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Updated FFMPEG to version 1.1.2, using this project: http://sourceforge.net/projects/ffmpeg4android/

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Sergii Pylypenko
2013-02-21 18:29:51 +02:00
parent 758a9658d2
commit fff7a99a41
3492 changed files with 886704 additions and 5414 deletions
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LIBRARIES-$(CONFIG_AVUTIL) += libavutil
LIBRARIES-$(CONFIG_SWSCALE) += libswscale
LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
LIBRARIES-$(CONFIG_AVCODEC) += libavcodec
LIBRARIES-$(CONFIG_AVFORMAT) += libavformat
LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice
LIBRARIES-$(CONFIG_AVFILTER) += libavfilter
COMPONENTS-yes = $(PROGS-yes)
COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils
COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler
COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters
COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES = $(COMPONENTS-yes:%=doc/%.1) $(LIBRARIES-yes:%=doc/%.3)
PODPAGES = $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/nut.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
DOCS-$(CONFIG_PODPAGES) += $(PODPAGES)
DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
all-$(CONFIG_DOC): doc
doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
TEXIDEP = awk '/^@(verbatim)?include/ { printf "$@: $(@D)/%s\n", $$2 }' <$< >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Idoc $< $@
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(INSTHEADERS)
$(M)$(SRC_PATH)/doc/doxy-wrapper.sh $(SRC_PATH) $^
install-man:
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES) "$(MANDIR)/man1"
endif
uninstall: uninstall-man
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(ALLMANPAGES))
docclean: clean
clean::
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 $(CLEANSUFFIXES:%=doc/%) doc/avoptions_*.texi
$(RM) -r doc/doxy/html
-include $(wildcard $(DOCS:%=%.d))
.PHONY: apidoc doc documentation

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Release Notes
=============
* 1.1 "Fire Flower" January, 2013
General notes
-------------
See the Changelog file for a list of significant changes. Note, there
are many more new features and bugfixes than whats listed there.
Bugreports against FFmpeg git master or the most recent FFmpeg release are
accepted. If you are experiencing issues with any formally released version of
FFmpeg, please try git master to check if the issue still exists. If it does,
make your report against the development code following the usual bug reporting
guidelines.
Of big interest to our Windows users, FFmpeg now supports building with the MSVC
compiler. Since MSVC does not support C99 features used extensively by FFmpeg,
this has been accomplished using a converter that turns C99 code to C89. See the
platform-specific documentation for more detailed documentation on building
FFmpeg with MSVC.
The used output sample format for several audio decoders has changed, make
sure you always check/use AVCodecContext.sample_fmt or AVFrame.format.

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@chapter Authors
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
@command{git log} in the FFmpeg source directory, or browsing the
online repository at @url{http://source.ffmpeg.org}.
Maintainers for the specific components are listed in the file
@file{MAINTAINERS} in the source code tree.

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All the numerical options, if not specified otherwise, accept in input
a string representing a number, which may contain one of the
SI unit prefixes, for example 'K', 'M', 'G'.
If 'i' is appended after the prefix, binary prefixes are used,
which are based on powers of 1024 instead of powers of 1000.
The 'B' postfix multiplies the value by 8, and can be
appended after a unit prefix or used alone. This allows using for
example 'KB', 'MiB', 'G' and 'B' as number postfix.
Options which do not take arguments are boolean options, and set the
corresponding value to true. They can be set to false by prefixing
with "no" the option name, for example using "-nofoo" in the
command line will set to false the boolean option with name "foo".
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
@code{a:1} stream specifier, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, the option is then applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams, for example @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4.
@item @var{stream_type}[:@var{stream_index}]
@var{stream_type} is one of: 'v' for video, 'a' for audio, 's' for subtitle,
'd' for data and 't' for attachments. If @var{stream_index} is given, then
matches stream number @var{stream_index} of this type. Otherwise matches all
streams of this type.
@item p:@var{program_id}[:@var{stream_index}]
If @var{stream_index} is given, then matches stream number @var{stream_index} in
program with id @var{program_id}. Otherwise matches all streams in this program.
@item #@var{stream_id}
Matches the stream by format-specific ID.
@end table
@section Generic options
These options are shared amongst the av* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help [@var{arg}]
Show help. An optional parameter may be specified to print help about a specific
item.
Possible values of @var{arg} are:
@table @option
@item decoder=@var{decoder_name}
Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@item encoder=@var{encoder_name}
Print detailed information about the encoder named @var{encoder_name}. Use the
@option{-encoders} option to get a list of all encoders.
@item demuxer=@var{demuxer_name}
Print detailed information about the demuxer named @var{demuxer_name}. Use the
@option{-formats} option to get a list of all demuxers and muxers.
@item muxer=@var{muxer_name}
Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@end table
@item -version
Show version.
@item -formats
Show available formats.
The fields preceding the format names have the following meanings:
@table @samp
@item D
Decoding available
@item E
Encoding available
@end table
@item -codecs
Show all codecs known to libavcodec.
Note that the term 'codec' is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
@item -decoders
Show available decoders.
@item -encoders
Show all available encoders.
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -loglevel @var{loglevel} | -v @var{loglevel}
Set the logging level used by the library.
@var{loglevel} is a number or a string containing one of the following values:
@table @samp
@item quiet
@item panic
@item fatal
@item error
@item warning
@item info
@item verbose
@item debug
@end table
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR} or @env{NO_COLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
The use of the environment variable @env{NO_COLOR} is deprecated and
will be dropped in a following FFmpeg version.
@item -report
Dump full command line and console output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel verbose}.
Setting the environment variable @code{FFREPORT} to any value has the
same effect. If the value is a ':'-separated key=value sequence, these
options will affect the report; options values must be escaped if they
contain special characters or the options delimiter ':' (see the
``Quoting and escaping'' section in the ffmpeg-utils manual). The
following option is recognized:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@end table
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
@end example
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are obviously per-stream, so the chapter on stream
specifiers applies to them
Note @option{-nooption} syntax cannot be used for boolean AVOptions,
use @option{-option 0}/@option{-option 1}.
Note2 old undocumented way of specifying per-stream AVOptions by prepending
v/a/s to the options name is now obsolete and will be removed soon.

