I'm writing a wrapper for FFMPEG and need help of a multimedia guru. The more I read about containers, formats and codecs the more confused I become, it all seems to be a pretty big mess. To convert a video file with ffmpeg you need to specify a few arguments for ffmpeg:
- input file
- some video/audio options (bit rates, sizes and so on)
- format (to force format)
- codec (to force codec)
- output file
My confusion is all about the format and output file extension. If I want to encode my file in MPEG-4, I'll use either 'libx264' or 'libxvid' codec. Then the encoded stream needs to be stored in a file. If I use '.mp4' as the output file, it'll be stored in that file. If I use '.mkv', the video will still be MPEG-4 but stored in a Matroska container. But what do I do with the format option? When I force the format to 'mp4', does it mean I don't have to enter output file, it'll be stored as '.mp4' anyway? Or if I use '-matroska' will I still have to put it into a container '.mkv'? Can I use a '-matroska' format but '.mp4' output file for example?
What's the difference between the format and output file in this context? Or what does forcing the format really do?
I've probably exhausted all possible ffmpeg argument combinations for encoding with a libx264 codec - none worked, the codec always either segfaults or tells me incorrect parameters. I've installed, uninstalled and re-installed all available versions of the codec - no difference. Did anyone have any luck with it? Are there any tricks or conflicts in different arguments? Does the input file have to be in any particular container/format/codec for the x264 to work on?
My examples: >ffmpeg -i video.mpg -vcodec libx264 video.mkv >... Error while opening encoder for output stream #0.0 - maybe incorrect parameters such as bit_rate, rate, width or height >ffmpeg -i video.mpg -vcodec libx264 -b 4000 -s hd1080 -f h264 video.mkv Same error If I use any other codec, they all work fine. give a working ffmpeg arguments example,
So I've been trying to convert .avi files to .mov using FFmpeg. I input something like this: Code: ffmpeg -i in.avi -b 1000 out.mov
And it says: Code: Unsupported codec for output stream #0.1 So I tried adding -acodec mp3 OR -acodec libmp3lame and it says unknown codec.
I went to install libmp3lame-dev using Code: sudo apt-get install libmp3lame-dev
And it comes up with this: Code: The following NEW packages will be installed: libmp3lame-dev
The following packages will be REMOVED: libdc1394-22{u} libfaac0{u} libxvidcore4{u} linux-headers-2.6.28-11{u} linux-headers-2.6.28-11-generic{u} I'm fairly sure I don't want those packages removed? Should mention I'm on Ubuntu 9.04 64-Bit.
I use Kino to transfer dv from my camera and also use it for video editing, but there are very few GUI options when it comes to transcoding the edited files into, say, h264. Kino is using ffmpeg, and I know there are a plethora of options. For instance, here is the output of the logfile from a file I made with h264 and the best quality available in Kino.#options: 768x576 fps=25/1 cabac=0 ref=1 deblock=0:0:0 analyse=0x1:0 me=dia subme=8 psy_rd=1.0:0.0 mixed_ref=0 me_range=4 chroma_me=0 trellis=0 8x8dct=0 cqm=0 deadzone=21,11 chroma_qp_offset=-2 threads=2 nr=0 decimate=1 mbaff=0 bframes=0 keyint=300 keyint_min=25 scenecut=0(pre) rc=abr bitrate=2048 ratetol=2.0 qcomp=0.50 qpmin=2 qpmax=31 qpstep=3 ip_ratio=1.25 aq=1:1.00Frankly, I am not that impressed with the result, and would like to use settings that give a better result, e.g. adjusting bitrate and more passes
I've been trying to re-mux some mkv files into mp4 files recently and ran into this issue. The mkv files contain an h.264 video stream, two aac audio streams (english and japanese) and one subtitle stream. All I'm trying to do is move all four streams into a mp4 container using the following command:
Since the first audio stream doesn't seem to be throwing an error I'm a little confused as to why the second (with the same codec) would give this error. I would appreciate any help with this problem, or alternative solutions to accomplishing the mux.
OpenCV doesn't work. I wonder why would anyone create a RPM package and not bother to check whether it actually works? Do they get paid per package at Red Hat? The problem seems to be in cap_ffmpeg.cpp. Somehow, OpenCV cannot resolve the ffmpeg CODEC libraries, but the annoying part is that it compiles and installs error free - it just doesn't work. Without any error messages, it is really hard to figure out what is wrong.
