merge
Originally committed as revision 6 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Fabrice Bellard
parent
de6d9b6404
commit
85f07f223d
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# Main ffmpeg Makefile
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# (c) 2000, 2001 Gerard Lantau
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#
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include config.mk
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CFLAGS= -O2 -Wall -g -I./libavcodec -I./libav
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LDFLAGS= -g
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ifdef CONFIG_GPROF
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CFLAGS+=-p
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LDFLAGS+=-p
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endif
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PROG= ffmpeg ffserver
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all: lib $(PROG)
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lib:
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make -C libavcodec all
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make -C libav all
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ffmpeg: ffmpeg.o libav/libav.a libavcodec/libavcodec.a
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gcc $(LDFLAGS) -o $@ $^ -lm
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ffserver: ffserver.o libav/libav.a libavcodec/libavcodec.a
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gcc $(LDFLAGS) -o $@ $^ -lm
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%.o: %.c
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gcc $(CFLAGS) -c -o $@ $<
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install: all
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install -s -m 755 $(PROG) $(PREFIX)/bin
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clean:
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make -C libavcodec clean
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make -C libav clean
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rm -f *.o *~ gmon.out TAGS $(PROG)
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distclean: clean
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rm -f Rules.mk config.h
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TAGS:
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etags *.[ch] libav/*.[ch] libavcodec/*.[ch]
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FFmpeg - (c) 2000,2001 Gerard Lantau.
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1) Introduction
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---------------
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ffmpeg is a hyper fast realtime audio/video encoder, a streaming
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server and a generic audio and video file converter.
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It can grab from a standard Video4Linux video source and convert it
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into several file formats based on DCT/motion compensation
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encoding. Sound is compressed in MPEG audio layer 2 or using an AC3
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compatible stream.
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What makes ffmpeg interesting ?
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- Innovative streaming technology : multiformat, real time encoding,
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simple configuration.
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- Simple and efficient video encoder: outputs MPEG1, H263, Real
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Video(tm), MPEG4, DIVX and MJPEG compatible bitstreams using the
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same encoder core.
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- Real time encoding (25 fps in 352x288 on a K6 500) using the
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video4linux API.
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- Generates I and P frames, which means it is far better than a MJPEG
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encoder.
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- Hyper fast MPEG audio layer 2 compression (50 times faster than
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realtime on a K6 500).
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- Hyper fast AC3 compatible encoder.
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- simple and very small portable C source code, easy to understand and
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to modify. It be may the smallest decent MPEG encoder :-)
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- optional non real time higher quality encoding (different motion
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estimators available).
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- Audio and Video decoders are in development.
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ffmpeg is made of two programs:
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* ffmpeg: soft VCR which encodes in real time to several formats. It
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can also encode from any supported input file format to any input
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supported format.
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* ffserver: high performance live broadcast streaming server based on
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the ffmpeg core encoders.
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2) Documentation
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----------------
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* Read doc/ffmpeg.txt and doc/ffserver.txt to learn the basic features.
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* Read doc/TODO to know what are the know bugs and missing features.
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* Read doc/README.dev if you want to contribute or use the codec or
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format libraries.
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3) Licensing:
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------------
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* See the file COPYING. ffmpeg and the associated library are licensed
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under the GNU General Public License. I may change the license of
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libavcodec and libav to LGPL if many people ask it (and if they
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submit good patches!).
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* This code should be patent free since it is very simple. I took care
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to use the same video encoder/decoder core for all formats to show
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that they really ARE THE SAME except for the encoding huffman codes.
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Gerard Lantau (glantau@yahoo.fr).
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Technical notes:
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---------------
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Video:
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-----
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- The decision intra/predicted macroblock is the algorithm suggested
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by the mpeg 1 specification.
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- only Huffman based H263 is supported, mainly because of patent
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issues.
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- MPEG4 is supported, as an extension of the H263 encoder. MPEG4 DC
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prediction is used, but not AC prediction. Specific VLC are used for
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intra pictures. The output format is compatible with Open DIVX
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version 47.
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- MJPEG is supported, but in the current version the huffman tables
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are not optimized. It could be interesting to add this feature for
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the flash format.
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- To increase speed, only motion vectors (0,0) are tested for real
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time compression. NEW: now motion compensation is done with several
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methods : none, full, log, and phods. The code is mmx/sse optimized.
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- In high quality mode, full search is used for motion
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vectors. Currently, only fcode = 1 is used for both H263/MPEG1. Half
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pel vectors are used.
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I also plan to improve the bitrate control which is too simplistic.
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Audio:
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-----
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- I rewrote the mpeg audio layer 2 compatible encoder from scratch. It
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is one of the simplest encoder you can imagine (800 lines of C code
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!). It is also one of the fastest because of its simplicity. There
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are still some problems of overflow. A minimal psycho acoustic model
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could be added. Currently, stereo is supported, but not joint
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stereo.
