merge

Originally committed as revision 6 to svn://svn.ffmpeg.org/ffmpeg/trunk
24 years ago · 85f07f223d
--- a/Makefile
+++ b/Makefile
@@ -0,0 +1,42 @@
 # Main ffmpeg Makefile
 # (c) 2000, 2001 Gerard Lantau
 #
 include config.mk

 CFLAGS= -O2 -Wall -g -I./libavcodec -I./libav 
 LDFLAGS= -g 
 ifdef CONFIG_GPROF
 CFLAGS+=-p
 LDFLAGS+=-p
 endif

 PROG= ffmpeg ffserver

 all: lib $(PROG)

 lib:
 	make -C libavcodec all
 	make -C libav all

 ffmpeg: ffmpeg.o libav/libav.a libavcodec/libavcodec.a
 	gcc $(LDFLAGS) -o $@ $^ -lm

 ffserver: ffserver.o libav/libav.a libavcodec/libavcodec.a
 	gcc $(LDFLAGS) -o $@ $^ -lm

 %.o: %.c
 	gcc $(CFLAGS) -c -o $@ $< 

 install: all
 	install -s -m 755 $(PROG) $(PREFIX)/bin

 clean: 
 	make -C libavcodec clean
 	make -C libav clean
 	rm -f *.o *~ gmon.out TAGS $(PROG) 

 distclean: clean
 	rm -f Rules.mk config.h

 TAGS:
 	etags *.[ch] libav/*.[ch] libavcodec/*.[ch]
--- a/README
+++ b/README
@@ -0,0 +1,73 @@
 FFmpeg - (c) 2000,2001 Gerard Lantau.

 1) Introduction
 ---------------

 ffmpeg is a hyper fast realtime audio/video encoder, a streaming
 server and a generic audio and video file converter. 

 It can grab from a standard Video4Linux video source and convert it
 into several file formats based on DCT/motion compensation
 encoding. Sound is compressed in MPEG audio layer 2 or using an AC3
 compatible stream.

 What makes ffmpeg interesting ?

 - Innovative streaming technology : multiformat, real time encoding,
  simple configuration.

 - Simple and efficient video encoder: outputs MPEG1, H263, Real
  Video(tm), MPEG4, DIVX and MJPEG compatible bitstreams using the
  same encoder core.

 - Real time encoding (25 fps in 352x288 on a K6 500) using the
  video4linux API.

 - Generates I and P frames, which means it is far better than a MJPEG
  encoder.

 - Hyper fast MPEG audio layer 2 compression (50 times faster than
  realtime on a K6 500).

 - Hyper fast AC3 compatible encoder.

 - simple and very small portable C source code, easy to understand and
  to modify. It be may the smallest decent MPEG encoder :-)

 - optional non real time higher quality encoding (different motion
  estimators available).

 - Audio and Video decoders are in development.

 ffmpeg is made of two programs:

 * ffmpeg: soft VCR which encodes in real time to several formats. It
  can also encode from any supported input file format to any input
  supported format.

 * ffserver: high performance live broadcast streaming server based on
  the ffmpeg core encoders.

 2) Documentation
 ----------------

 * Read doc/ffmpeg.txt and doc/ffserver.txt to learn the basic features.

 * Read doc/TODO to know what are the know bugs and missing features.

 * Read doc/README.dev if you want to contribute or use the codec or
  format libraries.

 3) Licensing:
 ------------

 * See the file COPYING. ffmpeg and the associated library are licensed
  under the GNU General Public License. I may change the license of
  libavcodec and libav to LGPL if many people ask it (and if they
  submit good patches!).

 * This code should be patent free since it is very simple. I took care
  to use the same video encoder/decoder core for all formats to show
  that they really ARE THE SAME except for the encoding huffman codes.

 Gerard Lantau (glantau@yahoo.fr).
--- a/doc/README.tech
+++ b/doc/README.tech
@@ -0,0 +1,46 @@
 Technical notes:
 ---------------

 Video:
 -----

 - The decision intra/predicted macroblock is the algorithm suggested
  by the mpeg 1 specification.

 - only Huffman based H263 is supported, mainly because of patent
  issues.