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.
i can't install multimedia codec, after installing i restart PC and then linux writes error kde4int i hadn't this error before, it begins after installing new monitor Samsung syncmaster B1940
I want to run the following in Totem or Audacious. It's a radio stream. Here's the link...LBC 97.3The only trouble is it keeps saying I need the following codec MMSH needs to be installed. Any idea how I go about this? All other multimedia codecs are working fine.
Running Opensuse KDE 11.2 64bit. Trying to backup DVD, but no sound after playing ripped copy. After looking at the settings in K9copy and DVD95, the sound section says "Unknown AC3" where it normally says "English AC3". I've tried mencoder and ffmpeg.
I've been having a somewhat disagreeable problem updating ffmpeg and all the other resources that need updating (libavcodec52 et al) for some time now. I'm asked to delete most of the software/libs that are needed for certain software (2man2dvd, etc...). At times it looks like I have some serious contortions to go through in order to update these rpm's. I'm using 11.0.
I checked for updates today and noticed that the Packman repository have switched to the new forked FFmpeg library: LibAV. This include gstreamer and other dependent libraries.
VLC runs without a problem but i wonder what other applications depending on FFmpeg will not run...
It looks like FFmpeg was removed from the repository.
I am having problems with ffmpeg. My goal is to capture a video stream from my webcam and feed that into a webcam-capturing program. But to get that to work, I will need ffmpeg to work. I need the following command to work, but I get an error:
Code: $ ffmpeg -b 100K -an -f video4linux2 -s 320x240 -r 10 -i $device -b 100K -f image2pipe -vcodec mjpeg - | perl -pi -e 's/\xFF\xD8/KIRSLESEP\xFF\xD8/ig' ffmpeg: relocation error: /usr/lib/libavfilter.so.2: symbol avformat_find_stream_info, version LIBAVFORMAT_53 not defined in file libavformat.so.53 with link time reference
I'm trying to record a video with sound through FFmpeg, and while it records the system sounds just fine, it doesn't record the microphone (in fact, it crashes on doing so!). The strange thing is that if I use arecord, it works just perfectly, but through ffmpeg it crashes.This is the command that I use to launch FFmpeg:
I've been enjoying using Rhythmbox to rip a stack of CDs. Initially I had problems getting it to recognise Lame but following uninstalling and reinstalling it, it seemed to be working well. That is, until I looked in my Music folder.
It turns out files have been placed in individual folders, named "number - title.udio" (e.g.: "11 - She Can Do What She Wants.udio". The mp3 file itself is named identically throughout - it's the string for the gstreamer settings, i.e. "x-raw-int,rate=44100,channels=2 ! lame name=enc preset=1002 ! xingmux ! id3v2muxmp3".
The settings within Rhythmbox's properties for folder and file naming are all set to the sensible default. This is a pain. In the short term I'm going to try EasyTAG to sort it out, but ultimately I need to sort this or find alternative software.
I've been reluctant to post regarding this topic as I have tried and retried all of the "noob solutions" to get flash sound working but no luck- I need help! I suspect that it has to do with my system showing two sound cards (0 and 1) ; I tried different asoundrc file options from the ALSA Wiki but no luck. I am still left with Flash=No Audio.
OpenSuse 11.2 linux 2.6.31.12-0.2-desktop x86_64 ATI fglrx video driver v10.4 ATI radeon hd3200 video adapter
VirtualBox (any version) crashes at startup if the video acceleration options are enabled. This was not a problem before installing fglrx. The crash occurs when VirtualBox queries the video driver about OpenGL, I am guessing from the log file.
To (supposedly) have access to all of the features possible in the video adapter I installed the ATI fglrx video driver. I (apparently foolishly) chose the custom, installation-specific path rather than the more general default path.
After installing the driver I saw no observable improvement in performance or feature set. Worse, there are couple of new features that are decidedly undesirable, and cannot be disabled. At least one pre-loaded font simply disappeared.
So I reverted to the original (open source?) radeon driver ("sax2 -r -m 0=radeon" at runlevel 3). All seemed welluntil I ran VirtualBox.
One thing I have not tried is to run the fglrx default installation. I fear making things worse.
Can anyone suggest what might have happened with the fglrx installation, and how to fix it?