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- I rewrote the AC3 audio encoder from scratch. It is fairly naive,
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but the result are quiet interesting at 64 kbit/s. It includes
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extensions for low sampling rates used in some Internet
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formats. Differential and coupled stereo is not handled. Stereo
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channels are simply handled as two mono channels.
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ffmpeg TODO list:
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----------------
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(in approximate decreasing priority order)
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Planned in next release:
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(DONE) - apply header fixes
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(DONE) - mpeg audio decoder.
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(DONE) - fix decode/encode codec string.
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(DONE) - fix EINTR error if VIDIOCSYNC.
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(DONE) - add CONFIG system.
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(DONE) - merge mplayer mmx accel.
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(DONE) - fix emms bug.
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(DONE) - add I263 handling
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(DONE) - add RV10 decoding.
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(DONE) - add true pgm support.
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(DONE) - msmpeg4 0x18 fix.
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- add qscale out.
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- add format autodetect with content (for example to distinguish
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mpegvideo/mpegmux).
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- add external alloc for libavcodec (avifile request).
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- fix -sameq in grabbing
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- find a solution to clear feed1.ffm if format change.
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- new grab architecture : use avformat instead of audio: and video:
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protocol.
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- correct PTS handling to sync audio and video.
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- fix 0 size picture in AVIs = skip picture
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BUGS:
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- fix audio/video synchro (including real player synchro bugs)
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- Improve the bit rate control for video codecs.
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- see ov511.o YUV problem (420 instead of 420P).
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- fix file caching pb in windows (add correct headers)
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- add low pass filter to suppress noise coming from cheap TV cards.
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- test/debug audio in flash format
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- sort out ASF streaming pbs.
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- Improve psycho acoustic model for AC3 & mpeg audio.
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FEATURES:
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- add MPEG4 in mpegmux support.
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- add RTP / multicast layer.
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- demux streams for CCTV : N streams in one stream. Add option to
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generate multiple streams.
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- add disconnect user option in stat.html.
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- deny & allow + password in ffserver.
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- graphical user interface.
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- animated gif as output format
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*************** 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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# Port on which the server is listening. You must select a different
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# port from your standard http web server if it is running on the same
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# computer.
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Port 8090
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# Address on which the server is bound. Only useful if you have
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# several network interfaces.
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BindAddress 0.0.0.0
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# Number of simultaneous requests that can be handled. Since FFServer
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# is very fast, this limit is determined mainly by your Internet
|
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# connection speed.
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MaxClients 1000
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# Access Log file (uses standard Apache log file format)
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# '-' is the standard output
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CustomLog -
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##################################################################
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# Definition of the live feeds. Each live feed contains one video
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# and/or audio sequence coming from an ffmpeg encoder or another
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# ffserver. This sequence may be encoded simultaneously with several
|
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# codecs at several resolutions.
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<Feed feed1.ffm>
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# You must use 'ffmpeg' to send a live feed to ffserver. In this
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# example, you can type:
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#
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# ffmpeg http://localhost:8090/feed1.ffm
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# ffserver can also do time shifting. It means that it can stream any
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# previously recorded live stream. The request should contain:
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# "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
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# a path where the feed is stored on disk. You also specify the
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# maximum size of the feed (100M bytes here). Default:
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# File=/tmp/feed_name.ffm FileMaxSize=5M
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File /tmp/feed1.ffm
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FileMaxSize 50M
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</Feed>
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##################################################################
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# Now you can define each stream which will be generated from the
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# original audio and video stream. Each format has a filename (here
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# 'test128.mpg'). FFServer will send this stream when answering a
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# request containing this filename.
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<Stream test1.mpg>
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# coming from live feed 'feed1'
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Feed feed1.ffm
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# Format of the stream : you can choose among:
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# mpeg : MPEG1 multiplexed video and audio
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# mpegvideo : only MPEG1 video
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# mp2 : MPEG audio layer 2
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# mp3 : MPEG audio layer 3 (currently sent as layer 2)
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# rm : Real Networks compatible stream. Multiplexed audio and video.
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# ra : Real Networks compatible stream. Audio only.
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# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
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# jpeg : Generate a single JPEG image.
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# asf : ASF compatible stream (Windows Media Player format). Not finished yet.
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# swf : Macromedia flash(tm) compatible stream
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# avi : AVI format (open divx video, mpeg audio sound)
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# master : special ffmpeg stream used to duplicate a server
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Format mpeg
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# Bitrate for the audio stream. Codecs usually support only a few different bitrates.
