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  • Optimal video resolution and encoding for recording games for YouTube?

    - by Rookie
    I want to record video from games, therefore I cannot use very large video resolution, but I still want to make the large video view to look as sharp as the original encoded video before upload. I tried to use YouTube's recommended 854x640 resolution, but it wasn't possible with h264 and the encoding software I used (Handbrake) converted it to a width of the nearest multiple of 4, which I think is a limitation of the h264 format. The video I encoded was sharp and fine quality, but when I uploaded it to YouTube, it lost a lot of quality and the preferred large video view looks almost as bad as a 320p video. I tried to wait a few days but it never got sharper (in case it didn't process it completely yet). So, which resolution and encoding options I should use, if I want the large video player to have the sharpest possible video, retaining the original video quality as good as possible? I noticed that recording with 640x480, the video was sharper than with 1280x720, so I'm not sure what im doing wrong here; both were h264. Is it anyhow possible to prevent YouTube from re-encoding the videos? I just wonder how people can make so sharp videos, while mine are all blurry after upload, but before upload they looked fine. I also tried YouTube's suggested bitrates with h264, but it didn't work any better.

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  • Converting from mp4 to Xvid avi using avconv?

    - by Ricardo Gladwell
    I normally use avidemux to convert mp4s to Xvid AVI for my Philips Streamium SLM5500. Normally I select MPEG-4 ASP (Xvid) at Two Pass with an average bitrate f 1500kb/s for video and AC3 (lav) audio and it converts correctly. However, I'm trying to using avconv so I can automate the process with a script, but when I do this the video stutters and stops playing part way through. I have a suspicion its something to do with a faulty audio conversion. The commands I'm using are as follows: avconv -y -i video.mp4 -pass 1 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi /dev/null avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi There is a bewildering array of arguments for avconv. Is there something I'm doing wrong? Is there a way I can script avidemux from a headless server? Please see command line output: $ avconv -y -i video.mp4 -pass 1 -vtag xvid -an -b:v 1500k -f avi /dev/null avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x7f4c40] w:720 h:404 pixfmt:yuv420p Output #0, avi, to '/dev/null': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 1, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Press ctrl-c to stop encoding frame=66227 fps=328 q=2.0 Lsize= 0kB time=2649.16 bitrate= 0.0kbits/s video:401602kB audio:0kB global headers:0kB muxing overhead -100.000000% $ avconv -y -i video.mp4 -pass 2 -vtag xvid -c:a ac3 -b:a 128k -b:v 1500k -f avi video.avi avconv version 0.8.5-6:0.8.5-0ubuntu0.12.10.1, Copyright (c) 2000-2012 the Libav developers built on Jan 24 2013 14:49:20 with gcc 4.7.2 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'video.mp4': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 Duration: 00:44:09.16, start: 0.000000, bitrate: 669 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 720x404 [PAR 1:1 DAR 180:101], 538 kb/s, 25 fps, 25 tbr, 100 tbn, 50 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2013-02-04 13:53:42 [buffer @ 0x12b4f00] w:720 h:404 pixfmt:yuv420p Incompatible sample format 's16' for codec 'ac3', auto-selecting format 'flt' [mpeg4 @ 0x12b3ec0] [lavc rc] Using all of requested bitrate is not necessary for this video with these parameters. Output #0, avi, to 'video.avi': Metadata: major_brand : isom minor_version : 1 compatible_brands: isomavc1 creation_time : 2013-02-04 13:53:38 ISFT : Lavf53.21.1 Stream #0.0(und): Video: mpeg4, yuv420p, 720x404 [PAR 1:1 DAR 180:101], q=2-31, pass 2, 1500 kb/s, 25 tbn, 25 tbc Metadata: creation_time : 2013-02-04 13:53:38 Stream #0.1(und): Audio: ac3, 44100 Hz, stereo, flt, 128 kb/s Metadata: creation_time : 2013-02-04 13:53:42 Stream mapping: Stream #0:0 -> #0:0 (h264 -> mpeg4) Stream #0:1 -> #0:1 (ac3 -> ac3) Press ctrl-c to stop encoding Input stream #0:1 frame changed from rate:44100 fmt:s16 ch:2 to rate:44100 fmt:flt ch:2 frame=66227 fps=284 q=2.2 Lsize= 458486kB time=2649.13 bitrate=1417.8kbits/s video:413716kB audio:41393kB global headers:0kB muxing overhead 0.741969%

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  • 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

