mirror of https://git.ffmpeg.org/ffmpeg.git
182 lines
6.4 KiB
Plaintext
182 lines
6.4 KiB
Plaintext
*************** FFMPEG soft VCR documentation *****************
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0) Introduction
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---------------
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FFmpeg is a very fast video and audio encoder. It can grab from
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files or from a live audio/video source.
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The command line interface is designed to be intuitive, in the sense
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that ffmpeg tries to figure out all the paramters, when
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possible. You have usually to give only the target bitrate you want.
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FFmpeg can also convert from any sample rate to any other, and
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resize video on the fly with a high quality polyphase filter.
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1) Video and Audio grabbing
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---------------------------
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* ffmpeg can use a video4linux compatible video source and any Open
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Sound System audio source:
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ffmpeg /tmp/out.mpg
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Note that you must activate the right video source and channel
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before launching ffmpeg. You can use any TV viewer such as xawtv by
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Gerd Knorr which I find very good. You must also set correctly the
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audio recording levels with a standard mixer.
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2) Video and Audio file format convertion
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-----------------------------------------
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* ffmpeg can use any supported file format and protocol as input :
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examples:
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ffmpeg -i /tmp/test%d.Y /tmp/out.mpg
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If will use the files:
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/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
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/tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...
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The Y files use twice the resolution of the U and V files. They are
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raw files, without header. They can be generated by all decent video
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decoders. You must specify the size of the image with the '-s' option
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if ffmpeg cannot guess it.
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* You can set several input files and output files:
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ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg
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Convert the audio file a.wav and the raw yuv video file a.yuv to mpeg file a.mpg
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* you can also do audio and video convertions at the same time:
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ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2
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Convert the sample rate of a.wav to 22050 Hz and encode it to MPEG audio.
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* you can encode to several formats at the same time and define a
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mapping from input stream to output streams:
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ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0
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convert a.wav to a.mp2 at 64 kbits and b.mp2 at 128 kbits. '-map
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file:index' specify which input stream is used for each output
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stream, in the order of the definition of output streams.
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NOTE: to see the supported input formats, use 'ffmpeg -formats'.
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2) Invocation
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-------------
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* The generic syntax is :
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ffmpeg [[options][-i input_file]]... {[options] output_file}...
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If no input file is given, audio/video grabbing is done.
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As a general rule, options are applied to the next specified
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file. For example, if you give the '-b 64' option, it sets the video
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bitrate of the next file. Format option may be needed for raw input
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files.
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By default, ffmpeg tries to convert as losslessly as possible: it
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uses the same audio and video parameter fors the outputs as the one
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specified for the inputs.
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* Main options are:
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-h show help
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-formats show available formats, codecs and protocols
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-L print the LICENSE
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-i filename input file name
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-y overwrite output files
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-t duration set recording time in seconds
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-f format set encoding format [guessed]
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-title string set the title
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-author string set the author
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-copyright string set the copyright
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-comment string set the comment
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* Video Options are:
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-s size set frame size [160x128]
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-r fps set frame rate [25]
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-b bitrate set the video bitrate in kbit/s [200]
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-vn disable video recording [no]
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* Audio Options are:
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-ar freq set the audio sampling freq [44100]
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-ab bitrate set the audio bitrate in kbit/s [64]
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-ac channels set the number of audio channels [1]
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-an disable audio recording [no]
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Advanced options are:
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-map file:stream set input stream mapping
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-g gop_size set the group of picture size [12]
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-intra use only intra frames [no]
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-qscale q use fixed video quantiser scale (VBR)
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-c comment set the comment string
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-vd device set video4linux device name [/dev/video]
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-vcodec codec force audio codec
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-me method set motion estimation method
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-ad device set audio device name [/dev/dsp]
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-acodec codec force audio codec
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The output file can be "-" to output to a pipe. This is only possible
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with mpeg1 and h263 formats.
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3) Protocols
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ffmpeg handles also many protocols specified with the URL syntax.
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Use 'ffmpeg -formats' to have a list of the supported protocols.
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The protocol 'http:' is currently used only to communicate with
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ffserver (see the ffserver documentation). When ffmpeg will be a
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video player it will also be used for streaming :-)
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4) File formats and codecs
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--------------------------
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Use 'ffmpeg -formats' to have a list of the supported output
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formats. Only some formats are handled as input, but it will improve
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in the next versions.
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5) Tips
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-------
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- For streaming at very low bit rate application, use a low frame rate
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and a small gop size. This is especially true for real video where
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the Linux player does not seem to be very fast, so it can miss
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frames. An example is:
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ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm
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- The parameter 'q' which is displayed while encoding is the current
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quantizer. The value of 1 indicates that a very good quality could
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be achieved. The value of 31 indicates the worst quality. If q=31
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too often, it means that the encoder cannot compress enough to meet
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your bit rate. You must either increase the bit rate, decrease the
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frame rate or decrease the frame size.
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- If your computer is not fast enough, you can speed up the
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compression at the expense of the compression ratio. You can use
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'-me zero' to speed up motion estimation, and '-intra' to disable
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completly motion estimation (you have only I frames, which means it
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is about as good as JPEG compression).
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- To have very low bitrates in audio, reduce the sampling frequency
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(down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).
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- To have a constant quality (but a variable bitrate), use the option
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'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
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quality).
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- When converting video files, you can use the '-sameq' option which
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uses in the encoder the same quality factor than in the decoder. It
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allows to be almost lossless in encoding.
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