i am using ubuntu 8.04.I wanted to clip from a movie so I ran
Code:
mencoder -ss 2453 -endpos 34 InpuTFile -acodec copy -vcodec copy -o Output
I changed the ss value still it cut from same frame of the video.It seems mencoder works faster than ffmpeg but jumps to keyframe hence cannot jump to accurately specified second.
In cdrecord's man page, EXAMPLES, it says:
Quote:
To copy an audio CD in the most accurate way, first run
cdda2wav dev=2,0 -vall cddb=0 -B -Owav
and then run
cdrecord dev=2,0 -v -dao -useinfo -text *.wav
Now, cdda2wav samples the drive output. But the signal is already sampled in the CD. Do you think this is accurate?
We are using thin client systems in our work environment. There is a central ubuntu server and by using thin clients, we are connecting to our homes. The problem is when I try to install an application, it reported me that I had 200 mb of disk space. But when i try to look from console, I see that /home folder has over 250 gb s of disk space. Even when I try to look from baobab, Disk Usage Analyzer in Ubuntu, i see that my home file system is full.
So what's the reason that I am receiving different kinds of disk space report from different sources? Our system admin here told me that some applications foolishly try to see the physical devices on the thin client and got confused as a result. Is this true?
I was just wondering if it is possible to output an encoding file to http, mms, udp etc... (eg. http://127.0.0.1).I found a way to input these streams and encode them I just cant find a command to output to these.
View 1 Replies View RelatedI have dual boot on my comp. Windows XP and Fedora 11 Now in both systems time zone is set to Belgrade ( which is my time zone), but when I setup clock in fedora to be, let's say 16.15h, then when I swich to windows it says time is 14.15h. When I setup in windows on 16.15h, and I swich to fedora, it says time is 18.15h. So I can't get accurate time on both systems in no way.
View 6 Replies View RelatedIs there a way to instruct mencoder to stop streaming/recording after a specified period associated with a loss of signal?I use mencoder to record over-the-air television recordings. Most of the time there are no problems.Occasionally a station's transmission signal disappears, caused by quirky atmospheric conditions, usually at night.Noticing these signal outages is easy by the mencoder error message:dvb_streaming_read, attempt N. %x failed with errno %y when reading %z bytes"Lengthy outages are unbearable because mencoder silently waits forever rather than quit.
The significant problem with lengthy signal outages is mencoder continues recording for the time specified by the -endpos parameter. Thus, for example, if the recording is scheduled for 2 hours and there is a 20 minute loss of signal, the recording does not end until 2 hours and 20 minutes after starting.Generally, when the interruption is lengthy the recording is ruined and I don't care to watch. I would like to programmatically tell mencoder to stop waiting and quit.Is there a way to do this? Say, after ten minutes of no signal, just terminate do not notice any parameters that might offer that option, but I could be overlooking something obvious. I tried the -skiplimit parameter with no success.
I'm new to commandline, and I'd like to:
$mencoder -ffourcc DX50 -oac mp3lame -ovc lavc about 10 files...
How to write that?
I have an Asus EEE PC900. Just installed 10.04 netbook remix and everything works fine straight out of the box. it works great and is a vast improvement on windows xp which was previously installed on the asus. Just one small thing - the battery meter is never accurate as far as time remaining goes - it currently says i have 19 hours 55 minutes to go - unfortunately this is not quite true i suspect. It was the same with 9.10, Does anyone have a solution for this?
View 4 Replies View RelatedI tried to trim a video in FFmpeg using this command:
Code:
ffmpeg -ss 00:20:48.500 -t 00:01:00 -i INPUT.mp4 -acodec copy -vcodec copy OUTPUT.mp4
But FFmpeg is not accurate and it started the video from a nearby point instead (from 00:24:46~). I tried to add 2 seconds to my starting point and it took another frame (not what I wanted).
The video source is H264 video with AAC audio.
I have tryed out this operator on program:
Code:
#include <iostream.h>
#include <iomanip.h>
#include <fstream.h>
ofstream myfile;
[code]...
which definitly is not the same number. I guess somewhere in the convertion from double to char* ("<<") something is not right and what can i do to save these double numbers in an accurate manner in a file?
Code:
mencoder The.Negotiator.mkv -of rawaudio -oac mp3lame -ovc copy -o sound-mencoder.mp3
I got a 195.3MB mp3 file
[code]...
I've multiple jpegs uploaded form IP cam via FTP, I use mencoder to periodically pack them into single avi file, problem is that sometimes one or two jpegs submitted by cam are broken, and this make mencoder exit, without producing movie
View 1 Replies View RelatedI'm using mencoder to capture audio from a Encore ENLTV-FM3 video capture device. I have recently noticed that, since one week ago, when the machine was forcibly restarted due to a power outage, all recordings are slightly pitched, they play back slower than they should.