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  • VLC desktop streaming

    - by StackedCrooked
    Edit I stopped using VLC and switched to GMax FLV Encoder. It does a much better job IMO. Original post I am sending my desktop (screen) as an H264 video stream to another machine that saves it to a file using the follwoing command lines: Sender of the stream: vlc -I dummy --sout='#transcode{vcodec=h264,vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' Receiver of the stream: vlc -I rc rtp://@:4444 --sout='#std{access=file,mux=ps,dst=/home/user/output.mp4}' --ipv4 This works, but there are a few issues: The file is not playable with most players. VLC is able to playback the file but with some weirdness: = it takes about 10 seconds before the playback actually begins. = seeking doesn't work. Can someone point me in the right direction on how to fix these issues? EDIT: I made a little progress. The initial delay in playback is because the player is waiting for a keyframe. By forcing the sender of the stream to create a new key-frame every 4 seconds I could decrease the delay: :screen-fps=10 --sout='#transcode{vcodec=h264,venc=x264{keyint=40},vb=512,scale=0.5} :rtp{mux=ts,dst=192.168.0.1,port=4444}' The seeking problem is not solved however, but I understand it a little better. The RTP stream is saved as a file in its original streaming format, which is normally not playable as a regular video file. VLC manages to play this file, but most other players don't. So I need to convert it to a regular video file. I am currently investigating whether I can do this with ffmpeg if I provide it with an SDP file for the recorded stream. All help is welcome!

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  • Split section of video with ffmpeg

    - by Rob
    I've been trying to get this to work and I'm really close, but something still isn't right. I have a 14 second clip I'm trying to cut out of a longer mp4 video. I got the video to cut to the right place with this command: ffmpeg -ss 00:05:13.0 -i ~/videos/trim_me.mp4 -vcodec h264 -acodec copy -t 00:00:14.0 ~/videos/trimmed.mp4 If I didn't specify -vcodec it was starting from an "I-Frame" (I guess) and wasn't the right place. The audio is starting from that spot as well, so I tried setting -acodec the same way: ffmpeg -ss 00:05:13.0 -i ~/videos/trim_me.mp4 -vcodec h264 -acodec aac -ac 2 -ab 225k -ar 48000 -strict -2 -t 00:00:14.0 ~/videos/trimmed.mp4 Which doesn't really help much. Setting -async 1 makes it take longer, and then the audio does match up, but not until 4 seconds into the video. :/ I'd ideally not like to install anything else and have a commandline solution for this.

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  • ffmpeg open webcam using YUYV but i want MJPEG

    - by Pavel
    I need ffmpeg to open webcam (logitech c910) in MJPEG mode, because the webcam can give ~24 using MJPEG "protocol" and only ~10 fps using the YUYV. Can i choose between them using ffmpeg command line? xx@(none) ~ $ v4l2-ctl --list-formats ioctl: VIDIOC_ENUM_FMT Index : 0 Type : Video Capture Pixel Format: 'YUYV' Name : YUV 4:2:2 (YUYV) Index : 1 Type : Video Capture Pixel Format: 'MJPG' (compressed) Name : MJPEG My current command line: ffmpeg -y -f alsa -i hw:3,0 -f video4linux2 -r 20 -s 1280x720 -i /dev/video0 -acodec libfaac -ab 128k -vcodec libx264 /tmp/web.avi ffmpeg produces corrupted h264 stream when i record from webcam, but normal h264 strem when i record from x11grab. Another codecs (mjpeg, mpeg4) works well with webcam... But this is another story.

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  • 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)

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  • How to unit test with lots of IO

    - by Eric
    I write Linux embedded software which closely integrates with hardware. My modules are such as : -CMOS video input with kernel driver (v4l2) -Hardware h264/mpeg4 encoders (texas instuments) -Audio Capture/Playback (alsa) -Network IO I'd like to have automated testing for those functionalities, such as integration testing. I am not sure how I can automate this process since most of the top level functionalities I face are IO bound. Sure, it is easy to test functions individually, but whole process checking means depending on tons of external dependencies only available at runtime.

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  • How can I maximum compress video files?

    - by EmmyS
    I received 4 .mov files from a client that they want on their mobile website via SlideShowPro. Each original file was between 200 and 400 mb. I've gotten each one down to about 30 mb using transmageddon as described here, but that's still really big for a mobile connection. Is there any way to shrink them even further? Maybe it's the settings; I used Output Format = MPEG4, Audio = AAC, Video = H264 (which is what is suggested by SlideShowPro.)

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  • Convert mp4 video to a format xbox 360 can play

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

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  • FFMPEG: how to add watermark to video?

