Search Results

Search found 277 results on 12 pages for 'stereo'.

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

Page 7/12 | < Previous Page | 3 4 5 6 7 8 9 10 11 12  | Next Page >

  • Obtaining projection matrix

    - by rita
    I have calibrated stereo images. I want to get a projection matrix using rotation matrix ,translation vector, radial vector and a camera matrix. How can get the projection matrix?

    Read the article

  • How to read the metadata of a MKV movie files in C#?

    - by Ole Jak
    How to read the metadata of a MKV movie files in C# (or C or C++ or AS3)? for example such creator metadatacreator hasKeyframes hasVideo hasAudio hasMetaData canSeekToEnd duration datasize videosize videocodecid audiosize audiocodecid audiosamplerate audiosamplesize stereo filesize lasttimestamp lastkeyframetimestamp lastkeyframelocation keyframes (filepositions, times) width height framerate videodatarate audiodatarate

    Read the article

  • How to Join N live MP3 streams into one using FFMPEG?

    - by Ole Jak
    How to Join N live MP3 streams (radio streams like such live KCDX mp3 stream http://mp3.kcdx.com:8000/stream ) into 1 using FFMPEG? (I have N incoming live mp3 streams I want to join them and stream out 1 live mp3 stream) I mean I want to mix sounds like thay N speakers speak at the same time (btw N stereo to 1 mono), please help. BTW: My problem is mainly how to make FFMPEG read from stream not from file... Would you mind giving some code examples, please.

    Read the article

  • How to Join N live MP3 streams into 1 using FFMPEG?

    - by Ole Jak
    How to Join N live MP3 streams (radio streams like such live KCDX mp3 stream http://mp3.kcdx.com:8000/stream ) into 1 using FFMPEG? (I have N incoming live mp3 streams I vant to join them and stream out 1 live mp3 stream) I mean I wanna to mix sounds like thay N speakers speak at the same time (btw N stereo to 1 mono), please help BTW: My problem is mainly how to make FFMPEG read from stream not from file... Would you mind giving some code examples, please...

    Read the article

  • Video Reconstruction

    - by chris barber
    How does reconstruction using video compare to using standard reconstruction using still images? What similarities and differences are there. Finally what can and cannot be reconstructed using standard stereo methods?

    Read the article

  • FFmpeg extract clip - stream frame rate differs from container frame rate (x264, aac)

    - by fideli
    Summary H.264 video seems to have a really high frame rate that requires a scaling factor to the applied to the duration of video that I'm trying to extract (900x lower). Body I'm trying to extract a clip from a movie that I have in MP4 format (created using Handbrake). After trying mencoder and VLC, I decided to give FFmpeg a shot since it was the least troublesome when it came to copying the codecs. That is, compared to mencoder and VLC, the resulting file was still playable in QuickTime (I know about Perian, etc, I'm just trying to learn how all this works). Anyway, my command was as follows: ffmpeg -ss 01:15:51 -t 00:05:59 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 During the copy, The following comes up: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from outofsight.mp4': Duration: 01:57:42.10, start: 0.000000, bitrate: 830 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x384, 25 tbr, 22500 tbn, 45k tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16 Output #0, mp4, to 'out.mp4': Stream #0.0(und): Video: libx264, yuv420p, 720x384, q=2-31, 90k tbn, 22500 tbc Stream #0.1(eng): Audio: libfaac, 48000 Hz, stereo, s16 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 2591 fps=2349 q=-1.0 size= 8144kB time=101.60 bitrate= 656.7kbits/s … Instead of a 5:59 duration clip, I get the entire rest of the movie. So, to test this, I ran the ffmpeg command with -t 00:00:01. What I got was exactly a 15:00 minute clip. So I did some black box engineering and decided to scale my -t option by calculating what value to enter given that 1 second was interpreted as 900 s. For my desired 359 s clip, I calculated 0.399 s and so my ffmpeg command became: ffmpeg -ss 01:15.51 -t 00:00:00.399 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 This works, but I have no idea why the duration is scaled by 900. Investigating further, each ffmpeg run has the line: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) 45000/25 = 1800. Must be a relation somewhere. Somehow, the obscenely high frame rate is causing issues with the timing. How is that frame rate so high? The best part about this is that the resulting clip.mp4 has the exact same feature (due to the copied video codec), and taking further clips from this needs the same scaling for the -t duration option. Therefore, I've made it available for anyone willing to check this out. Appendix The preamble for ffmpeg on my system (built using MacPorts ffmpeg port): FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/opt/local --disable-vhook --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libfaad --enable-libxvid --enable-libx264 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 1. 4. 0 / 1. 4. 0 libswscale 1. 7. 1 / 1. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jan 4 2010 21:51:51, gcc: 4.2.1 (Apple Inc. build 5646) (dot 1)

    Read the article

  • FFmpeg extract clip - stream frame rate differs from container frame rate (x264, aac)

