Ubuntu Multimedia :: Ffmpeg To Resample Audio But Leave Video Unchanged?
Mar 1, 2010
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:
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.
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.
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:
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 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 :
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.
I'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-
Anybody 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.
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).
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?
I spent about a half hour wrestling with different website tutorials about how to convert a file with ffmpeg and figuring out how to get all the video quality options right. Then I discovered you can just use the -sameq option and it figures it all out for you if you don't want to change the vid quality but just want it in another format. Thought I'd leave this on the site in case anyone else finds himself in the same boat.
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 have a video file in which the audio runs faster than the video, so they quickly go out of sync. The way to fix it would be to separate the audio and video streams, speed up the video (the audio is FINE, it's the video that's wrong), and then recombining them. What is the easiest way for doing that?
Is there an application that anyone knows about that I can use to convert either an .flv or .ogg file that contains both audio and video to just an audio .ogg file (preferably vorbis+theora) without audacity? I'm fairly certain audacity could accomplish this but it seems like overkill for what I'm trying to do and the computer I'm trying to use does not run it so well.
::EDIT:: I should also mention that I've tried looking on google. I did find downloadhelper extension for firefox which uses ffmpeg to convert the files but I don't see any obvious way to strip the video.
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've got a 1920x1080 video I've edited and rendered with Cinelerra, and I'm trying to use ffmpeg to transcode it to something smaller. However, when I use a command like this, for instance:
Code:
I inevitably get some weird green band at the bottom of the frame in the converted video. I know that there's some weird pixel stretching going on here, because the NTSC standard for 16:9 is 720x480 with rectangular pixels, and the 1080 version has square pixels, so I'm guessing the green band is an artifact of that process?
mplayer Naruto/Naruto Shippuuden - 185 v2.mp4 MPlayer 1.0rc4-4.4.5 (C) 2000-2010 MPlayer Team mplayer: could not connect to socket mplayer: No such file or directory
Failed to open LIRC support. You will not be able to use your remote control.
Playing Naruto/Naruto Shippuuden - 185.mp4. libavformat file format detected. [lavf] stream 0: video (h264), -vid 0 [lavf] stream 1: audio (aac), -aid 0, -alang und
I have ubuntu 9.10. Video is working , but sound not working. When I type, sudo aplay -l, a message **** List of Playback Hardware Devices **** shown. But there is no list. My computer is Intel 810 onboard sound AC '97 (codec ) system. More information in this page: [URL]
I've recorded some video in the field, i.e. outside, windy, noisy, etc. I would like to strip out the audio, clean it up in Audacity and then repatriate it back to the video. Video is in .mp4 format.
My work just installed a new surveillance system and it uses .box video files and .idx audio to go along with it. They have a windows program to view them but I would love to keep on using my Ubuntu laptop.