I'm trying, reeeeally trying to use Ekiga. But the video quality is horrible. So I started messing with the settings, lo-and-behold I find info about h.264 and how fantastic it is. Well, I say to myself, let's try that on for size.
What? No h.264 option in Ekiga. That's weird, especially considering they have supported it since 3.0 and I'm using the version in the Karmic repos, 3.2.5. Not free, ok, I'll just install it then, right?
x264 package didn't do it. A little ffmpeg install as well. libavcodec52, which uninstalled a buttload of other stuff also. Even dl'd and installed libopal3.6.4-plugins-non-free. All to no option for the h.264 codec option in Ekiga.
Oddly enough, the libopal non free plugins DID add the iLBC audio codec into Ekiga as on option. Why not h.264? What am I missing? Also, why is this SO difficult? There is not any decent documentation that I can find out there that goes over this. Google has failed me.
All I have in the video codecs is theora and h261.
So I know that Mac OSX post-production and Linux post-production are very different things. I'm hoping to give editing on Linux with Cinelerra a try and I'm wondering, what in FFMpeg's array of codec and container support is widely understood to be the best combination for editing video. I'm looking for the most lossless option so H.264 won't do. Is there a good Apple Intermediate Codec or Apple Pro Res equivalent? Is AVCHD the answer?
is there a legit (read: legal) way of getting the Fraunhofer MP3 codec in Ubuntu? Any commercial or freeware compressors that use it, or maybe even a standalone codec installer (like Radium for Windows, though IIRC that one's not exactly legal)?If not, is there perhaps a better codec for compressing MP3s? I've used Fraunhofer at 256Kbps CBR for years and never had a problem.
I am running Ubuntu 9.10 64 bit version. Want to capture a short audio stream from a DVD.
VLC has a command "Open Capture Device" that gives the user options for ripping (I think) the audio tracks. I made the relevant choices and got this message:
Streaming / Transcoding failed: It seems your FFMPEG (libavcodec) installation lacks the following encoder: MPEG AAC Audio. If you don't know how to fix this, ask for support from your distribution.
This is not an error inside VLC media player. Do not contact the VideoLAN project about this issue.
What is to be done?
I went to synaptic package manager and I have FFMPEG, libavcodec, and AAC installed.
Here's a plugin that lets VLC from wheezy-backports play h.265 content. I packaged it for the MEPIS 12 repo, but it should work on wheezy. [URL] ....
Or create your own packages against Debian's stock VLC 2.0.3, that might work.
You need to install the libde265 package first, then the plugin package.
You can use the static ffmpeg binaries from ffmpeg.org to encode to that h.265 (hevc) codec, but it will take quite a while.
Code: Select all./ffmpeg -i [inputfile] -vcodec hevc [outputfile] Try mp4 or mkv for the output file extension.
The good news is that it will create video of the same quality that is half the size of a comparable h.264 file. This is handy for putting movies on space-limited storage, such as on a phone or tablet. I do know that MX Player on Android can play h.265 video.
This may be a minor irritation, but I'm puzzled by it nonetheless. I'm ripping my CDs to MP3 using Sound Juicer with the lamemp3enc plugin and following gstreamer pipeline.
All was great a couple days ago, but suddenly the files' audio properties show 48 kbps bitrate when it should be well over 200. I even changed the pipeline to use "target=1 bitrate=256" with the same result. The files are being encoded as expected and that is reflected in the file size and the Statistics view in VLC.
I have two computers running Ubuntu 10.10. One has all the latest updates, but the other has not been updated in several days. This problem is happening on the former, but not the latter. gst-inspect lamemp3enc shows both computers have the same version of the encoder.
I thought I'd put it out to the forum here before submitting a bug in launchpad.
I just bought a new hard drive and decided to upgrade to 10.4.2 (I was running 9.10).I started doing the normal customizations that come with a new install. I tried to play an mp3 file and the sstem offered to install the new gstreamer codecs(ffmpeg,fluendo-mp3, and plugins-ugly).I click on install and it is stuck at preparing libavformat52.The terminal details state:Code: dpkg: warning: files list file for package 'libavformat52' missing, assuming package has no files currently installed.Preparing to replace libavformat52 4:0.5.1-1ubuntu1 (using./libavformat52_4%3a0.5.1-1ubuntu1_i386.deb).Unpacking replacement libavformat52.It was stuck so long and I could not kill it (did not know what to kill), so I rebooted once before this message.