    - by fideli
    Summary H.264 video seems to have a really high frame rate that requires a scaling factor to the applied to the duration of video that I'm trying to extract (900x lower). Body I'm trying to extract a clip from a movie that I have in MP4 format (created using Handbrake). After trying mencoder and VLC, I decided to give FFmpeg a shot since it was the least troublesome when it came to copying the codecs. That is, compared to mencoder and VLC, the resulting file was still playable in QuickTime (I know about Perian, etc, I'm just trying to learn how all this works). Anyway, my command was as follows: ffmpeg -ss 01:15:51 -t 00:05:59 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 During the copy, The following comes up: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from outofsight.mp4': Duration: 01:57:42.10, start: 0.000000, bitrate: 830 kb/s Stream #0.0(und): Video: h264, yuv420p, 720x384, 25 tbr, 22500 tbn, 45k tbc Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16 Output #0, mp4, to 'out.mp4': Stream #0.0(und): Video: libx264, yuv420p, 720x384, q=2-31, 90k tbn, 22500 tbc Stream #0.1(eng): Audio: libfaac, 48000 Hz, stereo, s16 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop encoding frame= 2591 fps=2349 q=-1.0 size= 8144kB time=101.60 bitrate= 656.7kbits/s … Instead of a 5:59 duration clip, I get the entire rest of the movie. So, to test this, I ran the ffmpeg command with -t 00:00:01. What I got was exactly a 15:00 minute clip. So I did some black box engineering and decided to scale my -t option by calculating what value to enter given that 1 second was interpreted as 900 s. For my desired 359 s clip, I calculated 0.399 s and so my ffmpeg command became: ffmpeg -ss 01:15.51 -t 00:00:00.399 -i outofsight.mp4 \ -acodec copy -vcodec copy clip.mp4 This works, but I have no idea why the duration is scaled by 900. Investigating further, each ffmpeg run has the line: Seems stream 0 codec frame rate differs from container frame rate: 45000.00 (45000/1) -> 25.00 (25/1) 45000/25 = 1800. Must be a relation somewhere. Somehow, the obscenely high frame rate is causing issues with the timing. How is that frame rate so high? The best part about this is that the resulting clip.mp4 has the exact same feature (due to the copied video codec), and taking further clips from this needs the same scaling for the -t duration option. Therefore, I've made it available for anyone willing to check this out. Appendix The preamble for ffmpeg on my system (built using MacPorts ffmpeg port): FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --prefix=/opt/local --disable-vhook --enable-gpl --enable-postproc --enable-swscale --enable-avfilter --enable-avfilter-lavf --enable-libmp3lame --enable-libvorbis --enable-libtheora --enable-libdirac --enable-libschroedinger --enable-libfaac --enable-libfaad --enable-libxvid --enable-libx264 --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/gcc-4.2 --arch=x86_64 libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 libavfilter 1. 4. 0 / 1. 4. 0 libswscale 1. 7. 1 / 1. 7. 1 libpostproc 51. 2. 0 / 51. 2. 0 built on Jan 4 2010 21:51:51, gcc: 4.2.1 (Apple Inc. build 5646) (dot 1) EDIT Not sure whether it was a bug or not, but it seems to be fixed now in my current version of ffmpeg, at least for this video (version 0.6.1 from MacPorts).

    Read the article

  • Crash in audio resampler with some audio rates - FFMPEG PHP ( Solved! )

    - by Olaf Erlandsen
    i have a problem with this command( FFMPEG PHP ): Command: ffmpeg -i 62f76f050494f0ed6a5997967c00c0c0.wmv -ss 0 -t 99 -y -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 -f flv 62f76f050494f0ed6a5997967c00c0c0.flv Output: FFmpeg version 0.6.5, Copyright (c) 2000-2010 the FFmpeg developers built on Jan 29 2012 17:52:15 with gcc 4.4.5 20110214 (Red Hat 4.4.5-6) configuration: --prefix=/usr --libdir=/usr/lib64 --shlibdir=/usr/lib64 --mandir=/usr/share/man --incdir=/usr/include --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic -fPIC' --enable-avfilter --enable-avfilter-lavf --enable-libdc1394 --enable-libdirac --enable-libfaac --enable-libfaad --enable-libfaadbin --enable-libgsm --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libx264 --enable-gpl --enable-nonfree --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-version3 --enable-x11grab libavutil 50.15. 1 / 50.15. 1 libavcodec 52.72. 2 / 52.72. 2 libavformat 52.64. 2 / 52.64. 2 libavdevice 52. 2. 0 / 52. 2. 0 libavfilter 1.19. 0 / 1.19. 0 libswscale 0.11. 0 / 0.11. 0 libpostproc 51. 2. 0 / 51. 2. 0 [asf @ 0xe81670]max_analyze_duration reached Input #0, asf, from '/var/www/resources/tmp/62f76f050494f0ed6a5997967c00c0c0.wmv': Metadata: WMFSDKVersion : 12.0.7601.17514 WMFSDKNeeded : 0.0.0.0000 IsVBR : 0 Duration: 00:00:50.87, bitrate: 2467 kb/s Stream #0.0: Audio: wmapro, 44100 Hz, stereo, flt, 256 kb/s Stream #0.1: Video: vc1, yuv420p, 950x460 [PAR 1:1 DAR 95:46], 25 fps, 25 tbr, 1k tbn, 25 tbc Output #0, flv, to '/var/www/resources/media/62f76f050494f0ed6a5997967c00c0c0.flv': Metadata: encoder : Lavf52.64.2 Stream #0.0: Video: flv, yuv420p, 950x460 [PAR 1:1 DAR 95:46], q=2-31, 200 kb/s, 1k tbn, 29.97 tbc Stream #0.1: Audio: libmp3lame, 11025 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 -> #0.0 Stream #0.0 -> #0.1 Press [q] to stop encoding frame= 72 fps= 0 q=5.0 size= 0kB time=10.91 bitrate= 0.0kbits/s Multiple frames in a packet from stream 0 Warning, using s16 intermediate sample format for resampling frame= 141 fps=139 q=5.0 size= 103kB time=8.15 bitrate= 103.2kbits/s frame= 220 fps=144 q=5.0 size= 875kB time=10.92 bitrate= 656.6kbits/s frame= 290 fps=143 q=5.0 size= 1525kB time=13.74 bitrate= 909.1kbits/s frame= 356 fps=141 q=5.0 size= 2153kB time=15.99 bitrate=1103.1kbits/s frame= 427 fps=141 q=5.0 size= 2847kB time=18.70 bitrate=1247.0kbits/s frame= 497 fps=141 q=5.0 size= 3771kB time=21.16 bitrate=1460.0kbits/s frame= 575 fps=142 q=5.0 size= 4695kB time=24.61 bitrate=1563.0kbits/s frame= 639 fps=141 q=5.0 size= 5301kB time=26.80 bitrate=1620.2kbits/s frame= 703 fps=139 q=5.0 size= 5829kB time=29.36 bitrate=1626.2kbits/s frame= 774 fps=139 q=5.0 size= 6659kB time=32.39 bitrate=1684.0kbits/s frame= 842 fps=139 q=5.0 size= 7915kB time=35.27 bitrate=1838.6kbits/s frame= 911 fps=139 q=5.0 size= 9011kB time=37.98 bitrate=1943.4kbits/s frame= 975 fps=138 q=5.0 size= 9788kB time=40.59 bitrate=1975.3kbits/s frame= 1041 fps=138 q=5.0 size= 10904kB time=43.83 bitrate=2037.9kbits/s frame= 1115 fps=138 q=5.0 size= 11795kB time=46.24 bitrate=2089.8kbits/s frame= 1183 fps=138 q=5.0 size= 12678kB time=48.74 bitrate=2130.7kbits/s frame= 1247 fps=137 q=5.0 size= 13964kB time=51.36 bitrate=2227.5kbits/s frame= 1271 fps=136 q=5.0 Lsize= 15865kB time=58.86 bitrate=2208.1kbits/s video:15366kB audio:462kB global headers:0kB muxing overhead 0.238956% Problem: Warning, using s16 intermediate sample format for resampling I've also tried changing the parameter From -ar 44100 to -ar 11025 Thanks! Solution: Read this link: http://en.wikipedia.org/wiki/MP3#Bit_rate