    - by DocWiki
    My Platform: Ubuntu 10.10 + FFMPEG 0.5.3(I installed ffmpeg from source) I try to add Watermark to a .MOV video with FFMPEG 0.5.3 imlib2.so (Please note FFMPEG 0.6+ dont support imlib2.so, so I use ffmpeg 0.5.3) Here is my code: ffmpeg -sameq -i example.mov -vhook '/usr/local/lib/vhook/imlib2.so -x 0 -y 0 -i /var/www/files/watermark.png' newexample.mov Here is the output: FFmpeg version 0.5.3, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-avfilter --enable-filter=movie --enable-avfilter-lavf 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 built on Jul 3 2011 12:05:08, gcc: 4.4.5 Seems stream 1 codec frame rate differs from container frame rate: 59.94 (5994/100) - 29.97 (30000/1001) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'example.mov': Duration: 00:03:14.06, start: 0.000000, bitrate: 3350 kb/s Stream #0.0(eng): Audio: aac, 48000 Hz, stereo, s16 Stream #0.1(eng): Video: h264, yuv420p, 1150x647, 29.97 tbr, 29.97 tbn, 59.94 tbc Output #0, mov, to 'newexample.mov': Stream #0.0(eng): Video: mpeg4, yuv420p, 1150x647, q=2-31, 200 kb/s, 90k tbn, 29.97 tbc Stream #0.1(eng): Audio: 0x0000, 48000 Hz, stereo, s16, 64 kb/s Stream mapping: Stream #0.1 - #0.0 Stream #0.0 - #0.1 Unsupported codec for output stream #0.1 What could be the possible problem? Is that AAC or H264 that is not supported? I installed libavcodec-extra-52, linfaac, libfaad and etc. but the error is the same. Do I have to install following this instruction? HOWTO: Install and use the latest FFmpeg and x264 or there is a simpler solution?

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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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  • Nginx flv audio pseudo stream works but video is not loading

    - by sarah
    I am working on a development server for a company & they want nginx webserver to work with. So the requirements for their company is, it should be capable of doing following things i.e hotlink protection, mp4 & flv pseudo stream & secure streaming. However nginx fulfills their requirements and i am configuring their server from past 2 days as i am new to this field so i've only acheived hotlinking prevention in past 2 days. But the problem on which i am stuck is flv pseudo streaming, to make work to mp4 pseudo stream it was just a piece of paper but i am really fuc*ed up with flv pseudo stream. I have converted my flv videos with flvmdi tools to insert many keyframes but the problem is , when i try to seek video from following keyframes that are generated by flvmdi i.e test.flv?start=2681223, video does not load but audio pseudo works fine. So it means no problem with my flv configuration in nginx.conf file. And the forum that i used to compile my nginx-1.2.1 is http://h264.code-shop.com/trac/wiki/Mod-H264-Streaming-Nginx-Version2 & by adding additional module --with-http_flv_module. This forum is really active, hopes i will resolve my problem as soon as you guys will provide me some guide.

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  • How to get bearable 2D and 3D performance on AMD Radeon HD 6950?

