Ubuntu Multimedia :: Convert Video To Audio Use Ffmpeg?
Jul 15, 2010I want to convert all video file in folder a become audio file in folder b use ffmpeg, how to type it in terminal?
View 7 RepliesI want to convert all video file in folder a become audio file in folder b use ffmpeg, how to type it in terminal?
View 7 Repliesi 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.
View 2 Replies View Relatedcommand line option in ffmpeg to do this
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)
[Code]....
I'm trying to write a bash script for gpodder to automatically convert video podcasts to play on my media player. I'm using ffmpeg for the conversions (compiled myself with all codecs enabled). I'd like to avoid resampling the video or audio whenever it's unnecessary but ffmpeg seems to want to resample my video even if I only give it audio parameters to change.For example I have a test video with the following parameters:
Code:
Stream #0.0(eng): Video: h264, yuv420p, 640x480, 1394 kb/s, 29.97 fps, 29.97 tbr, 2997 tbn, 5994 tbc
Stream #0.1(eng): Audio: aac, 44100 Hz, stereo, s16, 159 kb/s
[code]...
a movie is encoded with AC3 in 6 channel audio, what I get out is all of the sounds except for voices, which in 5.1 would be sent to the center channel. What I usually do is fire up avidemux and convert the audio to mp3 stereo, as converting to a 5.1 format usually ends up with a very odd sound (like running everything through an echo chamber). What I'd like to do is run a script to batch-convert these files from AC3 to MP3. The video format may vary, but they are usually XVID. I am comfortable at the command line, but I am not well-versed in audio/video tool terms. I don't need anything extravagant, I just want something that works. Heck, even if it is done one at a time, having a shell script that I can use to simply type:
tool.sh inputfile.avi outputfile.avi
I am using ffmpeg for merge wav files to a mov video. My doing is below
1. First extract audio (wav file) from video
2. Create wav file from mp3 track 1
3. Create wav file from mp3 track 2
4 Merge extract audio from video with track 1 and track2.
Now finally create a new video with original video's video stream and merged audio stream.
Process is working. However final video is 3-4 times greater than original one. I want that final video should be near about size of original video. As I understand, all three wav files (created from ) make video larger.
Example commands i using is as below:
Is there a way I can convert video with ffmpeg to be in mp4 format and play on either iphone or android?
I can't seem to get anything that I convert to play on a phone.
I can get other mp4 files I found to play on the phone but any file I convert into that format won't actually play.
I'm trying to encode a wmv file to flv with ffmpeg. The video codec is WMV3 and the audio codec is wmap (Windows Media Audio Professional). The command I use is:
Code:
/usr/bin/ffmpeg -i <input file>.wmv -s 640x350 -b 1024k -r 25 -ar 44100 -ac 2 -y <output file>.flv
Output:
Code:
FFmpeg version 0.5-svn17737+3:0.svn20090303-1ubuntu6, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-postproc --enable-swscale --enable-x11grab --extra-version=svn17737+3:0.svn20090303-1ubuntu6 --prefix=/usr --enable-avfilter --enable-avfilter-lavf --enable-libgsm --enable-libschroedinger --enable-libspeex --enable-libtheora --enable-libvorbis --enable-pthreads --disable-stripping --disable-vhook --enable-libdc1394 --enable-shared --disable-static .....
Stream mapping:
Stream #0.0 -> #0.0
Stream #0.1 -> #0.1
[wmv3 @ 0xb32c00]Extra data: 8 bits left, value: 0
Unsupported codec (id=86056) for input stream #0.1
As you can probably see the audio codec is not supported. Is there a way to encode WMV3+wmap with ffmpeg or any other tool in Linux? Windows Media Encoder in Windows is able to encode such files to a supported codec. For example: WMV3+WMA2/WMV2+WMA2, I could then encode it in Linux. I'm trying to find a way to directly encode WMV3+wmap in Linux.
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):
Code:
mkvextract tracks file.mkv 1:temp_video.avi 2:temp_audio.ogg 3:temp_sub.srt
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy file.avi
However, I keep getting this annoying error:
Code:
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:
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame file.avi
I'm still getting the same error, but when I use
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec libmp3lame -ac 2 file.avi
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
Code:
ffmpeg -i temp_video.avi -i temp_audio.ogg -vcodec copy -acodec copy file.avi
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 want to convert a song from ogg format to mp3 using ffmpeg the command i use to do this is
ffmpeg -i /home/john/Music/input.ogg to mp3 output.mp3
i then get then get a message back saying
"unknown format"
how do i correct this ?
The command line to convert to ogg theora
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
runs perfectly on 9.04 but on 10.04 it gives errors:
Code:
ffmpeg -y -i input.flv -sameq OUTPUT.ogv
FFmpeg version SVN-r0.5.1-4:0.5.1-1ubuntu1, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --extra-version=4:0.5.1-1ubuntu1 --prefix=/usr --enable-avfilter
[Code].....
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:
Quote:
ffmpeg -i in.dts/mkv -acodec ac3 -ac 5 -ab 448k out.ac3
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.
View 6 Replies View RelatedRecently 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?
View 3 Replies View RelatedI'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
[Code]....