    Read the article

  • what is the best mid/high-end class audio/music creation audio sound card?

    - by Chris
    Hello, I have a computershop myself, and I repair computers. But one of the things I really don't know (yet) is the performace od audio cards for music creation with midi. I have searched and searched and came up with some good reviews, but after browsing for a couple of hours I could't see the trees trough the forrest :-D (it's a dutch expression) At one moment I thought the M-Audio - Delta 1010LT would be a good PCIe card, later on I read that this card was released years ago. (but that could be false information) Also any personal expierence would be great, but not necessairy. I have searched a few cards, and I hope someone can help me make a choice for a friend of mine. He's buget is between $100 and $350 I know there are audio cards from $ 500 - $1850,- this is just too expensive. The following specs are crucial: ASIO Midi Mic in minimal 5.1, 7.1 recommended it's not for airplay, but just to compose music at home. using Ableton and midi keyboard. 1. M-Audio - Delta 1010LT: 8 x 8 analog I/O 2 mic preamps or line inputs S/PDIF digital I/O (coaxial) with 2-channel PCM SCMS copy protection control digital I/O supports surround-encoded AC-3 and DTS pass-through 1 x 1 MIDI I/O directly drive up to 7.1 surround (bass management software included) software controlled 36-bit internal DSP digital mixing/routing +4dbu/-10dBV operation individually switched in software word clock I/O for sample accurate device synchronization 2. RME HDSP 9632: * Stereo Analog Ein- und Ausgang, symmetrisch*, 24-Bit/192kHz, > 110 dB SNR * Optionale Erweiterungsboards mit je 4 symmetrischen Ein- und Ausgängen * Alle analogen I/Os voll 192 kHz-fähig, also keine Reduzierung der Kanalzahl * 1 x ADAT Digital In/Out, 96 kHz-fähig (S/MUX) * 1 x SPDIF Digital In/Out, 192 kHz-fähig * 1 x Breakout Kabel für koaxialen SPDIF-Betrieb* * Also bis zu 16 Ein-und Ausgänge gleichzeitig nutzbar! * 1 x Stereo Kopfhörerausgang, parallel zum analogen Ausgang, aber eigene Pegelanpassung * 1 x MIDI I/O für 16 Kanäle Hi-Speed MIDI über Breakout Kabel * DIGICheck, RMEs einzigartiges Meter- und Analysetool mit Spectral Analyser, Professionelle Level Meter 2/8/16-Kanalig, Vector Audio Scope und diversen weiteren Analysefunktionen * HDSP Meter Bridge: Frei skalierbare Levelmeter mit Peak- und RMS Berechnung in Hardware * TotalMix: 512-Kanal Mischer mit 40 Bit interner Auflösung 3. EMU 1212M (1212 M) PCIe: * Top kwaliteit convertors 24-bit/192kHz convertors. * Hardware gestuurde effecten. * DSP zero-latency hardware mixen en monitoring. * Analoge en digitale I/O plus MIDI. * EMU Production Tools Software Bundle - Cakewalk SONAR , Steinberg Cubase LE, Ableton Live E-MU Edition **EMU 1212M PCI-e inputs/outputs:** * 2 balanced jack inputs. * 2 balanced jack outputs. * 24-bit/192kHz ADAT I/O. * 24-bit/192kHz Coaxiale S/PDif I/O switchable to AES/EBU. * MIDI I/O. 4. M-Audio Audiophile 192: - Up to 24-bit/192kHz audio - 2 balanced analog inputs (1/4” TRS) - 2 balanced analog outputs (1/4” TRS) - S/PDIF digital I/O (coaxial RCA connectors) with 2-channel PCM - SCMS copy protection control - Digital I/O supports surround-encoded AC-3 and DTS pass-through - Direct hardware input monitoring via separate balanced 1/4” TRS monitor outputs - Software routing of inputs and outputs - Digital I/O can be routed to/from external effects - 16-channel MIDI I/O - ASIO, WDM, GSIF 2 and Core Audio driver support for compatibility with most applications - 64-bit driver support for Windows - PCI 2.2 compatibility - Apple G5 compatible - Incompatible exceptions - Includes Ableton Live Lite music production software, so you can make music right away - Works with other Delta cards Technical Specifcations: - Compatibility - ASIO - WDM - GSIF 2 - Core Audio