 - MPEG4 is supported, as an extension of the H263 encoder. MPEG4 DC
  prediction is used, but not AC prediction. Specific VLC are used for
  intra pictures. The output format is compatible with Open DIVX
  version 47.

 - MJPEG is supported, but in the current version the huffman tables
  are not optimized. It could be interesting to add this feature for
  the flash format.

 - To increase speed, only motion vectors (0,0) are tested for real
  time compression. NEW: now motion compensation is done with several
  methods : none, full, log, and phods. The code is mmx/sse optimized.

 - In high quality mode, full search is used for motion
  vectors. Currently, only fcode = 1 is used for both H263/MPEG1. Half
  pel vectors are used.

 I also plan to improve the bitrate control which is too simplistic.

 Audio:
 -----

 - I rewrote the mpeg audio layer 2 compatible encoder from scratch. It
  is one of the simplest encoder you can imagine (800 lines of C code
  !). It is also one of the fastest because of its simplicity. There
  are still some problems of overflow. A minimal psycho acoustic model
  could be added. Currently, stereo is supported, but not joint
  stereo.

 - I rewrote the AC3 audio encoder from scratch. It is fairly naive,
  but the result are quiet interesting at 64 kbit/s. It includes
  extensions for low sampling rates used in some Internet
  formats. Differential and coupled stereo is not handled. Stereo
  channels are simply handled as two mono channels.
--- a/doc/TODO
+++ b/doc/TODO
@@ -0,0 +1,63 @@
 ffmpeg TODO list:
 ----------------

 (in approximate decreasing priority order)

 Planned in next release:

 (DONE) - apply header fixes
 (DONE) - mpeg audio decoder.
 (DONE) - fix decode/encode codec string.
 (DONE) - fix EINTR error if VIDIOCSYNC.
 (DONE) - add CONFIG system. 
 (DONE) - merge mplayer mmx accel.
 (DONE) - fix emms bug.
 (DONE) - add I263 handling
 (DONE) - add RV10 decoding.
 (DONE) - add true pgm support.
 (DONE) - msmpeg4 0x18 fix.
 - add qscale out.
 - add format autodetect with content (for example to distinguish
  mpegvideo/mpegmux).
 - add external alloc for libavcodec (avifile request).
 - fix -sameq in grabbing
 - find a solution to clear feed1.ffm if format change.
 - new grab architecture : use avformat instead of audio: and video:
  protocol.
 - correct PTS handling to sync audio and video.
 - fix 0 size picture in AVIs = skip picture

 BUGS:

 - fix audio/video synchro (including real player synchro bugs)

 - Improve the bit rate control for video codecs.

 - see ov511.o YUV problem (420 instead of 420P).

 - fix file caching pb in windows (add correct headers)

 - add low pass filter to suppress noise coming from cheap TV cards.

 - test/debug audio in flash format

 - sort out ASF streaming pbs.

 - Improve psycho acoustic model for AC3 & mpeg audio.

 FEATURES:

 - add MPEG4 in mpegmux support.

 - add RTP / multicast layer.

 - demux streams for CCTV : N streams in one stream. Add option to
  generate multiple streams.

 - add disconnect user option in stat.html.

 - deny & allow + password in ffserver.

 - graphical user interface.

 - animated gif as output format
--- a/doc/ffmpeg.txt
+++ b/doc/ffmpeg.txt
@@ -0,0 +1,181 @@
 *************** FFMPEG soft VCR documentation *****************

 0) Introduction
 ---------------

  FFmpeg is a very fast video and audio encoder. It can grab from
  files or from a live audio/video source. 
  
  The command line interface is designed to be intuitive, in the sense
  that ffmpeg tries to figure out all the paramters, when
  possible. You have usually to give only the target bitrate you want.

  FFmpeg can also convert from any sample rate to any other, and
  resize video on the fly with a high quality polyphase filter.

 1) Video and Audio grabbing
 ---------------------------

 * ffmpeg can use a video4linux compatible video source and any Open
  Sound System audio source:

  ffmpeg /tmp/out.mpg 

  Note that you must activate the right video source and channel
  before launching ffmpeg. You can use any TV viewer such as xawtv by
  Gerd Knorr which I find very good. You must also set correctly the
  audio recording levels with a standard mixer.

