MB of additional disk space will be used. Do you want to continue [Y/n]? n Abort.

  Read the article

• How to play .mp4 movies?

  - by Andrew

  I have a .mp4 movie (it's an .mp4 file, and the "video codec" is Xvid, and the "audio codec" is mp3) that I want to play, but when I try to open it in MPlayer, it says "The steam is encrypted and decryption is not supported." I tried to apply the answer from this question and this wiki page, but neither worked. Any suggestions? The file came from an [apparently] less than reputable site, which I guess I can't link to. :( I've tried: sudo apt-get install ubuntu-restricted-extras sudo apt-get install libdvdread4 sudo /usr/share/doc/libdvdread4/install-css.sh sudo apt-get install gstreamer0.10-ffmpeg sudo apt-get install w64codecs sudo apt-get update

  Read the article

• How to automatically change volume level when un-/plugging headphones?

  - by htorque

  What I want is the following: When I plug in my headphones, I want the sound to be un-muted and set to a specific volume level. When I unplug my headphones, I want the sound to be muted (or set to a specific volume level). Setting the volume levels isn't the problem, but I somehow need to do this when un-/plugging the headphones, so I'm looking for a way to get notified of those events. I quickly found /proc/asound/card0/codec#0 to indicate whether headphones are plugged in or not, so I tried to monitor it using inotifywait and change the volume level based on modified notifications. Unfortunately inotifywait failed because proc isn't an ordinary filesystem. Are there other ways to do this (maybe via PulseAudio)? Audio device: Intel HDA, audio codec: Conexant CX20585.

  Read the article

• Converting mp4 to mp3

  - by aki

  I have a video I need to convert to mp3 (from the command line - not GUI) video.mp4 I tried: ffmpeg -i -b 192 video.mp4 video.mp3 with no success. I get the following error: WARNING: library configuration mismatch Seems stream 0 codec frame rate differs from container frame rate: 59.83 (29917/500) -> 59.75 (239/4) WARNING: The bitrate parameter is set too low. It takes bits/s as argument, not kbits/s Encoder (codec id 86017) not found for output stream #0.0 so I tried lame: lame -h -b 192 video.mp4 video.mp3 I get: Warning: unsupported audio format Am I missing something?

  Read the article

• Chrome ne supportera plus le H.264, Google ne veut soutenir que « des technologies et des codecs complètement ouverts »

  Chrome ne supportera plus le H.264 Google ne veut soutenir que « des technologies et des codecs complètement ouverts » Le billet de Google s'intitule « HTML Video Codec Support in Chrome ». Mais il aurait aussi bien pu s'appeler « pourquoi nous abandonnons le H.264 ». Chrome prend donc le même chemin que Firefox et ne supportera plus ? en natif tout du moins ? le codec vidéo soutenu par Apple. Motif invoqué : le H.264 est fermé et propriétaire. « Nous supportons WebM (VP8) et Theora, et nous envisageons d'ajouter à l'avenir le support d'autres codecs ouvertes et de qualité », écrit Mike Jazayeri, Product Manager chez Google. « Bien que...

  Read the article

• Streaming a webcam from Silverlight 4 (Beta)