    Read the article

  • Make exact mp4 (H264) format for uploading to youtube

    - by WHITECOLOR
    With ffmpeg I'm converting video from mp3 and picture to upload it to youtube. After upload, conversion fails. Reasons are unknown. I believe the problem is in format. By the way If I'm uploading file 5 minutes length, it fails if I upload 30 seconds of this file it succeeds. I have donwload mp4 file from youtube. Then I uploaded it, it is done very fast. So a nice solution would be to convert videos to the same format that is done by google. I got the following output by mpeg: ffmpeg version N-44264-g070b0e1 Copyright (c) 2000-2012 the FFmpeg developers built on Sep 7 2012 17:38:57 with gcc 4.7.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-pthreads --enable-runt ime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass - -enable-libcelt --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-l ibfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenj peg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheo ra --enable-libutvideo --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-li bvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --ena ble-zlib libavutil 51. 72.100 / 51. 72.100 libavcodec 54. 55.100 / 54. 55.100 libavformat 54. 25.105 / 54. 25.105 libavdevice 54. 2.100 / 54. 2.100 libavfilter 3. 16.100 / 3. 16.100 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 15.100 / 0. 15.100 libpostproc 52. 0.100 / 52. 0.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'youtubetrack0.mp4': Metadata: major_brand : mp42 minor_version : 0 compatible_brands: isommp42 creation_time : 2012-10-02 22:58:57 Duration: 00:06:46.66, start: 0.000000, bitrate: 176 kb/s Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yu v420p, 450x360, 78 kb/s, 6 fps, 6 tbr, 12 tbn, 12 tbc Metadata: creation_time : 1970-01-01 00:00:00 handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 95 kb/s Metadata: creation_time : 2012-10-02 22:58:57 handler_name : IsoMedia File Produced by Google, 5-11-2011 Is it possible to construct ffmpeg parameters so that that would give the same format that google internally does? Is the information above sufficient? I couldn't construct needed params. For example I don't understand how to set tbn and what 95 kb/s mean in "Stream #0:1(und): Audio:". Now I just do: ffmpeg -i videoimage.jpg -i audio.mp3 video.mp4 Info I've got: ffmpeg version N-44998-gdf82454 Copyright (c) 2000-2012 the FFmpeg developers built on Oct 2 2012 23:03:12 with gcc 4.7.1 (GCC) configuration: --disable-static --enable-shared --enable-gpl --enable-version3 --disable-pthreads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libass --enable-libcelt --enable-libopencore-amrnb --en able-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libnut --enable-libopenjpeg --enable-librtmp --enable-libschroedinger - -enable-libspeex --enable-libtheora --enable-libutvideo --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enab le-libxavs --enable-libxvid --enable-zlib libavutil 51. 73.101 / 51. 73.101 libavcodec 54. 63.100 / 54. 63.100 libavformat 54. 29.105 / 54. 29.105 libavdevice 54. 3.100 / 54. 3.100 libavfilter 3. 19.102 / 3. 19.102 libswscale 2. 1.101 / 2. 1.101 libswresample 0. 16.100 / 0. 16.100 libpostproc 52. 1.100 / 52. 1.100 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.25.105 Duration: 00:06:46.81, start: 0.000000, bitrate: 129 kb/s Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuvj420p, 450x360, 3392 kb/s, 25 fps, 25 tbr, 25 tbn, 50 tbc Metadata: handler_name : VideoHandler Stream #0:1(und): Audio: aac (mp4a / 0x6134706D), 44100 Hz, stereo, s16, 127 kb/s Metadata: handler_name : SoundHandler This video fails the conversion on youtube. I also tried to use other vcode parmam and extensions of output file (mp4, wmv, avi) but failed too. Would be greatful for help.