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AVUtil
======
libavutil is a small lightweight library of generally useful functions.
It is not a library for code needed by both libavcodec and libavformat.
Overview:
=========
adler32.c adler32 checksum
aes.c AES encryption and decryption
fifo.c resizeable first in first out buffer
intfloat_readwrite.c portable reading and writing of floating point values
log.c "printf" with context and level
md5.c MD5 Message-Digest Algorithm
rational.c code to perform exact calculations with rational numbers
tree.c generic AVL tree
crc.c generic CRC checksumming code
integer.c 128bit integer math
lls.c
mathematics.c greatest common divisor, integer sqrt, integer log2, ...
mem.c memory allocation routines with guaranteed alignment
Headers:
bswap.h big/little/native-endian conversion code
x86_cpu.h a few useful macros for unifying x86-64 and x86-32 code
avutil.h
common.h
intreadwrite.h reading and writing of unaligned big/little/native-endian integers
Goals:
======
* Modular (few interdependencies and the possibility of disabling individual parts during ./configure)
* Small (source and object)
* Efficient (low CPU and memory usage)
* Useful (avoid useless features almost no one needs)

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@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option @code{--list-bsfs}.
You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
the option @code{--enable-bsf=BSF}, or you can disable a particular
bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
Below is a description of the currently available bitstream filters.
@section aac_adtstoasc
@section chomp
@section dump_extradata
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
@section imx_dump_header
@section mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
@example
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
@example
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpega_dump_header
@section movsub
@section mp3_header_compress
@section mp3_header_decompress
@section noise
@section remove_extradata
@c man end BITSTREAM FILTERS

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FFmpeg currently uses a custom build system, this text attempts to document
some of its obscure features and options.
Makefile variables:
V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note you must have installed it
fate-list
Will list all fate/regression test targets
install
Install headers, libraries and programs.
libavformat/output-example
Build the libavformat basic example.
libavcodec/api-example
Build the libavcodec basic example.
libswscale/swscale-test
Build the swscale self-test (useful also as example).
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be build, this is useful to reduce
unneeded rebuilding when changing headers, but note you must force rebuilds
of files that actually need it by hand then.
make -j<num>
rebuild with multiple jobs at the same time. Faster on multi processor systems
make -k
continue build in case of errors, this is useful for the regression tests
sometimes but note it will still not run all reg tests.