    - by l0b0
    I have had an AMD Radeon HD 6950 (i.e., Cayman series) for a couple years now, and I have tried a lot of combinations of drivers and settings with terrible results. I'm completely at a loss as to how to proceed. The open source driver has much better 2D performance, but it offloads all OpenGL rendering to the CPU. What I've tried so far: All the latest stable Ubuntu releases in the period, plus one Linux Mint release. All the latest stable AMD Catalyst Proprietary Display Drivers, and currently 13.1. The unofficial wiki installation instructions for every Ubuntu version and the semi-official Ubuntu instructions. All the tips and tweaks I could find for Minecraft (Optifine, reducing settings to minimum), VLC (postprocessing at minimum, rendering at native video size), Catalyst Control Center (flipped every lever in there) and X11 (some binary toggles I can no longer remember). Results: Typically 13-15 FPS in Minecraft, 30 max (100+ in Windows with the same driver version). Around 10 FPS in Team Fortress 2 using the official Steam client. Choppy video playback, in Flash and with VLC. CPU use goes through the roof when rendering video (150% for 1080p on YouTube in Chromium, 100% for 1080p H264 in VLC). glxgears shows 12.5 FPS when maximized. fgl_glxgears shows 10 FPS when maximized. Hardware details from lshw: Motherboard ASUS P6X58D-E CPU Intel Core i7 CPU 950 @ 3.07GHz (never overclocked; 64 bit) 6 GB RAM Video card product "Cayman PRO [Radeon HD 6950]", vendor "Hynix Semiconductor (Hyundai Electronics)" 2 x 1920x1200 monitors, both connected with HDMI. I feel I must be missing something absolutely fundamental here. Is there no accelerated support for anything on 64-bit architectures? Does a dual monitor completely mess up the driver? $ fglrxinfo display: :0 screen: 0 OpenGL vendor string: Advanced Micro Devices, Inc. OpenGL renderer string: AMD Radeon HD 6900 Series OpenGL version string: 4.2.11995 Compatibility Profile Context $ glxinfo | grep 'direct rendering' direct rendering: Yes I am currently using the open source driver, with the following results: Full frame rate and low CPU load when playing 1080p video. Black screen (but music in the background) in Team Fortress 2. Similar performance in Minecraft as the Catalyst driver. In hindsight obvious, since both end up offloading the rendering to the CPU. My /var/log/Xorg.0.log after upgrading to AMD Catalyst 13.1. Some possibly important lines: (WW) Falling back to old probe method for fglrx (WW) fglrx: No matching Device section for instance (BusID PCI:0@3:0:1) found The generated xorg.conf. The disabled "monitor" 0-DFP9 is actually an A/V receiver, which sometimes confuses the monitor drivers when turned on/off (but not in Windows). All three "monitor" devices are connected with HDMI. Edit: Chris Carter's suggestion to use the xorg-edgers PPA (Catalyst 13.1) resulted in some improvement, but still pretty bad performance overall: Minecraft stabilizes at 13-17 FPS, but at least the CPU load is "only" at 45-60%. Still 150% CPU use for 1080p video rendering on YouTube in Chromium. Massive improvement for 1080p H264 in VLC: 40-50% CPU use and no visible jitter glxgears performance about doubled to 25-30 FPS when maximized. fgl_glxgears still at ~10 FPS when maximized.

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  • libav/ffmpeg: avcodec_decode_video2() returns -1 when separating demultiplexing and decoding

    - by unbekannt
    I'm using libav (from a C++ program on Linux and Windows) to decode video streams from a file, which works fine (decoding various formats like H264 and MPEG2) using avformat_open_input(), av_read_frame() and avcodec_decode_video2(). Now I have to separate demultiplexing and decoding. One class will call avformat_open_input() and av_read_frame() and then pass the AVPackets into a queue that is read by another class. There I use avcodec_alloc_context3() to get the AVCodecContext needed for avcodec_decode_video2(). I've tested that with a MPEG2 video stream and it works. Problems arise if I try to decode a H264 stream: avcodec_decode_video2() always returns -1 and outputs "no frame". I understand that additional data (SPS/PPS) is needed to decode this stream, so I've tried to replicate the original AVCodecContext from the demultiplexer in the decoder, but it won't work: Copying the content of the extradata field and setting all other values that differ from the default ones in the decoder: -1 is returned Using the same context (i.e. passing along the pointer) results in a crash I also tried to set CODEC_FLAG2_CHUNKS. avcodec_decode_video2() then always returns packet.size - 3 (??) and frameFinished is never set to 1. In my opinion I have a general problem here that will arise whenever settings from the original CodecContext are needed to decode the AVPackets. I'd be grateful for any hints on how to solve that problem!

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  • Use DivX settings to encode to mp4 with ffmpeg