    Read the article

  • Vlc And other Multi Media Problems

    - by MrMagu
    ive tried numerous Free Multimedia Programs on ubuntu and i get the same result when i go to watch one of my movies off my other hardrive it allways ends up full screening with a stuck image in the middle and vlc ends up looking pitch black. ive look all over the web for this issue im wondering if anyone is experinceing the same problems I Dont know if it has to do with the duel monitors ive tried adding and re adding the repositorys but it seems to be doing the same repetive thing for over a month now do i need to reinstall the whole system or what idk anyhelp please would be much appreciated Ty MrMagu Xorg Conf File nvidia-settings: X configuration file generated by nvidia-settings nvidia-settings: version 304.37 (buildd@allspice) Sun Sep 9 05:59:26 UTC 2012 Section "ServerLayout" Identifier "Layout0" Screen 0 "Screen0" 0 0 InputDevice "Keyboard0" "CoreKeyboard" InputDevice "Mouse0" "CorePointer" Option "Xinerama" "0" EndSection Section "Files" EndSection Section "InputDevice" # generated from default Identifier "Mouse0" Driver "mouse" Option "Protocol" "auto" Option "Device" "/dev/psaux" Option "Emulate3Buttons" "no" Option "ZAxisMapping" "4 5" EndSection Section "InputDevice" # generated from default Identifier "Keyboard0" Driver "kbd" EndSection Section "Monitor" # HorizSync source: edid, VertRefresh source: edid Identifier "Monitor0" VendorName "Unknown" ModelName "Ancor Communications Inc VE247" HorizSync 30.0 - 83.0 VertRefresh 50.0 - 76.0 Option "DPMS" EndSection Section "Monitor" Identifier "Monitor1" VendorName "Unknown" ModelName "Ancor Communications Inc VE247" HorizSync 30.0 - 83.0 VertRefresh 50.0 - 76.0 EndSection Section "Device" Identifier "Device0" Driver "nvidia" VendorName "NVIDIA Corporation" BoardName "Quadro FX 1500" EndSection Section "Device" Identifier "Device1" Driver "nvidia" VendorName "NVIDIA Corporation" BoardName "Quadro FX 1500" BusID "PCI:1:0:0" Screen 1 EndSection Section "Screen" Removed Option "metamodes" "DFP-0: nvidia-auto-select +0+0, DFP-1: nvidia-auto-select +1920+0; DFP-0: 1920x1080 +0+0, DFP-1: 1920x1080 +1920+0" Removed Option "metamodes" "DFP-0: nvidia-auto-select +0+0; DFP-0: 1920x1080 +0+0; DFP-0: 1920x1080_60 +0+0; DFP-0: 1680x1050 +0+0; DFP-0: 1680x1050_60 +0+0; DFP-0: 1440x900 +0+0; DFP-0: 1440x900_60 +0+0; DFP-0: 1280x1024 +0+0; DFP-0: 1280x1024_75 +0+0; DFP-0: 1280x1024_60 +0+0; DFP-0: 1280x960 +0+0; DFP-0: 1280x960_60 +0+0; DFP-0: 1152x864 +0+0; DFP-0: 1152x864_75 +0+0; DFP-0: 1024x768 +0+0; DFP-0: 1024x768_75 +0+0; DFP-0: 1024x768_70 +0+0; DFP-0: 1024x768_60 +0+0; DFP-0: 800x600 +0+0; DFP-0: 800x600_75 +0+0; DFP-0: 800x600_72 +0+0; DFP-0: 800x600_60 +0+0; DFP-0: 800x600_56 +0+0; DFP-0: 640x480 +0+0; DFP-0: 640x480_75 +0+0; DFP-0: 640x480_72 +0+0; DFP-0: 640x480_60 +0+0; DFP-0: nvidia-auto-select @1920x720 +0+0" Removed Option "metamodes" "DFP-0: nvidia-auto-select +0+0; DFP-0: 1920x1080 +0+0; DFP-0: 1920x1080_60 +0+0; DFP-0: 1680x1050 +0+0; DFP-0: 1680x1050_60 +0+0; DFP-0: 1440x900 +0+0; DFP-0: 1440x900_60 +0+0; DFP-0: 1280x1024 +0+0; DFP-0: 1280x1024_75 +0+0; DFP-0: 1280x1024_60 +0+0; DFP-0: 1280x960 +0+0; DFP-0: 1280x960_60 +0+0; DFP-0: 1152x864 +0+0; DFP-0: 1152x864_75 +0+0; DFP-0: 1024x768 +0+0; DFP-0: 1024x768_75 +0+0; DFP-0: 1024x768_70 +0+0; DFP-0: 1024x768_60 +0+0; DFP-0: 800x600 +0+0; DFP-0: 800x600_75 +0+0; DFP-0: 800x600_72 +0+0; DFP-0: 800x600_60 +0+0; DFP-0: 800x600_56 +0+0; DFP-0: 640x480 +0+0; DFP-0: 640x480_75 +0+0; DFP-0: 640x480_72 +0+0; DFP-0: 640x480_60 +0+0" Identifier "Screen0" Device "Device0" Monitor "Monitor0" DefaultDepth 24 Option "TwinView" "1" Option "TwinViewXineramaInfoOrder" "DFP-0" Option "Stereo" "0" Option "nvidiaXineramaInfoOrder" "DFP-0" Option "metamodes" "DFP-0: nvidia-auto-select +0+0, DFP-1: 1920x1080 +1920+0; DFP-0: 1920x1080 +0+0, DFP-1: nvidia-auto-select +1920+0; DFP-0: 1920x1080_60 +0+0; DFP-0: 1680x1050 +0+0; DFP-0: 1680x1050_60 +0+0; DFP-0: 1440x900 +0+0; DFP-0: 1440x900_60 +0+0; DFP-0: 1280x1024 +0+0; DFP-0: 1280x1024_75 +0+0; DFP-0: 1280x1024_60 +0+0; DFP-0: 1280x960 +0+0; DFP-0: 1280x960_60 +0+0; DFP-0: 1152x864 +0+0; DFP-0: 1152x864_75 +0+0; DFP-0: 1024x768 +0+0; DFP-0: 1024x768_75 +0+0; DFP-0: 1024x768_70 +0+0; DFP-0: 1024x768_60 +0+0; DFP-0: 800x600 +0+0; DFP-0: 800x600_75 +0+0; DFP-0: 800x600_72 +0+0; DFP-0: 800x600_60 +0+0; DFP-0: 800x600_56 +0+0; DFP-0: 640x480 +0+0; DFP-0: 640x480_75 +0+0; DFP-0: 640x480_72 +0+0; DFP-0: 640x480_60 +0+0" SubSection "Display" Depth 24 EndSubSection EndSection Section "Screen" Identifier "Screen1" Device "Device1" Monitor "Monitor1" DefaultDepth 24 Option "Stereo" "0" Option "metamodes" "DFP-1: 1920x1080 +0+0" SubSection "Display" Depth 24 EndSubSection EndSection Section "InputClass" Identifier "Mouse Remap" MatchProduct "Saitek Cyborg R.A.T.7 Mouse" MatchDevicePath "/dev/input/event*" Option "ButtonMapping" "1 2 3 4 5 6 7 8 9 0 0 0 0 0 0" EndSection Pictures http://tinypic.com/r/34o7j1h/6 http://tinypic.com/r/3rn20/6 Ty so Much Mr Magu

