I ran ffmpeg -i /sumeet/clip/friends introduction song.mp4 -s 160x120 -acodec amr_nb -vcodec mpeg4 -r 30 -ar 8000 -ac 1 ~/Desktop/friends.mp4 now I compared sample video created by mobile file I created above.code...
I have some .3GP movies that where made with my mobile phone. I can only play the video and not hear the audio. Is there a way to convert these movies to other format so I can edit them. And is there a way to play a .3GP movie in a mediaplayer with the proper codecs?
I am told I need the output file to comply with this
Video Resolution: 480*320 Video Bitrate: 768 kbps Audio Bitrate: 128 kbps Video Format: MPEG-4 (be sure not to use H264, as it�s not supported in the current firmware)
i am using Ubuntu 11.04 on my computer system. I urgently need a good video converter for converting videos.I have already installed FFmpeg and men-coder,Winff etc. The problem is each has its own drawback.For instance ffmpeg cannot convert a .avi to .3gp with audio working. My preferences are the converter should be user friendly, should support all popular video formats.
So I have Postfix working great and I've always used webmail if I needed to send email from PC's outside of $mynetworks. So fast forward to today where I got my 1st Android powered mobile phone and I can configure the Android mail client to send/receive IMAP email but my question is do I need to become an open relay to allow my random wireless providers dynamic range of IP's to send mail via Postfix? Seems extremely vulnerable and scary to think I would have to allow my providers IP range to relay mail via my MTA. I started reading a bit and I think I need SASL authentication and since both Postfix 2.8.1 & Dovecot 2.0.11 are both configured / using TLS, is there anything else I would need or you recommend for sending email from my Android powered mobile?
Recently if I have been getting Unsolicited Bulk Email (UBE) errors when using mobile phones to send. The ip in postfix is allowed to relay but I suspect that the mobile provider is using DHCP to issue address to the phone (though I have check the ip addresses are always allocated the same) I have checked via a fixed ip that is allowed in the relay and this does not generate the same error. Does anyone know what I can change in Amavis or Postfix to sort this out, because legal mail is being rejected via the mobile phones at the moment.
I have Ubuntu 9.04 and just installed Sound Converter. I am trying to convert a bunch of .ogg files to mp3 to play on my iPod and it's not working so well. In the Sound Converter options I have is set to convert to high quality mp3. I choose the folder that the files are in and after a moment (slow laptop) Sound Converter populates, I hit 'convert' and it shows that the conversion completes in two seconds. All that it did was create the new folder structure of artist/album but there is nothing in there. Not sure what I am missing. I used Sound Converter before and it worked fine.
I'm trying to convert a .MTS video file to .mov, and I need to use ffmpeg, because I want it to be scriptable.I managed to convert the file to .avi using this command:Code:ffmpeg -i INPUT.MTS -vcodec libxvid -b 18000k -acodec libmp3lame -deinterlace -ab 192k -s 1280x720 -r 50 OUTPUT.avi
The problem is i unable to stream this kuraiamr_nb.mov file at my streaming server. when i play this file in quicktime player its stating that this is a unsupported media type.
I am looking for an option to convert .vob file to mp4 format. Command "ffmpeg -i S.vob x.mp4" is giving error message "[libfaac @ 01cac110] Unable to load libfaac.dll". I tried to convert in other formats like .avi with the same command and got success. Conversion is decreasing the quality like hell. Though size of the file after conversion is as big as original then why this bad quality. Some where I have read if I will not specify any argument it will keep the quality same but here it is not happening. Not sure why....
I need to play or preferably convert (i.e. to MP3) old SNG files, which contain voice records. From what I could find, it's basically a MIDI created by synthetiser. I think it was recorded by some ancient VLC player. I failed so far to play it on anything I could download.
I read through more of this thread than I wanted to. [URL] ....
Does libav provide command line utilities to do something like converting ovg to avi, or determining the length of an mp3 file? Why is there absolutely no explanation as to why libav was preferred over ffmeg?