    - by sjngm
    I'm used to use VirtualDub to encode a video to AVI container with DivX-codec (and MP3 for audio). Now I'm planning to use ffmpeg to encode videos to MP4 container with h264-codec. What I've figured out is that I need to use libx264 and one of those presets to make anything work. However, I'm amazed about the video bitrate ffmpeg uses for encoding. What I currently have is this little batch file: @ECHO OFF SETLOCAL SET IN=source.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-fpre "%FFMPEG_PATH%\presets\libx264-lossless_slow.ffpreset" SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %VIDEO% %PRESET% test.mp4 ENDLOCAL With this I tell ffmpeg to use 1978k as the bitrate, but ffmpeg uses 15000k+! I tried other presets, but they don't use my specified bitrate. Here are the presets I have: libx264-baseline.ffpreset libx264-ipod320.ffpreset libx264-ipod640.ffpreset libx264-lossless_fast.ffpreset libx264-lossless_max.ffpreset libx264-lossless_medium.ffpreset libx264-lossless_slow.ffpreset libx264-lossless_slower.ffpreset libx264-lossless_ultrafast.ffpreset ffmpeg version: FFmpeg git-N-29181-ga304071 libavutil 50. 40. 1 / 50. 40. 1 libavcodec 52.120. 0 / 52.120. 0 libavformat 52.108. 0 / 52.108. 0 libavdevice 52. 4. 0 / 52. 4. 0 libavfilter 1. 79. 0 / 1. 79. 0 libswscale 0. 13. 0 / 0. 13. 0 Note that I don't use the latest version as it has problems with spaces in filenames. Here's what seems to be the full parameter list DivX 6.9.2 uses: -bvnn 1978000 -vbv 218691200,100663296,100663296 -dir "C:\Users\sjngm\AppData\Roaming\DivX\DivX Codec" -w -b 1 -use_presets=1 -preset=10 -windowed_fullsearch=2 -thread_delay=1 What command line parameters would that be for ffmpeg? EDIT: Going with slhck's suggestion I tried a new 32-bit version. I have no idea if that is 0.9 or newer, I can't find that info. ffmpeg version N-36890-g67f5650 libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 I reworked my batch file to look like this (interestingly enough I can't find parameter -vprofile in the documentation): @ECHO OFF SETLOCAL SET IN=VTS_01_1.avs SET FFMPEG_PATH=C:\Program Files (x86)\ffmpeg SET PRESET=-vprofile high -preset veryslow SET AUDIO=-acodec libmp3lame -ab 128000 SET VIDEO=-vcodec libx264 -vb 1978000 "%FFMPEG_PATH%\ffmpeg.exe" -i %IN% %AUDIO% %PRESET% %VIDEO% test.mp4 ENDLOCAL I see that it now uses the bitrate properly (thanks to LongNeckbeard for pointing out that the lossless-stuff ignores the bitrate!). Just in case you wonder how I came up with the 1978000, I'm using this formula which I found valid for DivX-files (I'm guessing the bitrate won't change that much for h264): width * height * 25 * 0.22 / 1000 I'm not sure if the 0.22 correlates with the CRF somehow. Overall I forgot to say the I will use a two-pass scenario, which is why I don't use the CRF here. I will try to read more about this. Currently I'm just trying to get something running that shows me that I'm doing something right (ffmpeg isn't the easiest tool to understand ;)). C:\Program Files (x86)\ffmpeg\ffmpeg.exe" -i VTS_01_1.avs -acodec libmp3lame -ab 128000 -vcodec libx264 -vb 1978000 -vprofile high -preset veryslow test.mp4 The output is now: ffmpeg version N-36890-g67f5650 Copyright (c) 2000-2012 the FFmpeg developers built on Jan 16 2012 21:57:13 with gcc 4.6.2 configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-runtime-cpudetect --enable-avisynth --enable-bzlib --enable-frei0r --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 51. 34.100 / 51. 34.100 libavcodec 53. 56.105 / 53. 56.105 libavformat 53. 30.100 / 53. 30.100 libavdevice 53. 4.100 / 53. 4.100 libavfilter 2. 59.100 / 2. 59.100 libswscale 2. 1.100 / 2. 