    Read the article

  • How to stop intermittent playback on airtunes (iTunes + Airport Express)?

    - by Alex B
    Is there a definitive answer for how to stop intermittent "skips" or pauses while playing music from iTunes to an airport express connected to my home stereo? When I read other forums I see a wealth of posts that say "I did XYZ and it's fixed" followed by "I tried XYZ and it didn't work." This does not appear to be signal strength related. The green light on the Airport Express does not turn to yellow/orange. Other wireless devices have no trouble connecting at the same or greater distances from the wireless router.

    Read the article

  • Strange. Asterisk key plays random Windows sound when pressed

    - by Charles
    This is a new one on me. When I press the "asterisk", or * button on my number pad (but not SHIFT+8), Windows makes either a "Exclamation" or "Windows ding" sound. I haven't noticed a pattern to which sound is made. Logitech K200 keyboard No special key mapping software or Logitech software running Realtek sound to stereo through optical cable. Visual Studio 2010, Chrome, Fiddler, WinRAR, Notepad++, and Dropbox running. No unusual behavior otherwise. A solution isn't terribly important but my curiosity is both piqued and stumped. This doesn't normally happen and nothing odd has taken place otherwise. Ideas?

    Read the article

  • No sound (USB speakers) in Ubuntu 9.04 Jaunty Jackalope

    - by Mike
    I just installed Ubuntu 9.04 and have been unable to get sound functioning. I have a set of USB speakers (not USB powered with a stereo plug... they are totally USB). I've tried plugging them directly into the tower as well as into the USB port on my Dell 2405FPW monitor. Both USB ports are functioning correctly (I tested by sticking a flash drive in there - they both read it), and the speakers are functioning correctly in Windows. If it's relevant, I have an SB Audigy 2 sound card that came with the computer, but is not being used. Any ideas? Thanks! EDIT: These are the speakers - Logitech S-150 USB Speakers

    Read the article

  • Live noise-filter on line-in

    - by Damon Gant
    I'm running the following setup: Xbox 360 is hooked up to my (PC) screen via HDMI/DVI converter. Because the Xbox has no dedicated sound output, except for optical S/PIDF, I'm also using the AV/RCA output, namely just the audio, which is connected to an old stereo, which is then connected to my PCs line-in. I'm now experiencing a some of noise. I'm using one of the standard "Realtek High Definition Audio" cards, which doesn't seem to offer this kind of functionality. Is there a software that will playback audio right off a device while running filters on it? It doesn't have to create a device on its own, I just want to listen to it. Here's a sample: http://puu.sh/1suY6

    Read the article

  • How do I use wine's new multi-channel audio support?

    - by Forest
    Wine supposedly added Multi-channel support to the ALSA driver in a recent release. However, I haven't found any documentation about it or any reports of anyone successfully using it. I have 5.1 surround sound through ALSA (not Pulse) on my linux system, yet games in wine don't seem to use anything more than stereo, no matter which ALSA device I select in winecfg. Can someone explain what must be done to get wine games to use all my speakers for positional audio, or failing that, point me toward some useful documentation on the new multi-channel support?

    Read the article

  • Realtek HD Audio 5.1 Optical Input (from Xbox 360)

    - by Shevek
    I'm trying to connect up my Xbox 360 digital audio to my 5.1 speakers via my PC (my LCD has dual input, DVI for the PC, D-SUB for the Xbox). The motherboard has a Realtek ALC888 chipset and I have a 5.1 speaker system connected via 3 x 3.5mm jacks (FR/FL, RR/RL, C/LFE) and I get full 5.1 output from the PC. I have connected the optical audio cable from the Xbox to the Optical In on the motherboard's backplate. With the Xbox in Digital Stereo mode I get 2 channel audio from the Xbox, through the PC, to the speakers. With the Xbox in Dolby Digital 5.1 mode I get no sound at all. I have the latest Realtek drivers installed in Win 7 32-bit. Questions: Is it possible to use the full 5.1 DD from the Xbox? If so, am I missing some option(s) in the Realtek setup? Do I need some other piece of software to do this? (AC3Filter or FFDShow perhaps) Many thanks