I've got a bunch of HD files in .mkv format that I want to convert to a format that the 360 will read (currently just trying .avi files) so I can stream the media to my xbox360. I'd rather avoid transcoding the entire file, so I tried unpacking the .mkv file into audio and video (and subtitle if necessary) and repacking said file into avi using ffmpeg. The repacking is where I am having the problem. Here's the commands I tried at first (it might look familiar):
Error while opening codec for output stream #0.1 - maybe incorrect parameters such as bit_rate, rate, width or height Where stream #0.1 is the output audio stream to the .avi file. I think I have narrowed it down to the fact that the input streams are all 6 channel (5.1 surround sound) which mp2 doesn't support (the ffmpeg audio default) so I did this instead:
to change the sound to stereo it works. Problem is I'd like to keep my media in surround sound. Any thoughts on how to get all 6 channels in the new mp3 file? I'm fairly certain that mp3s support 5.1. And after some messing around, I haven't found that my ffmpeg can transcode audio format from 5.1 back to 5.1. So since (I think) the 360 can play .ac3 files I decided to just copy both the video and audio file to the .avi file sans transcoding using
however now about halfway through, I run into this error:
Code:
NULL @ 0x22f3a00]error, non monotone timestamps 62562 >= 62562
av_interleaved_write_frame(): Error while opening file I can't seem to find any solution to this problem anywhere online, does anyone know how to get around this? I'd prefer to use ffmpeg for this if possible. (so I can write a script and just attack an entire directory at once) For what it's worth, I'm using the latest ffmpeg in the ubuntu (karmic) default repositories.
I have a .mkv container with proper video (h264/~6000kbps) and audio (DTS/1500kbps). For some reason I want to have similar quality but with another video and audio codecs. For video it goes well and ffmpeg is very helpful here, no problems. Also, I want DTS to become AC3/320-448kbps/5-6 channels. Commands like this:
gives me not proper reordered sound in channels, so I hear center on the right, music on the left, etc.
Is there a way to make that properly with ffmpeg?
Also, I have a question about multi core encoding cause I have a quad processor. Ffmpeg seems not using them in all the way, I get something like 25% of CPU usage while encoding. Command -threads, even using the h264, does not make any extra load of cpu.
I'm making little animated bits in GIMP with the gap package. I can only get GIMP to encode the files as .avi. Problem here is that when I try to use any converter, it compresses 80MB files down to 1MB-4MB files and, unsurprisingly, the video quality is terrible. WinFF and ffmpeg (via terminal) output at a drastically reduced size. Is there some way to keep the files big/big-ish? I don't want to lose any detail.
Recently I installed gtk-recordMyDesktop, so that I could record my movements and send the Media File to my dad, so he could learn how to use Ubuntu too [plug]My problem is converting the .ogv file to .avi - I have tried ffmpeg but it loses quality dramatically.Can anyone tell me how to get better quality video out of ffmpeg, or let me know another way to convert the .ogv file to .avi or something that is widely used?
I've been recording one of my classes with my android phone, and the file it gives me is an .amr file. I'd like to convert that to .mp3, and I was hoping I could use ffmpeg to do it via the command line. Here's what I did, and the output that I got:
Quote:
roger@KarmaLaptop:~/Desktop$ ffmpeg -i Net202-week6.amr Net202-week6.mp3 FFmpeg version 0.6-4:0.6-2ubuntu6, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 5 2010 22:36:53 with gcc 4.4.5 configuration: --extra-version=4:0.6-2ubuntu6 --prefix=/usr --enable-avfilter
I have a file with about 6 .flv files and I wish to batch convert them to libmp3lame. I have tried making a #!bin/bash script with all the files in e.g.
I have inserted all the filenames individually into the script but when I ran it I got too many errors and i was wondering if someone knew a quicker way to do it e.g. a script that would batch convert all .flv files in that folder to .mp3 format.
I recently developed a taste for the Alac format and ffmpeg will oblige with this line of code Code: ffmpeg -i <input> -acodec alac <output>.m4a and this worked beautifully one file at a time and How does one do all the files in a given folder? Is there an asterisk one adds as in shntool.
I'm running ClipBucket on my Ubuntu 10.04 LTS server x64.Conversion of AVI to FLV works just fine, however when I try to upload an MPG file, it fails. Apparently I should recompile ffmpeg to allow for MPG > FLV conversions.
It returns an error dealing just with the h264 codec saying that I need to use a vpre parameter? I can't find any documentation on using the vpre parameter.