1.100 libswresample 0. 6.100 / 0. 6.100 libpostproc 51. 2.100 / 51. 2.100 Input #0, avs, from 'VTS_01_1.avs': Duration: 00:58:46.12, start: 0.000000, bitrate: 0 kb/s Stream #0:0: Video: rawvideo (YV12 / 0x32315659), yuv420p, 576x448, 77414 kb/s, 25 tbr, 25 tbn, 25 tbc Stream #0:1: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 48000 Hz, 2 channels, s16, 1536 kb/s File 'test.mp4' already exists. Overwrite ? [y/N] y w:576 h:448 pixfmt:yuv420p tb:1/1000000 sar:0/1 sws_param: [libx264 @ 05A2C400] using cpu capabilities: MMX2 SSE2Fast FastShuffle SSEMisalign LZCNT [libx264 @ 05A2C400] profile High, level 3.1 [libx264 @ 05A2C400] 264 - core 120 r2120 0c7dab9 - H.264/MPEG-4 AVC codec - Copyleft 2003-2011 - http://www.videolan.org/x264.html - options: cabac=1 ref=16 deblock=1:0:0 analyse=0x3:0x133 me=umh subme=10 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=24 chroma_me=1 trellis=2 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=3 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=8 b_pyramid=2 b_adapt=2 b_bias=0 direct=3 weightb=1 open_gop=0 weightp=2 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=60 rc=abr mbtree=1 bitrate=1978 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 ip_ratio=1.40 aq=1:1.00 Output #0, mp4, to 'test.mp4': Metadata: encoder : Lavf53.30.100 Stream #0:0: Video: h264 (![0][0][0] / 0x0021), yuv420p, 576x448, q=-1--1, 1978 kb/s, 25 tbn, 25 tbc Stream #0:1: Audio: mp3 (i[0][0][0] / 0x0069), 48000 Hz, 2 channels, s16, 128 kb/s Stream mapping: Stream #0:0 -> #0:0 (rawvideo -> libx264) Stream #0:1 -> #0:1 (pcm_s16le -> libmp3lame) Press [q] to stop, [?] for help frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 0 fps= 0 q=0.0 size= 0kB time=00:00:00.00 bitrate= 0.0kbits/s frame= 3 fps= 1 q=22.0 size= 39kB time=00:00:00.04 bitrate=8063.8kbits/ frame= 8 fps= 2 q=22.0 size= 82kB time=00:00:00.24 bitrate=2801.3kbits/ frame= 13 fps= 3 q=23.0 size= 120kB time=00:00:00.44 bitrate=2229.5kbits/ frame= 16 fps= 4 q=23.0 size= 147kB time=00:00:00.56 bitrate=2156.7kbits/ frame= 20 fps= 4 q=22.0 size= 175kB time=00:00:00.72 bitrate=1987.4kbits/ : video:4387kB audio:273kB global headers:0kB muxing overhead 0.260038% [libx264 @ 05A2C400] frame I:2 Avg QP:19.53 size: 29850 [libx264 @ 05A2C400] frame P:76 Avg QP:22.24 size: 19541 [libx264 @ 05A2C400] frame B:359 Avg QP:25.93 size: 8210 [libx264 @ 05A2C400] consecutive B-frames: 0.5% 0.5% 0.0% 8.2% 17.2% 52.2% 16.0% 5.5% 0.0% [libx264 @ 05A2C400] mb I I16..4: 5.4% 75.3% 19.3% [libx264 @ 05A2C400] mb P I16..4: 1.3% 16.5% 2.2% P16..4: 36.3% 28.6% 12.7% 1.8% 0.2% skip: 0.4% [libx264 @ 05A2C400] mb B I16..4: 0.4% 3.8% 0.3% B16..8: 40.0% 18.4% 4.7% direct:18.5% skip:13.9% L0:45.4% L1:38.1% BI:16.5% [libx264 @ 05A2C400] final ratefactor: 20.35 [libx264 @ 05A2C400] 8x8 transform intra:83.1% inter:68.5% [libx264 @ 05A2C400] direct mvs spatial:99.2% temporal:0.8% [libx264 @ 05A2C400] coded y,uvDC,uvAC intra: 64.9% 83.4% 49.2% inter: 49.0% 50.4% 4.4% [libx264 @ 05A2C400] i16 v,h,dc,p: 25% 22% 27% 26% [libx264 @ 05A2C400] i8 v,h,dc,ddl,ddr,vr,hd,vl,hu: 10% 7% 23% 9% 10% 10% 10%10% 13% [libx264 @ 05A2C400] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 12% 11% 13% 9% 12% 11% 10% 9% 12% [libx264 @ 05A2C400] i8c dc,h,v,p: 42% 28% 16% 14% [libx264 @ 05A2C400] Weighted P-Frames: Y:18.4% UV:7.9% [libx264 @ 05A2C400] ref P L0: 29.1% 11.3% 15.7% 7.3% 6.9% 4.9% 5.1% 3.4%3.9% 2.7% 2.8% 1.8% 1.7% 1.2% 1.4% 0.9% [libx264 @ 05A2C400] ref B L0: 68.8% 11.4% 5.5% 2.9% 2.3% 1.9% 1.5% 1.1%1.1% 1.0% 0.9% 0.7% 0.5% 0.3% 0.1% [libx264 @ 05A2C400] ref B L1: 91.9% 8.1% [libx264 @ 05A2C400] kb/s:2055.88 As far as I'm concerned it doesn't look that bad to me.