 2) Video and Audio file format convertion
 -----------------------------------------

 * ffmpeg can use any supported file format and protocol as input : 

 examples:

  ffmpeg -i /tmp/test%d.Y /tmp/out.mpg 

 If will use the files: 
 /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, 
 /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...

 The Y files use twice the resolution of the U and V files. They are
 raw files, without header. They can be generated by all decent video
 decoders. You must specify the size of the image with the '-s' option
 if ffmpeg cannot guess it.

 * You can set several input files and output files:

  ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg

 Convert the audio file a.wav and the raw yuv video file a.yuv  to mpeg file a.mpg

 * you can also do audio and video convertions at the same time:

  ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2

 Convert the sample rate of a.wav to 22050 Hz and encode it to MPEG audio.

 * you can encode to several formats at the same time and define a
  mapping from input stream to output streams:

  ffmpeg -i /tmp/a.wav -ab 64 /tmp/a.mp2 -ab 128 /tmp/b.mp2 -map 0:0 -map 0:0

  convert a.wav to a.mp2 at 64 kbits and b.mp2 at 128 kbits. '-map
  file:index' specify which input stream is used for each output
  stream, in the order of the definition of output streams.

 NOTE: to see the supported input formats, use 'ffmpeg -formats'.

 2) Invocation
 -------------

 * The generic syntax is :

  ffmpeg [[options][-i input_file]]... {[options] output_file}...

  If no input file is given, audio/video grabbing is done.

  As a general rule, options are applied to the next specified
  file. For example, if you give the '-b 64' option, it sets the video
  bitrate of the next file. Format option may be needed for raw input
  files.

  By default, ffmpeg tries to convert as losslessly as possible: it
  uses the same audio and video parameter fors the outputs as the one
  specified for the inputs.

 * Main options are:

 -h                  show help
 -formats            show available formats, codecs and protocols
 -L                  print the LICENSE
 -i filename         input file name
 -y                  overwrite output files
 -t duration         set recording time in seconds
 -f format           set encoding format                  [guessed]
 -title string       set the title
 -author string      set the author
 -copyright string   set the copyright
 -comment string     set the comment

 * Video Options are:

 -s size      set frame size                       [160x128]
 -r fps       set frame rate                       [25]
 -b bitrate   set the video bitrate in kbit/s      [200]
 -vn          disable video recording              [no]

 * Audio Options are:

 -ar freq     set the audio sampling freq          [44100]
 -ab bitrate  set the audio bitrate in kbit/s      [64]
 -ac channels set the number of audio channels     [1]
 -an          disable audio recording              [no]

 Advanced options are:

 -map file:stream    set input stream mapping
 -g gop_size         set the group of picture size        [12]
 -intra              use only intra frames                [no]
 -qscale q           use fixed video quantiser scale (VBR)
 -c comment          set the comment string
 -vd device          set video4linux device name          [/dev/video]
 -vcodec codec       force audio codec
 -me method          set motion estimation method
 -ad device          set audio device name                [/dev/dsp]
 -acodec codec       force audio codec

 The output file can be "-" to output to a pipe. This is only possible
 with mpeg1 and h263 formats. 

 3) Protocols

 ffmpeg handles also many protocols specified with the URL syntax.

 Use 'ffmpeg -formats' to have a list of the supported protocols.

 The protocol 'http:' is currently used only to communicate with
 ffserver (see the ffserver documentation). When ffmpeg will be a
 video player it will also be used for streaming :-)

 4) File formats and codecs
 --------------------------

 Use 'ffmpeg -formats' to have a list of the supported output
 formats. Only some formats are handled as input, but it will improve
 in the next versions.

 5) Tips
 -------

 - For streaming at very low bit rate application, use a low frame rate
  and a small gop size. This is especially true for real video where
  the Linux player does not seem to be very fast, so it can miss
  frames. An example is:

  ffmpeg -g 3 -r 3 -t 10 -b 50 -s qcif -f rv10 /tmp/b.rm

 - The parameter 'q' which is displayed while encoding is the current
  quantizer. The value of 1 indicates that a very good quality could
  be achieved. The value of 31 indicates the worst quality. If q=31
  too often, it means that the encoder cannot compress enough to meet
  your bit rate. You must either increase the bit rate, decrease the
  frame rate or decrease the frame size.