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@chapter Decoders
@c man begin DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available decoders using the configure option @code{--list-decoders}.
You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
A description of some of the currently available video decoders
follows.
@section rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
Specify the assumed field type of the input video.
@table @option
@item -1
the video is assumed to be progressive (default)
@item 0
bottom-field-first is assumed
@item 1
top-field-first is assumed
@end table
@end table
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
@section ffwavesynth
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps. When stored in VobSub, the
palette is normally specified in the index file; in Matroska, the palette is
stored in the codec extra-data in the same format as in VobSub. In DVDs, the
palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@end table
@c man end SUBTILES DECODERS

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@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg which allow to read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option "--list-demuxers".
You can disable all the demuxers using the configure option
"--disable-demuxers", and selectively enable a single demuxer with
the option "--enable-demuxer=@var{DEMUXER}", or disable it
with the option "--disable-demuxer=@var{DEMUXER}".
The option "-formats" of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The syntax and meaning of the pattern is specified by the
option @var{pattern_type}.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
This demuxer accepts the following options:
@table @option
@item framerate
Set the framerate for the video stream. It defaults to 25.
@item loop
If set to 1, loop over the input. Default value is 0.
@item pattern_type
Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
A sequence pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between @var{start_number} and
@var{start_number}+@var{start_number_range}-1, and all the following
numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@item glob
Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@item start_number
Set the index of the file matched by the image file pattern to start
to read from. Default value is 0.
@item start_number_range
Set the index interval range to check when looking for the first image
file in the sequence, starting from @var{start_number}. Default value
is 5.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} for creating a video from the images in the file
sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
input frame rate of 10 frames per second:
@example
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv
@end example
@end itemize
@section applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
@section sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
script looks like that:
@example
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
@end example
A SBG script can mix absolute and relative timestamps. If the script uses
either only absolute timestamps (including the script start time) or only
relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the @var{NOW} reference for relative timestamps will be
taken from the current time of day at the time the script is read, and the
script layout will be frozen according to that reference. That means that if
the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
@section concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packet had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
and each next file starts where the previous one finishes. Note that it is
done globally and may cause gaps if all streams do not have exactly the same
length.
All files must have the same streams (same codecs, same time base, etc.).
This script format can currently not be probed, it must be specified explicitly.
@subsection Syntax
The script is a text file in extended-ASCII, with one directive per line.
Empty lines, leading spaces and lines starting with '#' are ignored. The
following directive is recognized:
@table @option
@item @code{file @var{path}}
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
@end table
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
TED does not provide links to the captions, but they can be guessed from the
page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
contains a bookmarklet to expose them.
This demuxer accepts the following option:
@table @option
@item start_time
Set the start time of the TED talk, in milliseconds. The default is 15000
(15s). It is used to sync the captions with the downloadable videos, because
they include a 15s intro.
@end table
Example: convert the captions to a format most players understand:
@example
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@c man end INPUT DEVICES