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  • ffmpeg help converting

    - by ellman121
    so I've been trying to reencode a .mp4 video I have into a .avi so my cousin can use it on his Windows machine. He's not very tech-savvy, and doesn't want to deal with downloading any new programs to open .mp4 videos, but thats beside the point. The current string I'm using is ffmpeg -i Courage.Under.Fire.1996.BRRip.H264.AAC.5.1ch.Gopo.mp4 -sameq -acodec copy -vcodec copy CourageUnderFire.avi It produces the video, however doesn't give me any audio. Any assistance?

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  • Unable to watch a high resolution and high frame rate video

    - by Abhijith Madhav
    I have a video of a tennis match whose Resolution = 1280 * 720 Codec = H264 Frame rate = 50fps (Copy paste from info given by totem media player) My laptop is not powerful enough to play this video smoothly. How can I reduce the frame-rate of this video so that my laptop can play it? I have observed that my laptop can play videos with 25fps without an issue. I use ubuntu. I wouldn't mind using windows to edit/re-encode this video.

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  • Best video codec for filmed powerpoint presentation

    - by rslite
    I have some presentations that are filmed. The audio is the presenter and the video is all the Powerpoint slides (size 1024x768, video codec H264, audio codec AAC). I would like to reduce their final file size since a 1 hour presentation is about 800 MB. Most of it is the video part which as I said is mostly powerpoint slides that don't change much over a matter of several seconds. Which codec would be better suited to encode this images and reduce the size of the end file?

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  • VLC or ffserver for http streaming

    - by Maciek Sawicki
    Hi, I don't like asking "OR" questions, but I can't find any comparison on the Internet. I need live http streaming using h264 in flv container. I managed to achieve this with VLC, but with big latency. I have some problems with ffmpeg and I can not make it to work, but I wonder if would it works better then VLC?

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  • When spliting MP4s with ffmpeg how do I include metadata?

    - by Josh
    I have a few MP4s that i want to upload to my flickr account but they have a maximum size of 500mb as mine is only about 550 i was planing to simply split them in half then upload them, but i want to make sure all the meta data is included but it does not seem to be. I have tried each of the following with no luck, (at the end of this post i have the original and the new ffprobe outputs): ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_meta_data SANY0069.MP4:SANY0069A.MP4 SANY0069A.MP4 with the this one I manually produced the individual meta tags that i took from this command ffmpeg -i SANY0069A.MP4 -f ffmetadata meta.txt ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -metadata major_brand="mp42" -metadata minor_version="1" -metadata compatible_brands="mp42avc1" -metadata creation_time="2012-09-29 09:05:50" -metadata comment="SANYO DIGITAL CAMERA CA9" -metadata comment-eng="SANYO DIGITAL CAMERA CA9" SANY0069A.MP4 using the output of the former command i also tried this: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -f ffmetadata -i meta.txt SANY0069A.MP4 Output: sample output from my first command: ffmpeg -ss 00:00:00.00 -t 00:04:19.35 -i SANY0069.MP4 -acodec copy -vcodec copy -map_metadata 0:0 SANY0069A.MP4 ffmpeg version 0.8.12, Copyright (c) 2000-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 File 'SANY0069A.MP4' already exists. Overwrite ? [y/N] y Output #0, mp4, to 'SANY0069A.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 encoder : Lavf53.5.0 Stream #0.0(eng): Video: libx264, yuv420p, 1280x720 [PAR 1:1 DAR 16:9], q=2-31, 9007 kb/s, 30k tbn, 29.97 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Stream mapping: Stream #0.0 -> #0.0 Stream #0.1 -> #0.1 Press [q] to stop, [?] for help frame= 7773 fps=4644 q=-1.0 Lsize= 289607kB time=00:04:19.35 bitrate=9147.4kbits/s video:285416kB audio:4033kB global headers:0kB muxing overhead 0.054571% and finaly, when i compare the ffprobe of the original and the first split part i get the 2 following outputs: original ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069.MP4': Metadata: major_brand : mp42 minor_version : 1 compatible_brands: mp42avc1 creation_time : 2012-09-29 09:05:50 comment : SANYO DIGITAL CAMERA CA9 comment-eng : SANYO DIGITAL CAMERA CA9 Duration: 00:08:38.71, start: 0.000000, bitrate: 9142 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9007 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 2012-09-29 09:05:50 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 2012-09-29 09:05:50 Split ffprobe version 0.8.12, Copyright (c) 2007-2011 the FFmpeg developers built on Jun 13 2012 09:57:38 with gcc 4.6.3 20120306 (Red Hat 4.6.3-2) configuration: --prefix=/usr --bindir=/usr/bin --datadir=/usr/share/ffmpeg --incdir=/usr/include/ffmpeg --libdir=/usr/lib64 --mandir=/usr/share/man --arch=x86_64 --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m64 -mtune=generic' --enable-bzlib --enable-libcelt --enable-libdc1394 --enable-libdirac --enable-libfreetype --enable-libgsm --enable-libmp3lame --enable-libopenjpeg --enable-librtmp --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxvid --enable-x11grab --enable-avfilter --enable-postproc --enable-pthreads --disable-static --enable-shared --enable-gpl --disable-debug --disable-stripping --shlibdir=/usr/lib64 --enable-runtime-cpudetect libavutil 51. 9. 1 / 51. 9. 1 libavcodec 53. 8. 0 / 53. 8. 0 libavformat 53. 5. 0 / 53. 5. 0 libavdevice 53. 1. 1 / 53. 1. 1 libavfilter 2. 23. 0 / 2. 23. 0 libswscale 2. 0. 0 / 2. 0. 0 libpostproc 51. 2. 0 / 51. 2. 0 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'SANY0069A.MP4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 creation_time : 1970-01-01 00:00:00 encoder : Lavf53.5.0 comment : SANYO DIGITAL CAMERA CA9 Duration: 00:04:19.37, start: 0.000000, bitrate: 9146 kb/s Stream #0.0(eng): Video: h264 (Constrained Baseline), yuv420p, 1280x720 [PAR 1:1 DAR 16:9], 9015 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc Metadata: creation_time : 1970-01-01 00:00:00 Stream #0.1(eng): Audio: aac, 48000 Hz, stereo, s16, 127 kb/s Metadata: creation_time : 1970-01-01 00:00:00 I know this is incredibly long but its actually a quite simple question. I thought it would be best to provide as much detail as possible. any advice here would be great, Thanks

