I'm not well enough versed in the differences between ffmpeg & mencoder to know which one is better.

yt2mp3(){ for j in `seq 1 301`;do i=`curl -s gdata.youtube.com/feeds/api/users/$1/uploads\?start-index=$j\&max-results=1|grep -o "watch[^&]*"`;ffmpeg -i `wget youtube.com/$i -qO-|grep -o 'url_map"[^,]*'|sed -n '1{s_.*|__;s_\\\__g;p}'` -vn -ab 128k "`youtube-dl -e ${i#*=}`.mp3";done;} squeezed the monster (and nifty ☺) command from 7776 from 531 characters to 284 characters, but I don't see a way to get it down to 255. This is definitely a kludge!

Gives stereo, 16bit, 44.1kHz (default in Ubuntu/Medibuntu ffmpeg). -aq 2 = 220-250kbit/s VBR, lower number is better quality. 2 or 3 should be good for most people. If you want the best mp3 q you should remove -aq and use -ab 320k to get 320kbit/s, but that is probably overkill for most .flv videos.

ffmpeg [source specification if needed] -i $src -an -vcodec libx264 -coder 0 -threads 0 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -level 13 -g 15 -crf 25 -s 320x224 -aspect 16:9 -r 15 -b 186000 -vb 186000 -minrate 176700 -maxrate 195300 -bt 9300 -bufsize 262500 -muxrate 195300 -vglobal 1 -f rtp rtp://$dstIP:$dstVideoPort1 -vn -acodec libfaac -async 2 -flags +global_header -ac 1 -ar 44100 -ab 64000 -f rtp rtp://$dstIP:$dstAudioPort1 -newaudio -an -vcodec libx264 -coder 0 -threads 0 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -level 13 -g 20 -crf 25 -s 320x224 -aspect 16:9 -r 20 -b 286000 -vb 286000 -minrate 271700 -maxrate 300300 -bt 14300 -bufsize 367500 -muxrate 300300 -vglobal 1 -f rtp rtp://$dstIP:$dstVideoPort2 -newvideo -vn -acodec libfaac -async 2 -flags +global_header -ac 1 -ar 44100 -ab 64000 -f rtp rtp://$dstIP:$dstAudioPort2 -newaudio -an -vcodec libx264 -coder 0 -threads 0 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -level 30 -g 25 -crf 25 -s 480x336 -aspect 16:9 -r 25 -b 386000 -vb 386000 -minrate 366700 -maxrate 405300 -bt 19300 -bufsize 472500 -muxrate 405300 -vglobal 1 -f rtp rtp://$dstIP:$dstVideoPort3 -newvideo -vn -acodec libfaac -async 2 -flags +global_header -ac 2 -ar 44100 -ab 64000 -f rtp rtp://$dstIP:$dstAudioPort3 -newaudio -an -vcodec libx264 -coder 0 -threads 0 -flags +loop -cmp +chroma -partitions +parti4x4+partp8x8+partb8x8 -subq 5 -trellis 1 -refs 1 -coder 0 -me_range 16 -keyint_min 25 -sc_threshold 40 -i_qfactor 0.71 -rc_eq 'blurCplx^(1-qComp)' -qcomp 0.6 -qmin 10 -qmax 51 -level 30 -g 25 -crf 25 -s 480x336 -aspect 16:9 -r 25 -b 686000 -vb 686000 -minrate 651700 -maxrate 720300 -bt 34300 -bufsize 787500 -muxrate 720300 -vglobal 1 -f rtp rtp://$dstIP:$dstVideoPort4 -newvideo -vn -acodec libfaac -async 2 -flags +global_header -ac 2 -ar 44100 -ab 64000 -f rtp rtp://$dstIP:$dstAudioPort4 -newaudio Show Sample Output

sxga ==> 1280x1024 vga ==> 800x600 ------------------------------------------------ xwininfo -root | grep geometry ------------------------------------------------

A simple command to extract audio from flv/mp4 video file. Just change extentions...

Now we can capture only a specific window (we have to chose by clicking on it) ffmpeg complains about "Frame size must be a multiple of 2" so we calculate the upper even number with (g)awk trickery. We remove the grep, we are already using (g)awk here ....why losing time with grep !!! ;) Show Sample Output

-i sets the source file -r sets the output frame rate, set it to the same frame rate as the input to output each frame -f sets the output format, trough it might be guessed by the extension

-i sets the source file -f and -acodec both set the output to be raw audio, PCM signed 16-bit little endian

This *does not change the video encoding*, so it's fast (almost purely I/O-bound) and results in a file of nearly the same size. However, OSX (and possibly other programs) will more easily play/seek the file when wrapped as MOV. For example, you can QuickLook the resulting file. This basically does the same as the commercial ClipWrap program, except using the free program ffmpeg. Show Sample Output

This encodes it in ogg format. Does on-the-fly encoding of the incoming stream. Great for radio streams as they're often flv format.

Before you use this command you want to replace everything after the "https:" with the url of the video which you want to download. This string and it's switches will use "youtube-dl" to download the Youtube url into the directory/folder where it is called from. It will output the video using the same name as Youtube uses.

This will dump a raw BGRA pixel stream and WAV which must then be converted to video: ffmpeg -f rawvideo -c:v rawvideo -s 1280x720 -r 12 -pix_fmt bgra -i "${i%.*}".bgra -c:v libx264 -preset veryslow -qp 0 -movflags +faststart -i "${i%.*}".wav -c:a libfdk_aac -b:a 384k "${i%.*}".mp4 ; rm "${i%.*}".bgra "${i%.*}".wav Our example generates an x264/720p/12fps/AAC best-quality MP4. To get dump-gnash, first install the build-dependencies for gnash (this step is OS-specific). Then: git clone http://git.savannah.gnu.org/r/gnash.git ; cd gnash ; ./autogen.sh ; ./configure --enable-renderer=agg --enable-gui=dump --disable-menus --enable-media=ffmpeg --disable-jemalloc ; make

Works on *.mp4 as well. Show Sample Output

to view on another box: nc <server address> <port> | ffplay - use -r to adjust FPS and -q to adjust compression. use on trusted network only as nc is unencrypted.

Uses ffmpeg to convert all that annoying .FLAC files to MP3 files keeping all the Artist's information in them. There's not much more to it. Show Sample Output

It loops through all files in current directory that have flac extension and converts them to mp3 files with bitrate of 320kpbs using ffmpeg and default codec.

Replace video and audio extension according to your needs

ffmpeg and avconv didnt have this feature. I use this command to hardsubs mkv files to mp4

I use this command to stream live video to facebook from a vps. you need first convert the file to flv i use mpv to encode with hardsubs a file. and then run ffmpeg to stream the file.

ffprobe doesn't throw an error and was actually made to do exactly that. Usually ffprobe is located in the same folder as ffmpeg. https://ffmpeg.org/ffprobe.html#Description

Converts an OGG file to MP3 at desired quality, preserving the id3v2 tags.