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\input texinfo @c -*- texinfo -*-
@settitle Developer Documentation
@titlepage
@center @titlefont{Developer Documentation}
@end titlepage
@top
@contents
@chapter Developers Guide
@section API
@itemize @bullet
@item libavcodec is the library containing the codecs (both encoding and
decoding). Look at @file{doc/examples/decoding_encoding.c} to see how to use
it.
@item libavformat is the library containing the file format handling (mux and
demux code for several formats). Look at @file{ffplay.c} to use it in a
player. See @file{doc/examples/muxing.c} to use it to generate audio or video
streams.
@end itemize
@section Integrating libavcodec or libavformat in your program
You can integrate all the source code of the libraries to link them
statically to avoid any version problem. All you need is to provide a
'config.mak' and a 'config.h' in the parent directory. See the defines
generated by ./configure to understand what is needed.
You can use libavcodec or libavformat in your commercial program, but
@emph{any patch you make must be published}. The best way to proceed is
to send your patches to the FFmpeg mailing list.
@section Contributing
There are 3 ways by which code gets into ffmpeg.
@itemize @bullet
@item Submitting Patches to the main developer mailing list
see @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@item Committing changes to a git clone, for example on github.com or
gitorious.org. And asking us to merge these changes.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@section Coding Rules
@subsection Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@subsection Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
@code{//!} with @code{///} and similar. Also @@ syntax should be employed
for markup commands, i.e. use @code{@@param} and not @code{\param}.
@example
/**
* @@file
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar@{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
int myfunc(int my_parameter)
...
@end example
@subsection C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};})
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};})
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@subsection Naming conventions
All names are using underscores (_), not CamelCase. For example, @samp{avfilter_get_video_buffer} is
a valid function name and @samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in the CamelCase
There are following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For variables and functions declared as @code{static} no prefixes are required.
@item
For variables and functions used internally by the library, @code{ff_} prefix
should be used.
For example, @samp{ff_w64_demuxer}.
@item
For variables and functions used internally across multiple libraries, use
@code{avpriv_}. For example, @samp{avpriv_aac_parse_header}.
@item
For exported names, each library has its own prefixes. Just check the existing
code and name accordingly.
@end itemize
@subsection Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@subsection Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" allow tabs in Makefiles
autocmd FileType make set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@example
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end example
@section Development Policy
@enumerate
@item
Contributions should be licensed under the LGPL 2.1, including an
"or any later version" clause, or the MIT license. GPL 2 including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
@item
You must not commit code which breaks FFmpeg! (Meaning unfinished but
enabled code which breaks compilation or compiles but does not work or
breaks the regression tests)
You can commit unfinished stuff (for testing etc), but it must be disabled
(#ifdef etc) by default so it does not interfere with other developers'
work.
@item
You do not have to over-test things. If it works for you, and you think it
should work for others, then commit. If your code has problems
(portability, triggers compiler bugs, unusual environment etc) they will be
reported and eventually fixed.
@item
Do not commit unrelated changes together, split them into self-contained
pieces. Also do not forget that if part B depends on part A, but A does not
depend on B, then A can and should be committed first and separate from B.
Keeping changes well split into self-contained parts makes reviewing and
understanding them on the commit log mailing list easier. This also helps
in case of debugging later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@item
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove functionality from the code. Just improve!
Note: Redundant code can be removed.
@item
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@item
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@item
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
area changed: Short 1 line description
details describing what and why and giving references.
@item
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@item
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@item
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel instead. If no one answers within a reasonable
timeframe (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@item
Subscribe to the ffmpeg-cvslog mailing list. The diffs of all commits
are sent there and reviewed by all the other developers. Bugs and possible
improvements or general questions regarding commits are discussed there. We
expect you to react if problems with your code are uncovered.
@item
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@item
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@item
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@item
Remember to check if you need to bump versions for the specific libav*
parts (libavutil, libavcodec, libavformat) you are changing. You need
to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@item
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@item
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@end enumerate
We think our rules are not too hard. If you have comments, contact us.
Note, these rules are mostly borrowed from the MPlayer project.
@anchor{Submitting patches}
@section Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-)
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
Run the @ref{Regression tests} before submitting a patch in order to verify
it does not cause unexpected problems.
Patches should be posted as base64 encoded attachments (or any other
encoding which ensures that the patch will not be trashed during
transmission) to the ffmpeg-devel mailing list, see
@url{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel}
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@section New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
@item
If it has a fourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@section patch submission checklist
@enumerate
@item
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
@item
Did you sign off your patch? (git commit -s)
See @url{http://git.kernel.org/?p=linux/kernel/git/torvalds/linux.git;a=blob_plain;f=Documentation/SubmittingPatches} for the meaning
of sign off.
@item
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher and the noise bitstream filter. Your decoder or demuxer
should not crash or end in a (near) infinite loop when fed damaged data.
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to ftp://upload.ffmpeg.org
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider to add a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@end enumerate
@section Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
@anchor{Regression tests}
@section Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
Running 'make fate' accomplishes this, please see @url{fate.html} for details.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
@subsection Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be inlcuded in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductionary message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@bye

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SRC_PATH="${1}"
DOXYFILE="${2}"
shift 2
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@INCLUDE = ${DOXYFILE}
INPUT = $@
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HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html
HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
EOF

