mirror of https://github.com/FFmpeg/FFmpeg.git
Fabrice Bellard
24 years ago
parent
de6d9b6404
commit
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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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@ -0,0 +1,73 @@ |
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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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@ -0,0 +1,46 @@ |
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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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@ -0,0 +1,63 @@ |
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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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@ -0,0 +1,181 @@ |
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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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|
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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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@ -0,0 +1,261 @@ |
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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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AudioChannels 1 |
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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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# Bitrate for the video stream. |
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VideoBitRate 64 |
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# Number of frames per second |
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VideoFrameRate 3 |
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# Size of the video frame : WxH (default: 160x128) |
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# W : width, H : height |
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# The following abbreviation are defined : sqcif, qcif, cif, 4cif |
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VideoSize 160x128 |
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# transmit only intra frames (useful for low bitrates) |
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VideoIntraOnly |
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# If non intra only, an intra frame is transmitted every VideoGopSize |
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# frames Video synchronization can only begin at an I frames. |
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#VideoGopSize 12 |
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# Suppress audio |
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#NoAudio |
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# Suppress video |
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#NoVideo |
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</Stream> |
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# second mpeg stream with high frame rate |
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<Stream test2.mpg> |
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Feed feed1.ffm |
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Format mpegvideo |
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VideoBitRate 128 |
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VideoFrameRate 25 |
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#VideoSize 352x240 |
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VideoGopSize 25 |
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</Stream> |
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################################################################## |
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# A stream coming from a file : you only need to set the input |
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# filename and optionnally a new format. Supported conversions: |
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# avi -> asf |
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# |
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<Stream file.asf> |
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#File "/tmp/file.avi" |
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File "tmp/file.avi" |
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# avi must be converted to asf to be streamed |
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Format asf |
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</Stream> |
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# another file streaming |
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<Stream file.mp3> |
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File "tmp/file.mp3" |
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</Stream> |
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################################################################## |
||||
# 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