 - If your computer is not fast enough, you can speed up the
  compression at the expense of the compression ratio. You can use
  '-me zero' to speed up motion estimation, and '-intra' to disable
  completly motion estimation (you have only I frames, which means it
  is about as good as JPEG compression).

 - To have very low bitrates in audio, reduce the sampling frequency
  (down to 22050 kHz for mpeg audio, 22050 or 11025 for ac3).

 - To have a constant quality (but a variable bitrate), use the option
  '-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
  quality).

 - When converting video files, you can use the '-sameq' option which
  uses in the encoder the same quality factor than in the decoder. It
  allows to be almost lossless in encoding.
--- a/doc/ffserver.conf
+++ b/doc/ffserver.conf
@@ -0,0 +1,261 @@
 # Port on which the server is listening. You must select a different
 # port from your standard http web server if it is running on the same
 # computer.

 Port 8090

 # Address on which the server is bound. Only useful if you have
 # several network interfaces.

 BindAddress 0.0.0.0

 # Number of simultaneous requests that can be handled. Since FFServer
 # is very fast, this limit is determined mainly by your Internet
 # connection speed.

 MaxClients 1000

 # Access Log file (uses standard Apache log file format)
 # '-' is the standard output

 CustomLog -

 ##################################################################
 # Definition of the live feeds. Each live feed contains one video
 # and/or audio sequence coming from an ffmpeg encoder or another
 # ffserver. This sequence may be encoded simultaneously with several
 # codecs at several resolutions.

 <Feed feed1.ffm>

 # You must use 'ffmpeg' to send a live feed to ffserver. In this
 # example, you can type:
 # 
 # ffmpeg http://localhost:8090/feed1.ffm

 # ffserver can also do time shifting. It means that it can stream any
 # previously recorded live stream. The request should contain:
 # "http://xxxx?date=[YYYY-MM-DDT][[HH:]MM:]SS[.m...]".You must specify
 # a path where the feed is stored on disk. You also specify the
 # maximum size of the feed (100M bytes here). Default:
 # File=/tmp/feed_name.ffm FileMaxSize=5M

 File /tmp/feed1.ffm
 FileMaxSize 50M

 </Feed>

 ##################################################################
 # Now you can define each stream which will be generated from the
 # original audio and video stream. Each format has a filename (here
 # 'test128.mpg'). FFServer will send this stream when answering a
 # request containing this filename.

 <Stream test1.mpg>

 # coming from live feed 'feed1'
 Feed feed1.ffm

 # Format of the stream : you can choose among:
 # mpeg       : MPEG1 multiplexed video and audio
 # mpegvideo  : only MPEG1 video
 # mp2        : MPEG audio layer 2
 # mp3        : MPEG audio layer 3 (currently sent as layer 2)
 # rm         : Real Networks compatible stream. Multiplexed audio and video.
 # ra         : Real Networks compatible stream. Audio only.
 # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
 # jpeg       : Generate a single JPEG image.
 # asf        : ASF compatible stream (Windows Media Player format). Not finished yet.
 # swf        : Macromedia flash(tm) compatible stream
 # avi        : AVI format (open divx video, mpeg audio sound)
 # master     : special ffmpeg stream used to duplicate a server

 Format mpeg

 # Bitrate for the audio stream. Codecs usually support only a few different bitrates. 

 AudioBitRate 32

 # Number of audio channels : 1 = mono, 2 = stereo

 AudioChannels 1

 # Sampling frequency for audio. When using low bitrates, you should
 # lower this frequency to 22050 or 11025. The supported frequencies
 # depend on the selected audio codec.

 AudioSampleRate 44100

 # 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>
--- a/ffmpeg.c
+++ b/ffmpeg.c
--- a/ffserver.c
+++ b/ffserver.c
--- a/libav/Makefile
+++ b/libav/Makefile
@@ -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