  - by Ken Smith

  The new webcam stuff in Silverlight 4 is darned cool. By exposing it as a brush, it allows scenarios that are way beyond anything that Flash has. At the same time, accessing the webcam locally seems like it's only half the story. Nobody buys a webcam so they can take pictures of themselves and make funny faces out of them. They buy a webcam because they want other people to see the resulting video stream, i.e., they want to stream that video out to the Internet, a lay Skype or any of the dozens of other video chat sites/applications. And so far, I haven't figured out how to do that with It turns out that it's pretty simple to get a hold of the raw (Format32bppArgb formatted) bytestream, as demonstrated here. But unless we want to transmit that raw bytestream to a server (which would chew up way too much bandwidth), we need to encode that in some fashion. And that's more complicated. MS has implemented several codecs in Silverlight, but so far as I can tell, they're all focused on decoding a video stream, not encoding it in the first place. And that's apart from the fact that I can't figure out how to get direct access to, say, the H.264 codec in the first place. There are a ton of open-source codecs (for instance, in the ffmpeg project here), but they're all written in C, and they don't look easy to port to C#. Unless translating 10000+ lines of code that look like this is your idea of fun :-) const int b_xy= h->mb2b_xy[left_xy[i]] + 3; const int b8_xy= h->mb2b8_xy[left_xy[i]] + 1; *(uint32_t*)h->mv_cache[list][cache_idx ]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[0+i*2]]; *(uint32_t*)h->mv_cache[list][cache_idx+8]= *(uint32_t*)s->current_picture.motion_val[list][b_xy + h->b_stride*left_block[1+i*2]]; h->ref_cache[list][cache_idx ]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[0+i*2]>>1)]; h->ref_cache[list][cache_idx+8]= s->current_picture.ref_index[list][b8_xy + h->b8_stride*(left_block[1+i*2]>>1)]; The mooncodecs folder within the Mono project (here) has several audio codecs in C# (ADPCM and Ogg Vorbis), and one video codec (Dirac), but they all seem to implement just the decode portion of their respective formats, as do the java implementations from which they were ported. I found a C# codec for Ogg Theora (csTheora, http://www.wreckedgames.com/forum/index.php?topic=1053.0), but again, it's decode only, as is the jheora codec on which it's based. Of course, it would presumably be easier to port a codec from Java than from C or C++, but the only java video codecs that I found were decode-only (such as jheora, or jirac). So I'm kinda back at square one. It looks like our options for hooking up a webcam (or microphone) through Silverlight to the Internet are: (1) Wait for Microsoft to provide some guidance on this; (2) Spend the brain cycles porting one of the C or C++ codecs over to Silverlight-compatible C#; (3) Send the raw, uncompressed bytestream up to a server (or perhaps compressed slightly with something like zlib), and then encode it server-side; or (4) Wait for someone smarter than me to figure this out and provide a solution. Does anybody else have any better guidance? Have I missed something that's just blindingly obvious to everyone else? (For instance, does Silverlight 4 somewhere have some classes I've missed that take care of this?)