I am trying to use ffmpeg and mencoder to slice a portion of my video out but the audio is always 2 seconds out of sync.
ffmpeg -i bird.avs.MP4 -acodec copy -vcodec copy -ss 00:08:40 -t 00:01:00 test.mp4
mencoder -ss 00:08:40 -endpos 00:01:00 -oac copy -ovc copy bird.avs.MP4 -o test.mp4
mencoder -ss 00:08:40 -endpos 00:01:00 -oac copy -ovc copy -noskip bird.avs.MP4 -o test.mp4
[Code]....
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.
Code:
ffmpeg -i filename.flv -sameq -acodec libmp3lame -f asf filename.mp3
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.
View 3 Replies View RelatedToday I was trying to convert a video into Gray-scale.(Black & White) I was successful for avi, mp4 ..
And Now trying for 3gp..
Its giving me error which says like The parameter like bit rate, height , width provided is wrong..
what exactly extra parameter I need to put for 3gp
I tried to convert an AVI-video to a MP4-video for my ipod. Therefore I used ffmpeg as I can read in the FAQ
Code:
ffmpeg -i in.avi -acodec libfaac -ab 128kb -vcodec mpeg4 -b 1200kb -mbd 2 -flags +mv4 -trellis 2 -cmp 2 -subcmp 2 -s 320x180 -metadata title=X out.mp4
The problem is, that the given encoder libfaac is not installed and I can't get it through the pack-adminstration
[Code]....
when i had windows xp i used to convert almost any videos video into audio (the whole video into audio) by going to [url], downloading vdownloader and just inserting the url in the box provided; waiting for a couple of minutes and having an mp3 format file at the location that i want it to be in the desktop. when i searched for something similar for linux, because vdownloader was an .exe file and obviously wasn't going to work on ubuntu.
i found that i could get a video downloaded from videos but not an audio - i wait for the video to finish buffering; click on places; computer; file system; tmp; then rename the video folder and copy/relocate it to where ever on the desktop. ending with this video format "flash video (video/x-flv)". my question is: is there a linux program where i get to convert the entire video format to an audio format, possibly mp3?
I creating a mini soft for screencast (audio + video) with ffmpeg.For the video,it's ok.For the sound, the capture of sound of my webcam (/dev/dsp1) it's ok.
Code:
ffmpeg -f oss -ar 44100 -i /dev/dsp1 -acodec mp2 -ab 128k test.mp3
But for my audio desktop :
[code]...
I'm looking for configuring a Ubuntu box, we need to capture alsa output into ffmpeg with jack or OSS but not pulseaudio, as that tends to glitch badly. We have things working, but we are sort of hacking around in the dark, we play music, but as we aren't linux audio experts ourselves.
View 1 Replies View RelatedI'm having trouble to find the right ffmpeg options to encode a video that can be read on a htc G1 cell phone. I have used several codecs and formats but none is working.
I have followed these instruction to install ffmpeg and x264 [URL]
Here is my ffmpeg config :
Code:
FFmpeg version SVN-r24953, Copyright (c) 2000-2010 the FFmpeg developers
built on Aug 27 2010 22:44:01 with gcc 4.4.1
configuration: --enable-gpl --enable-version3 --enable-nonfree --enable-postproc --enable-libfaac --enable-libmp3lame --enable-libopencore-amrnb --enable-
[Code].....
I am trying to get a listing of video information using ffmpeg or mplayer. I have found a way BUT it is only returning 1 piece of info..
Here is what I'm doing in PHP:
print exec("/usr/local/bin/mplayer -identify file.flv -ao null -vo null -frames 0 2>/dev/null | grep ^ID_");
But this will not provide an entire listing, it will only print out 1 thing such as:
ID_AUDIO_CODEC=mp3
I can specify what exactly I want to know something like:
print exec("/usr/local/bin/mplayer -identify file.flv -ao null -vo null -frames 0 2>/dev/null | grep ^ID_VIDEO_WIDTH");
but I want to just have all the info available at once instead of having to loop through like 20 times..
is there a command to create a video from a mp3 and a png suitable to upload on ..... if anybody knows?
View 4 Replies View RelatedAnybody had any success in getting ffmpeg to work as advertised with video capture from a webcam? I really want to convert the webcam output to VP8 or H264, but apparently ffmpeg can't even capture the webcam with a video4linux device.
Code:
I 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 some videos in an mkv container that are 1920 pixels wide, but less than 1080 pixels high. This causes problems when playing the videos on a PS3 (after converting to an AVCHD system), because the PS3 won't centre the video, leaving a very large black bar at the bottom but none at the top. Is there a way to use mencoder or ffmpeg to losslessly add padding to the top and bottom to make the video 1920x1080?
If I use
Code:
ffmpeg -i video.mkv -acodec copy -vcodec copy -scodec copy -padtop 132 -padbottom 132 out.mkv
then ffmpeg won't do the padding, and the resulting file is basically exactly the same. If I use
Code:
ffmpeg -i video.mkv -acodec copy -vpre libx264-lossless_fast -scodec copy -padtop 132 -padbottom 132 out.mkv
I get
Code:
Output #0, matroska, to 'anr.mkv':
Metadata:
encoder : Lavf52.64.2
[code]......