    Read the article

  • FFmpeg audio dont work in converted videos

    - by Juddy Swaft
    NOTICE: when i convert videos via terminal and download them from ftp into pc the audio works fine. I use: if($ext == "avi" && $convert_avi == true) { $convert_source = _VIDEOS_DIR_PATH.$new_name; $conv_name = substr(md5($file['name'].rand(1,888)), 2, 10).".mp4"; $converted_file = _VIDEOS_DIR_PATH.$conv_name; $ffmpeg_command = 'ffmpeg -i '.$convert_source.' -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 '.$converted_file; echo exec($ffmpeg_command); $sql = "UPDATE pm_temp SET url = '".$conv_name."' WHERE url = '".$new_name."' LIMIT 1"; $result = @mysql_query($sql); unlink($convert_source); } This code to convert avi to mp4 ffmpeg concole output: root@1tb:~# ffmpeg -i sample.avi -acodec libmp3lame -vcodec libx264 -s 1280x720 -ar 44100 -async 44100 -r 29.970 -ac 2 -qscale 5 goodsample.mp4 ffmpeg version 0.7.15, Copyright (c) 2000-2013 the FFmpeg developers built on Feb 22 2013 07:18:58 with gcc 4.4.5 configuration: --enable-libdc1394 --prefix=/usr --extra-cflags='-Wall -g ' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-avfilter --enable-libdirac --disable-decoder=libdirac --enable-libfreetype --enable-libschroedinger --disable-encoder=libschroedinger - s libavutil 50. 43. 0 / 50. 43. 0 libavcodec 52.123. 0 / 52.123. 0 libavformat 52.111. 0 / 52.111. 0 libavdevice 52. 5. 0 / 52. 5. 0 libavfilter 1. 80. 0 / 1. 80. 0 libswscale 0. 14. 1 / 0. 14. 1 libpostproc 51. 2. 0 / 51. 2. 0 [mp3 @ 0x191d4100] Header missing [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected Input #0, avi, from 'sample.avi': Metadata: encoder : VirtualDubMod 1.5.10.2 (build 2540/release) Duration: 00:01:01.81, start: 0.000000, bitrate: 1194 kb/s Stream #0.0: Video: mpeg4, yuv420p, 640x352 [PAR 1:1 DAR 20:11], 23.98 tbr, Stream #0.1: Audio: mp3, 48000 Hz, stereo, s16, 128 kb/s [buffer @ 0x191d1c80] w:640 h:352 pixfmt:yuv420p tb:1/1000000 sar:1/1 sws_param: [scale @ 0x191d6880] w:640 h:352 fmt:yuv420p -> w:1280 h:720 fmt:yuv420p flags:0 [libx264 @ 0x191ce5a0] Default settings detected, using medium profile [libx264 @ 0x191ce5a0] using SAR=45/44 [libx264 @ 0x191ce5a0] using cpu capabilities: MMX2 SSE2Fast SSSE3 FastShuffle S [libx264 @ 0x191ce5a0] profile High, level 3.1 [libx264 @ 0x191ce5a0] 264 - core 118 - H.264/MPEG-4 AVC codec - Copyleft 2003-2 6 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_off 1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_l Output #0, mp4, to 'goodsample.mp4': Metadata: encoder : Lavf52.111.0 Stream #0.0: Video: libx264, yuv420p, 1280x720 [PAR 45:44 DAR 20:11], q=2-31 Stream #0.1: Audio: libmp3lame, 44100 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help [mp3 @ 0x191d4100] Header missing Error while decoding stream #0.1 [mpeg4 @ 0x191d1dc0] Invalid and inefficient vfw-avi packed B frames detected [mp3 @ 0x191d4100] incomplete frame 9467kB time=00:01:00.32 bitrate=1285.5kbits/ Error while decoding stream #0.1 frame= 1852 fps= 20 q=29.0 Lsize= 9652kB time=00:01:01.72 bitrate=1280.9kbits video:9121kB audio:483kB global headers:0kB muxing overhead 0.499688% frame I:11 Avg QP:16.78 size: 51456 [libx264 @ 0x191ce5a0] frame P:784 Avg QP:20.81 size: 8954 [libx264 @ 0x191ce5a0] frame B:1057 Avg QP:26.06 size: 1659 [libx264 @ 0x191ce5a0] consecutive B-frames: 22.0% 3.1% 7.5% 67.4% [libx264 @ 0x191ce5a0] mb I I16..4: 31.1% 59.8% 9.1% [libx264 @ 0x191ce5a0] mb P I16..4: 1.8% 2.6% 0.2% P16..4: 24.3% 7.0% 4.0 [libx264 @ 0x191ce5a0] mb B I16..4: 0.1% 0.1% 0.0% B16..8: 22.7% 0.8% 0.2 [libx264 @ 0x191ce5a0] 8x8 transform intra:57.0% inter:72.6% [libx264 @ 0x191ce5a0] coded y,uvDC,uvAC intra: 44.4% 33.3% 10.3% inter: 7.6% 5. [libx264 @ 0x191ce5a0] i16 v,h,dc,p: 68% 14% 8% 10% [libx264 @ 0x191ce5a0] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 14% 27% 5% 7% 7% 6 [libx264 @ 0x191ce5a0] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 28% 14% 14% 6% 10% 9% 7 [libx264 @ 0x191ce5a0] i8c dc,h,v,p: 67% 13% 17% 3% [libx264 @ 0x191ce5a0] Weighted P-Frames: Y:1.9% UV:0.4% [libx264 @ 0x191ce5a0] ref P L0: 62.2% 12.8% 10.3% 14.5% 0.2% [libx264 @ 0x191ce5a0] ref B L0: 88.1% 5.5% 6.4% [libx264 @ 0x191ce5a0] ref B L1: 95.7% 4.3% [libx264 @ 0x191ce5a0] kb/s:1209.03 I know there is couple errors tough, but i dont know hot to fix it. Also i would be very thankfull if someone can help reduce video size but is not main problem video weights as original avi but sill.