  Read the article

• Encode audio to aac with libavcodec

  - by ryan

  I'm using libavcodec (latest git as of 3/3/10) to encode raw pcm to aac (libfaac support enabled). I do this by calling avcodec_encode_audio repeatedly with codec_context-frame_size samples each time. The first four calls return successfully, but the fifth call never returns. When I use gdb to break, the stack is corrupt. If I use audacity to export the pcm data to a .wav file, then I can use command-line ffmpeg to convert to aac without any issues, so I'm sure it's something I'm doing wrong. I've written a small test program that duplicates my problem. It reads the test data from a file, which is available here: http://birdie.protoven.com/audio.pcm (~2 seconds of signed 16 bit LE pcm) I can make it all work if I use FAAC directly, but the code would be a little cleaner if I could just use libavcodec, as I'm also encoding video, and writing both to an mp4. ffmpeg version info: FFmpeg version git-c280040, Copyright (c) 2000-2010 the FFmpeg developers built on Mar 3 2010 15:40:46 with gcc 4.4.1 configuration: --enable-libfaac --enable-gpl --enable-nonfree --enable-version3 --enable-postproc --enable-pthreads --enable-debug=3 --enable-shared libavutil 50.10. 0 / 50.10. 0 libavcodec 52.55. 0 / 52.55. 0 libavformat 52.54. 0 / 52.54. 0 libavdevice 52. 2. 0 / 52. 2. 0 libswscale 0.10. 0 / 0.10. 0 libpostproc 51. 2. 0 / 51. 2. 0 Is there something I'm not setting, or setting incorrectly in my codec context, maybe? Any help is greatly appreciated! Here is my test code: #include <stdio.h> #include <libavcodec/avcodec.h> void EncodeTest(int sampleRate, int channels, int audioBitrate, uint8_t *audioData, size_t audioSize) { AVCodecContext *audioCodec; AVCodec *codec; uint8_t *buf; int bufSize, frameBytes; avcodec_register_all(); //Set up audio encoder codec = avcodec_find_encoder(CODEC_ID_AAC); if (codec == NULL) return; audioCodec = avcodec_alloc_context(); audioCodec->bit_rate = audioBitrate; audioCodec->sample_fmt = SAMPLE_FMT_S16; audioCodec->sample_rate = sampleRate; audioCodec->channels = channels; audioCodec->profile = FF_PROFILE_AAC_MAIN; audioCodec->time_base = (AVRational){1, sampleRate}; audioCodec->codec_type = CODEC_TYPE_AUDIO; if (avcodec_open(audioCodec, codec) < 0) return; bufSize = FF_MIN_BUFFER_SIZE * 10; buf = (uint8_t *)malloc(bufSize); if (buf == NULL) return; frameBytes = audioCodec->frame_size * audioCodec->channels * 2; while (audioSize >= frameBytes) { int packetSize; packetSize = avcodec_encode_audio(audioCodec, buf, bufSize, (short *)audioData); printf("encoder returned %d bytes of data\n", packetSize); audioData += frameBytes; audioSize -= frameBytes; } } int main() { FILE *stream = fopen("audio.pcm", "rb"); size_t size; uint8_t *buf; if (stream == NULL) { printf("Unable to open file\n"); return 1; } fseek(stream, 0, SEEK_END); size = ftell(stream); fseek(stream, 0, SEEK_SET); buf = (uint8_t *)malloc(size); fread(buf, sizeof(uint8_t), size, stream); fclose(stream); EncodeTest(32000, 2, 448000, buf, size); }

  Read the article

• HLS video segmenting complications. How to create a transport stream with ffmpeg

  - by Agzam

  I have h264 videos, and currently we're using Apple's HTTP Video Streaming tools and mediafilesegmenter to segment these files. What I need to do is to switch to alternative segmenter based on this very popular open-sourced segmenter The problem is that this segmenter does not just take any video, but takes only MPEG-TS videos. So I have to convert my h264 videos to TS first. I can do that with ffmpeg. I'm using this: ffmpeg -i encoded.mp4 -vcodec h264 -i encoded.mp4 -sameq -acodec aac -strict experimental -f mpegts output.ts But this creates fairly larger output. And the reason is that Apple's segmenter keeps the same codec - AVC and the same audio codec - AAC, whereas ffmpeg changes video format to MPEG Video. The question is: can I somehow keep the same AVC video codec and still convert video to a transport stream? So my goal is to keep the same video quality and same video codecs as Apple's medifilesegmenter does. UPD: Okay... it seems that ffmpeg CAN split videos into segments: ffmpeg -i encoded.mp4 -c copy -map 0 -vbsf h264_mp4toannexb -f segment -segment_time 10 -segment_list test.m3u8 -segment_format mpegts segment%d.ts That's still has one problem: it doesn't create http live streaming index file. (-segment_list creates a file with list of segments, but it doesn't look like HLS index). So, you still have to create index file

  Read the article

• Decoding not reversing unicode encoding in Django/Python

  - by PhilGo20

  Ok, I have a hardcoded string I declare like this name = u"Par Catégorie" I have a # -- coding: utf-8 -- magic header, so I am guessing it's converted to utf-8 Down the road it's outputted to xml through xml_output.toprettyxml(indent='....', encoding='utf-8') And I get a UnicodeDecodeError: 'ascii' codec can't decode byte 0xc3 in position 3: ordinal not in range(128) Most of my data is in French and is ouputted correctly in CDATA nodes, but that one harcoded string keep ... I don't see why an ascii codec is called. what's wrong ?

  Read the article

< Previous Page | 4 5 6 7 8 9 10 11 12 13 14 15  | Next Page >