I narrowed down the problem to the following command line:
$ time mencoder -really-quiet -tv driver=v4l2:device=/dev/video1:chanlist=us-cable:audiorate=32000:alsa:adevice=hw.1:input=0:amode=1:normid=11 -endpos 00:10:00 -ovc copy -oac pcm -of rawaudio -o test-32000.wav tv://69
real 9m54.886s
user 0m5.536s
sys 0m1.740s
$ ls -l test-32000.wav
-rw-r--r--@ 1 martin martin 76800000 Mar 15 17:20 test-32000.wav
Somehow, mencode managed to gather precisely 10 minutes worth of raw audio in 9m 55s. That's not physically possible, unless the capture device's A/D converters are "overclocked". I can't think of any other explanation besides hardware failure. Can that be? Could it be that something got burnt during the power outage and now the capture device's internal clock went nuts?
Since the machine's restart, I've also noticed dmesg is flooded with entries like this:
CE: hpet increased min_delta_ns to XXX nsec
Which seem to indicate that the computer's high precision event timer is somehow out of sync. Does this have to do with the audio issue? Can it be that the audio converter's sample rate is linked to the HPET? I'm totally lost here. Has anyone bumped into something similar?
i want to convert my mp4 video into h264 format, but how to convert with mencoding,
View 1 Replies View RelatedI am getting following error while trying to record an ".avi" file
Movie Creation Error: Error 8 - Failed to execute child process "mencoder" (No such file or directory)
[root@localhost iNSpect_4_beta_3]#
I have found that if I change the FPS of a video, the audio is out of sync.
Is there a way in mencoder to have it correct this, and maintain the correct sync?
I don't think it is just a matter of audio/video delay. I have tried many times to correct this via that method, and it doesn't come close. Although, if I encode the video to another format, but with the original fps, the audio is sync'd.
compile mplayer & mencoder with VP8 support? also what with VP8 support in rpmfusion?
View 3 Replies View RelatedA recent update to ffmpeg, in which I've learned was an upgrade to the restricted ffmpeg as opposed to the unrestricted default one that is in the official repos, has now made it unable for me to use mplayer or Devede. Reading up about this, especially from here: [URL].. I have learned it's a frequent problem when installing the restricted ffmpeg. Basically what Devede says at startup is:
Code:
Cant's find the following programs:
mplayer
mencoder
Devede needs them to work. If you want to use Devede, you must install all of them. Using the terminal to open mplayer (or mencoder), I get this error: mplayer: relocation error: mplayer: symbol codec_wav_tags, version LIBAVFORMAT_52 not defined in file libavformat.so.52 with link time reference From my research, seems as though the problem lies in libavformat52 not being installed (or removed) and libavformat-extra-52 is installed, which is the same thing, but because it's the restricted version, it is called something different. Anybody know if that is correct?
Also, checking up, libavformat-unstripped-52 isn't installed but available, so should I install that to replace the missing libavformat52, as when I sudo aptitude install libavformat52, it says it conflicts with libavformat-extra-52?
I installed libavformat-unstripped-52, which didn't conflict with the already installed libavformat-extra-52, just to see if it would basically just replace the missing libavformat52, which conflicted with libavformat-extra-52 and couldn't be installed safely, but after I did, nothing changed. I still get the same error with opening Devede and mplayer (or mencoder) in the terminal.
I have an 8Gb mkv file, which I tried to encode to avi using mencoder, it took more than 6hrs and when completed, the quality of the avi was pretty poor.Is there a way to increase the quality or is there a program that can encode mkv files? The mkv will not play on my system without stuttering badly.
View 8 Replies View RelatedWhere to get guidance as new user of acidrip? I can't find a "how to"
View 7 Replies View RelatedI recorded a TV show on my pc using mythbuntu software (ubuntu version of mythtv). It cut out all the commercials and then transcoded it to a nice 40 min nuv file that I put onto a usb thumb drive. Now it's portable and vlc can play the file...on a pc. I want to burn it onto a DVD so I can put it in any NTSC DVD player. Also, the file was recorded at 480x480 and I want to change the size to widescreen (16:9). Additionally, the top of the screen recorded a thin horizontal line of junk that needs to be deleted.
Now, I've googled around and found this line in ffmpeg that should work:
ffmpeg -i V-RedSky.nuv -croptop 4 -y -target ntsc-dvd -sameq -aspect 16:9 RedSky.mpg
The new 16:9 video size is great, the annoying horizontal line got cropped, and I got a nice mpg file...but the audio is now out of sync. I tried ffmpeg with fewer options and the audio still got out of sync.