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  • Setting font size of Closed Captions on iPhone using ffmpeg or mencoder

    - by forthrin
    Does anyone know how to either: Make ffmpeg set subtitle font size in the output video file Make mencoder produce an iPhone-compatible video file (with subtitles) I finally found out how to get Closed Captions video on iPhone, with mkv and srt files as source material. The secret was using the mov_text subtitle codec in ffmpeg (and turning on Closed Captions in the iPhone settings of course): ffmpeg -y -i in.mkv -i in.srt -map 0:0 -map 0:1 -map 1:0 -vcodec copy -acodec aac -ab 256k -scodec mov_text -strict -2 -metadata title="Title" -metadata:s:s:0 language=eng out.mp4 However, the font size appears very small on the iPhone, and I can't find out how to set it with ffmpeg (the iPhone has no option for this). I found out that mencoder has a -subfont-text-scale option, but I don't have a lot of experience with this program. The following, my best attempt so far, produces an output file which is not playable on the iPhone. sudo port install mplayer +mencoder_extras +osd mencoder in.mkv -sub in.srt -o out.mp4 -ovc copy -oac faac -faacopts br=256:mpeg=4:object=2 -channels 2 -srate 48000 -subfont-text-scale 10 -of lavf -lavfopts format=mp4 PS! As requested, here is the output from mencoder: 192 audio & 400 video codecs success: format: 0 data: 0x0 - 0xb64b9d2f libavformat version 54.6.101 (internal) libavformat file format detected. [matroska,webm @ 0x1015c9a50]Unknown entry 0x80 [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (ac3), -aid 0, -alang eng VIDEO: [H264] 1280x544 0bpp 49.894 fps 0.0 kbps ( 0.0 kbyte/s) [V] filefmt:44 fourcc:0x34363248 size:1280x544 fps:49.894 ftime:=0.0200 ========================================================================== Opening audio decoder: [ffmpeg] FFmpeg/libavcodec audio decoders libavcodec version 54.23.100 (internal) AUDIO: 48000 Hz, 2 ch, s16le, 448.0 kbit/29.17% (ratio: 56000->192000) Selected audio codec: [ffac3] afm: ffmpeg (FFmpeg AC-3) ========================================================================== ** MUXER_LAVF ***************************************************************** REMEMBER: MEncoder's libavformat muxing is presently broken and can generate INCORRECT files in the presence of B-frames. Moreover, due to bugs MPlayer will play these INCORRECT files as if nothing were wrong! ******************************************************************************* OK, exit. videocodec: framecopy (1280x544 0bpp fourcc=34363248) VIDEO CODEC ID: 28 AUDIO CODEC ID: 15002, TAG: 0 Writing header... [mp4 @ 0x1015c9a50]Codec for stream 0 does not use global headers but container format requires global headers [mp4 @ 0x1015c9a50]Codec for stream 1 does not use global headers but container format requires global headers Then the following repeats itself for every frame: Pos: 0.0s 1f ( 2%) 0.00fps Trem: 0min 0mb A-V:0.000 [0:0] [mp4 @ 0x1015c9a50]malformated aac bitstream, use -absf aac_adtstoasc Error while writing frame. I recognize -absf aac_adtstoasc as an ffmpeg option (does mencoder spawn ffmpeg?), but I don't know how to pass this option on (my hunch is this is not even the origin of the problem).

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