    Read the article

  • Enabling Surround sound on a Realtek ALC892 via SPDIF, on Windows 7

    - by Alex
    I have a problem with my ALC892, on an ASRock mainboard (ASRock 890FX Deluxe4). I get only stereo sounds if I use SPDIF connection, in general. My amp shows that is getting surround sound only when I use the Test feature of Windows 7. This test feature allows to know which formats are supported by the audio chip. The tests render correctly both Dolby Digital and DTS. You can find this test under Sounds, Playback Devices, Select Digital Audio, then "Properties". I am using Windows 7 x64, with the latest drivers from the official Realtek website. I also tested other driver versions, both from the Realtek website and from the ASRock one, but had no luck. Thanks for the help. Some specs: CPU: AMD Phenom II X4 965 MOBO: ASRock 890FX Deluxe4 (with onboard Realtek ALC892) Audio amp: Onkyo R-380 (works fine with other sources like PS3 and Xbox 360)

    Read the article

  • Use all 5.1 speakers with a 2.1 audio source

    - by thegreyspot
    Hi! I just bought a 5.1 surround sound speaker set for my computer in my bedroom. The rear speakers are next to me in bed while the front speakers are at the other end of the bed at my feet. While I enjoy the surround sound during movies that support 5.1 sound, I would like to have my rear speakers working when listening to podcasts, or other 2.1 channel sound. How can I do this? When I enable "Speaker Fill" in the Realtek Hd Audio manager the sound only comes out of the front and center speakers with a few background noises that come out the rear ones. But since my ears are closer to the rear speakers, I'd rather have the sound come out of them. Let me know of any ideas! Hmm seems like the only option is to set the rear speakers to "Front Speakers" and change it to stereo in the Realtek HD audio. But still that take alot of steps and it doesnt not use the center speaker Thanks

    Read the article

  • 3.5mm headphone jack -> Component video?

    - by Isaac Waller
    Hello, I have a strange cord that has a male 3.5mm headphone jack on one side, and some component video and audio ouputs on the other. When I plug the headphone side into my computer, and the other side into my TV, I get audio across the cable, but no video. How can I send some video over the headphone jack in OS X? I know you can, because this cable is actually for a Beyblade toy and it somehow transferred video over this cable. Thanks, Isaac I think I found some more information (from Wikipedia:) "A four conductor version (of a 3.5mm headphone jack) is becoming a de facto standard output connector for compact camcorders, providing stereo sound plus a video signal." So how can I do this on Mac?

    Read the article

  • How to recover broken MJPEG AVI file after a camera crash

    - by Jam K
    I was recording a video when my KODAK EASYSHARE M531 Digital Camera fell to the ground and turned off. When i turned it on nothing happened to the camera but the video file is somehow broken. Obviously the camera did not have time to convert and close it properly. $file myvid.avi myvid.AVI: data where it should be: another_vid.AVI: RIFF (little-endian) data, AVI, 640 x 480, ~30 fps, video: Motion JPEG, audio: uLaw (stereo, 12000 Hz)` I would like to fix my video which I believe needs to be converted to AVI. Any idea? VLC does not play the file and returns: could not determine type of stream

    Read the article

  • Convert audio file to FLAC with ffmpeg?

    - by elpsk
    can I convert one of this format to compatible 16000.0 Sample Rate FLAC file? kAudioFormatLinearPCM = 'lpcm', kAudioFormatAppleIMA4 = 'ima4', kAudioFormatMPEG4AAC = 'aac ', kAudioFormatMACE3 = 'MAC3', kAudioFormatMACE6 = 'MAC6', kAudioFormatULaw = 'ulaw', kAudioFormatALaw = 'alaw', kAudioFormatMPEGLayer1 = '.mp1', kAudioFormatMPEGLayer2 = '.mp2', kAudioFormatMPEGLayer3 = '.mp3', kAudioFormatAppleLossless = 'alac' I tried using ffmpeg ffmpeg -i audio.xxx -acodec flac audio.flac but result is FFmpeg version CVS, Copyright (c) 2000-2004 Fabrice Bellard Mac OSX universal build for ffmpegX configuration: --enable-memalign-hack --enable-mp3lame --enable-gpl --disable-vhook --disable-ffplay --disable-ffserver --enable-a52 --enable-xvid --enable-faac --enable-faad --enable-amr_nb --enable-amr_wb --enable-pthreads --enable-x264 libavutil version: 49.0.0 libavcodec version: 51.9.0 libavformat version: 50.4.0 built on Apr 15 2006 04:58:19, gcc: 4.0.1 (Apple Computer, Inc. build 5250) Input #0, wsaud, from 'audio.alac': Duration: 00:00:03.8, start: 0.000000, bitrate: 199 kb/s Stream #0.0: Audio: adpcm_ima_ws, 24931 Hz, stereo, 199 kb/s Unable for find a suitable output format for 'audio.flac' I also installed flac codec for mac, but nothing... I tried also use convtoflac.sh (from http://legroom.net/software/convtoflac) but result is similar. Any idea to convert in flac?

    Read the article

facebook twitter digg google delicious technorati stumbleupon myspace wordpress linkedin gmail igoogle windows live tumblr viadeo yahoo buzz yahoo mail yahoo bookmarks favorites email print

< Previous Page | 3 4 5 6 7 8 9 10 11 12  | Next Page >