So, I'm thinking I should try different software. Maybe mencoder. How could I put the above ffmpeg options into mencoder?
I edited some mpeg videos using Mencoder to crop unwanted footage from the beginning and the end. It worked, but now the file does not have a "visual" icon, i.e. a still from the video. The properties of the new file also reveal that it thinks it's an audio file! The file still plays (as a video) okay though.A typical command I used to extract just the first 14 seconds of the video was:
mencoder "e03.mpg" -o ee03a.mpg -ss 0:0:0 -endpos 0:0:14 -oac copy -ovc copyI did try to make up a new icon by making a snapshot of the movie playing, then using the image to replace the icon in the properties window. (I am using Nautilus file manager in Ubuntu). The icon it now gives me looks like a small sheet of paper with one end curled up.I can appreciate that editing the video length might have "lost" the icon information, but is there any way I can put it back again
I'm trying to convert OGG clip to avi using mencoder:
Code:
mencoder myclip.ogg -ovc lavc -oac mp3lame -o myclip.avi
It works OK except that while the original clip has 16:9 aspect ration, the resulting AVI is 4:3
I'm trying to convert a video in .ogm format to .avi format so I can stream it to my PS3 via uShare. I'm using Mencoder and the following command to do it (to transcode the audio from vorbis to mp3 and the video from mpeg4 to xvid):
Code: mencoder input.ogm -oac mp3lame -ovc xvid -xvidencopts pass=1 -o output.avi.The only problem is, the video contains 2 audio streams - one Japanese and one English. Needless to say, I need the English one. The above command only seems to transcode the Japanese audio though!Here's what ffmpeg has to say about the input file:
Code:
Input #0, ogg, from 'input.ogm':
Duration: 00:24:16.12, start: 0.000000, bitrate: 1280 kb/s[code]...
what I can do to specify the correct audio stream to encode? Also, how can I keep the bitrate the same as the output.avi seems to default to 96kbps?I'm not sure which audio stream is which but a little trial and error will sort that out in no time.
I'm trying to transcode and merge sub with mencoder in order to watch a video on my nokia cellphone.
I'm using this command :
Code:
without success. The transcoding works, but no subtitle is merged.
The transcoding log is here
I'm using Karmic.
So I typically use cat to put my avi files together.. but today I figure lets see about ripping over so I can watch on my Droid.
As nobody has a xvid or vlc player on the droid (yet.. coders are doing alot for Android ya know) I will have to convert the video down and change the format.. So.. can anybody tell me a mencoder command line directive to get as close to this as possible (I found ipod settings leave the actual video lacking).
Video
Format MP4
Encoder H.264
Resolution 854 x 480 (or 720 x 480; 480 x 320; 320 x 240)
Frame Rate 12 fps - 30 fps
[Code]....
how to remove audio from an .ogv video (GTKRcordmydesktop) while converting to .avi using mencoder?
I've started with this command:
Code:
mencoder -ovc lavc -ofps 15 -oac mp3lame -af volnorm=1:1 out-2.ogv -o sound_editing.avi
Using Miksoft's Mobile Media Converter, and I cannot convert files to MPEG4.[URL]Converting to MPEG1/2 works fine, but trying to convert video for my son's PSP always produces an error. Here's the output:
Code:
>> Command executed:
"/opt/MIKSOFT/MobileMediaConverter/lib/mencoder" -oac lavc -ovc lavc -of lavf -lavcopts
[code]....
I'm looking for help to write a little script to convert a whole folder of .mkv to .avi. This is the mencoder bit i've been using to do so
Code:
mencoder -mc 0 -noskip -vf expand=:::::16/9,scale=720:480,hqdn3d,harddup -ovc xvid -oac mp3lame -xvidencopts fixed_quant=3.8:me_quality=6:noqpel:nogmc:trellis:chroma_me:chroma_opt:hq_ac:vhq=4:lumi_mask:max_key_interval=300:quant_type=mpeg:max_bframes=2:closed_gop:nopacked:profile=asp5:autoaspect:bvhq=1 -lameopts vbr=2:q=6:aq=2 -o [file destination] [source file] 1/"
and it seems to work fine. what i'm looking for is the ability to go into a directory and use the script like so
[Code]...
I have some video files (.avi and .swf) & I need to convert them to the following format:
Stream #0.0(eng): Video: mjpeg, yuvj422p, 320x240, 30 tbr, 30 tbn, 30 tbc
Stream #0.1(eng): Audio: pcm_u8, 8000 Hz, mono, s16, 64 kb/s
how to do this with mencoder or ffmpeg?