Ubuntu Multimedia :: FFmpeg Unsupported Codec For Output Stream
Feb 26, 2010
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'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.
I have a video - taken on my mobile phone. In totem, I get sound and video: lovely. However, in Openshot I don't get sounds, and in WinFF it tells me "Unsupported codec (id=7372" for the audio stream. Why, when both are using FFMpeg do the installed codecs work differently in different programs?
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,
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
For X-Mas I got myself a nice little nettop running off an ION motherboard. how to enable HDMI audio output, but got it to work. I've realized that when I'm in XBMC (which is admittedly the only place I watch movies) certain sounds like gunshots will be extremely louder than something like dialog. My friend believes that I should mess around with the channels (to fake the center channel from the left and right) I don't want to threaten Linux, how lightweight I can make it which is critical to an HTPC, but considering XBMC just got a Netflix addon
I'm still trying to find out if my coby mp3 player will actually play mtv video files as is advertised.
ffmpeg -formats does list mtv but the only command I really ever used was one to convert a vid to an mp3 so I tried Code: ffmpeg -i test.mp4 -acodec copy output.mtv it returns Code: Unable to find a suitable output format for 'output.mtv' I can't find any mtv files online for purchase or free for that matter, so I know this is all pretty obscure but shouldn't there be a way to convert them since ffmpeg lists mtv format?
I am using Ubuntu 10.04 and would like to use WebCamStudio with Skype 2.1 beta for ubuntu 32 bit. I though skype displays WebCamStudio as one of the webcams its not able to stream output of WebCamStudio to other person...
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.
Is it possible to output one playback stream to multiple devices simultaneously with the current PulseAudio / Phonon setup? The PulseAudio mixer only has radio buttons to choose one device per playback stream. I believe the hardware is capable of this, since I remember doing that before we had PulseAudio. How can I duplicate an audio stream?
Here's one application scenario: I am travelling with my family, all crammed in small hotel room. My wife and me want to watch a movie on my laptop without waking up our kids. I just happen to have one analogue headphone available and one wireless USB headset with me. (Of course, the low tech solution is to bring an 3,5mm Y-cable to attach two analogue headsets, but I would really love to use the USB headset together with the analogue one.)
Another similar thing that bugs me is that my laptop's built-in speakers now always seem dead when an analogue headphone is plugged in. This is mostly what one wants, and before PulseAudio, one had to manually switch them off which was generally annoying. However, the downside is for example with notifications.
For example, before PulseAudio, I could configure Skype to always ring over the laptop's built-in speakers, regardless of whether the analogue headphones were plugged in. This is no longer possible, since PulseAudio does not distinguish between built-in speakers and built-in analogue port any more, while old Alsa did. So in my office, where some analogue headphones are plugged into the docking station, I never hear Skype ringing if I don't wear the headphones.
Code: $ffmpeg -i /data/sumeet/video/hollywood/you don't know jack/You.Dont.Know.Jack.2010.DVDRipwww.theevolution.org_by_digoloko.rmvb FFmpeg version git-1dbd813, Copyright (c) 2000-2010 the FFmpeg developers built on Oct 1 2010 19:28:12 with gcc 4.4.3
[Code].....
At least one output file must be specified Actually the video has embedded subtitles. I thing that is stream 0.2 in real media container. How can i pull that data out ?
I have two machines on a local area network (xp box and xubuntu box) and I want audio from both machines to be played from the same set of speakers. The problem is, the xubuntu machine doesn't have any sound output. There is no onboard sound card and all expansion slots are pci-x, so short of buying a pci-x sound card my only option for playing sound is to route audio through LAN to my xp computer.
I already have a program that will let me play music on one computer from another's speakers, but I am trying to set up a stream so that games and internet sound can be heard. Is it possible for me to do this?
I'm working on some scheduled task script files to keep nightly backups of some of our database information in place, and it's a bit annoying when they blow up. I know how to redirect stdout and stderr to a flat file I can view when I come in, and I know that 2>&1 maps them both to the same file (whatever was named in 1). However, I'm running into some cron-time situations where it's easier to have the two streams together, and other cron-time situations where it's easier to have them separated. I can't really tell which is going to happen; is there some way I could create both kinds of output file for my scripts, so that I've got a std_err only file and an interleaved std_out/std_err file?
Note: I've looked at the 'tee' command, but I don't think it will work for what I'm after. 'tee' appears to only work with stdout; I'm trying to work with stderr.
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'm helping a friend set up his new pc and we're having a lot of trouble getting his second monitor to work.He has a GeForce GTS250 video card which has a HDMI output port as well as VGA. His main monitor is connected via the VGA and works fine. The second monitor is a Soniq QV320H TV which he wants to connect via HDMI. The nvidia settings configuration shows both monitors (set up as separate x-screens) but the only output on the Soniq is a black screen which flashes "unsupported" about every 2 seconds. Currently the resolution is set to auto (1280x1024) but we've also tried lower resolutions (down to 800x600) with no success.
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.
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.
I've a HP Mini 210 and sometimes it becomes difficult to watch HD Movies. Some videos in 720p I can watch perfectly, while others start crashing after 1 minute..They are mostly in MKV, which from what I understand, it's just a container for the video/audio and can be in several formats. I know that VLC has the Transcode feature, and I'd like to use that to transcode the videos to a video Codec that requires the least CPU processing.