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AudioBitRate 32
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# Number of audio channels : 1 = mono, 2 = stereo
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||||
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||||
AudioChannels 1
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||||
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||||
# Sampling frequency for audio. When using low bitrates, you should
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||||
# lower this frequency to 22050 or 11025. The supported frequencies
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||||
# depend on the selected audio codec.
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AudioSampleRate 44100
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||||
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||||
# Bitrate for the video stream.
|
||||
VideoBitRate 64
|
||||
|
||||
# Number of frames per second
|
||||
VideoFrameRate 3
|
||||
|
||||
# Size of the video frame : WxH (default: 160x128)
|
||||
# W : width, H : height
|
||||
# The following abbreviation are defined : sqcif, qcif, cif, 4cif
|
||||
VideoSize 160x128
|
||||
|
||||
# transmit only intra frames (useful for low bitrates)
|
||||
VideoIntraOnly
|
||||
|
||||
# If non intra only, an intra frame is transmitted every VideoGopSize
|
||||
# frames Video synchronization can only begin at an I frames.
|
||||
#VideoGopSize 12
|
||||
|
||||
# Suppress audio
|
||||
#NoAudio
|
||||
|
||||
# Suppress video
|
||||
#NoVideo
|
||||
|
||||
</Stream>
|
||||
|
||||
# second mpeg stream with high frame rate
|
||||
|
||||
<Stream test2.mpg>
|
||||
Feed feed1.ffm
|
||||
Format mpegvideo
|
||||
VideoBitRate 128
|
||||
VideoFrameRate 25
|
||||
#VideoSize 352x240
|
||||
VideoGopSize 25
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# A stream coming from a file : you only need to set the input
|
||||
# filename and optionnally a new format. Supported conversions:
|
||||
# avi -> asf
|
||||
#
|
||||
|
||||
<Stream file.asf>
|
||||
|
||||
#File "/tmp/file.avi"
|
||||
File "tmp/file.avi"
|
||||
# avi must be converted to asf to be streamed
|
||||
Format asf
|
||||
|
||||
</Stream>
|
||||
|
||||
# another file streaming
|
||||
<Stream file.mp3>
|
||||
|
||||
File "tmp/file.mp3"
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Real with audio only at 32 kbits
|
||||
|
||||
<Stream test.ra>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format rm
|
||||
AudioBitRate 32
|
||||
NoVideo
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Real with audio and video at 64 kbits
|
||||
|
||||
<Stream test.rm>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format rm
|
||||
|
||||
AudioBitRate 32
|
||||
VideoBitRate 20
|
||||
VideoFrameRate 2
|
||||
VideoIntraOnly
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Mpeg audio layer 2 at 64 kbits.
|
||||
|
||||
<Stream test.mp2>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format mp2
|
||||
AudioBitRate 64
|
||||
AudioSampleRate 44100
|
||||
|
||||
</Stream>
|
||||
|
||||
<Stream test1.mp2>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format mp2
|
||||
AudioBitRate 32
|
||||
AudioSampleRate 16000
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Multipart JPEG
|
||||
|
||||
<Stream test.mjpg>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format mpjpeg
|
||||
|
||||
VideoFrameRate 2
|
||||
VideoIntraOnly
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Multipart JPEG
|
||||
|
||||
<Stream test.jpg>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format jpeg
|
||||
|
||||
# the parameters are choose here to take the same output as the
|
||||
# Multipart JPEG one.
|
||||
VideoFrameRate 2
|
||||
VideoIntraOnly
|
||||
#VideoSize 352x240
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Another stream : Flash
|
||||
|
||||
<Stream test.swf>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format swf
|
||||
|
||||
VideoFrameRate 2
|
||||
VideoIntraOnly
|
||||
|
||||
</Stream>
|
||||
|
||||
|
||||
##################################################################
|
||||
# Another stream : ASF compatible
|
||||
|
||||
<Stream test.asf>
|
||||
|
||||
Feed feed1.ffm
|
||||
Format asf
|
||||
|
||||
AudioBitRate 64
|
||||
AudioSampleRate 44100
|
||||
VideoFrameRate 2
|
||||
VideoIntraOnly
|
||||
|
||||
</Stream>
|
||||
|
||||
##################################################################
|
||||
# Special stream : server status
|
||||
|
||||
<Stream stat.html>
|
||||
|
||||
Format status
|
||||
|
||||
</Stream>
|
||||
File diff suppressed because it is too large
Load Diff
|
|
@ -0,0 +1,22 @@
|
|||
include ../config.mk
|
||||
CFLAGS= -O2 -Wall -g -I../libavcodec
|
||||
|
||||
OBJS= rm.o mpeg.o asf.o avienc.o jpegenc.o swf.o wav.o raw.o \
|
||||
avidec.o ffm.o \
|
||||
avio.o aviobuf.o utils.o \
|
||||
udp.o http.o file.o grab.o audio.o img.o
|
||||
|
||||
LIB= libav.a
|
||||
|
||||
all: $(LIB)
|
||||
|
||||
$(LIB): $(OBJS)
|
||||
rm -f $@
|
||||
$(AR) rcs $@ $(OBJS)
|
||||
|
||||
%.o: %.c
|
||||
$(CC) $(CFLAGS) -c -o $@ $<
|
||||
|
||||
clean:
|
||||
rm -f *.o *~ *.a
|
||||
|
||||
Loading…
Reference in New Issue