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<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
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@chapter Encoders
@c man begin ENCODERS
Encoders are configured elements in FFmpeg which allow the encoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available encoders using the configure option @code{--list-encoders}.
You can disable all the encoders with the configure option
@code{--disable-encoders} and selectively enable / disable single encoders
with the options @code{--enable-encoder=@var{ENCODER}} /
@code{--disable-encoder=@var{ENCODER}}.
The option @code{-codecs} of the ff* tools will display the list of
enabled encoders.
@c man end ENCODERS
@chapter Audio Encoders
@c man begin AUDIO ENCODERS
A description of some of the currently available audio encoders
follows.
@section ac3 and ac3_fixed
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
The @var{ac3} encoder uses floating-point math, while the @var{ac3_fixed}
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The @var{ac3_fixed} encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option @code{-acodec ac3_fixed} in order to use it.
@subsection AC-3 Metadata
The AC-3 metadata options are used to set parameters that describe the audio,
but in most cases do not affect the audio encoding itself. Some of the options
do directly affect or influence the decoding and playback of the resulting
bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for
audio data, and will thus affect the quality of the output. Those will be
indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available
documents.
@itemize
@item @uref{http://www.atsc.org/cms/standards/a_52-2010.pdf,A/52:2010 - Digital Audio Compression (AC-3) (E-AC-3) Standard}
@item @uref{http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf,A/54 - Guide to the Use of the ATSC Digital Television Standard}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf,Dolby Metadata Guide}
@item @uref{http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf,Dolby Digital Professional Encoding Guidelines}
@end itemize
@subsubsection Metadata Control Options
@table @option
@item -per_frame_metadata @var{boolean}
Allow Per-Frame Metadata. Specifies if the encoder should check for changing
metadata for each frame.
@table @option
@item 0
The metadata values set at initialization will be used for every frame in the
stream. (default)
@item 1
Metadata values can be changed before encoding each frame.
@end table
@end table
@subsubsection Downmix Levels
@table @option
@item -center_mixlev @var{level}
Center Mix Level. The amount of gain the decoder should apply to the center
channel when downmixing to stereo. This field will only be written to the
bitstream if a center channel is present. The value is specified as a scale
factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6dB gain
@end table
@item -surround_mixlev @var{level}
Surround Mix Level. The amount of gain the decoder should apply to the surround
channel(s) when downmixing to stereo. This field will only be written to the
bitstream if one or more surround channels are present. The value is specified
as a scale factor. There are 3 valid values:
@table @option
@item 0.707
Apply -3dB gain
@item 0.500
Apply -6dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Audio Production Information
Audio Production Information is optional information describing the mixing
environment. Either none or both of the fields are written to the bitstream.
@table @option
@item -mixing_level @var{number}
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the @code{room_type} option is not the default value, the @code{mixing_level}
option must not be -1.
@item -room_type @var{type}
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the @code{mixing_level}
option and the @code{room_type} option have the default values.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx large
Large Room
@item 2
@itemx small
Small Room
@end table
@end table
@subsubsection Other Metadata Options
@table @option
@item -copyright @var{boolean}
Copyright Indicator. Specifies whether a copyright exists for this audio.
@table @option
@item 0
@itemx off
No Copyright Exists (default)
@item 1
@itemx on
Copyright Exists
@end table
@item -dialnorm @var{value}
Dialogue Normalization. Indicates how far the average dialogue level of the
program is below digital 100% full scale (0 dBFS). This parameter determines a
level shift during audio reproduction that sets the average volume of the
dialogue to a preset level. The goal is to match volume level between program
sources. A value of -31dB will result in no volume level change, relative to
the source volume, during audio reproduction. Valid values are whole numbers in
the range -31 to -1, with -31 being the default.
@item -dsur_mode @var{mode}
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround
(Pro Logic). This field will only be written to the bitstream if the audio
stream is stereo. Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx off
Not Dolby Surround Encoded
@item 2
@itemx on
Dolby Surround Encoded
@end table
@item -original @var{boolean}
Original Bit Stream Indicator. Specifies whether this audio is from the
original source and not a copy.
@table @option
@item 0
@itemx off
Not Original Source
@item 1
@itemx on
Original Source (default)
@end table
@end table
@subsection Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the @code{center_mixlev}
and @code{surround_mixlev} options if it supports the Alternate Bit Stream
Syntax.
@subsubsection Extended Bitstream Information - Part 1
@table @option
@item -dmix_mode @var{mode}
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt
(Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx ltrt
Lt/Rt Downmix Preferred
@item 2
@itemx loro
Lo/Ro Downmix Preferred
@end table
@item -ltrt_cmixlev @var{level}
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lt/Rt mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -ltrt_surmixlev @var{level}
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lt/Rt mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@item -loro_cmixlev @var{level}
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the
center channel when downmixing to stereo in Lo/Ro mode.
@table @option
@item 1.414
Apply +3dB gain
@item 1.189
Apply +1.5dB gain
@item 1.000
Apply 0dB gain
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain (default)
@item 0.500
Apply -6.0dB gain
@item 0.000
Silence Center Channel
@end table
@item -loro_surmixlev @var{level}
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the
surround channel(s) when downmixing to stereo in Lo/Ro mode.
@table @option
@item 0.841
Apply -1.5dB gain
@item 0.707
Apply -3.0dB gain
@item 0.595
Apply -4.5dB gain
@item 0.500
Apply -6.0dB gain (default)
@item 0.000
Silence Surround Channel(s)
@end table
@end table
@subsubsection Extended Bitstream Information - Part 2
@table @option
@item -dsurex_mode @var{mode}
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX
(7.1 matrixed to 5.1). Using this option does @b{NOT} mean the encoder will actually
apply Dolby Surround EX processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Surround EX Off
@item 2
@itemx off
Dolby Surround EX On
@end table
@item -dheadphone_mode @var{mode}
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone
encoding (multi-channel matrixed to 2.0 for use with headphones). Using this
option does @b{NOT} mean the encoder will actually apply Dolby Headphone
processing.
@table @option
@item 0
@itemx notindicated
Not Indicated (default)
@item 1
@itemx on
Dolby Headphone Off
@item 2
@itemx off
Dolby Headphone On
@end table
@item -ad_conv_type @var{type}
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D
conversion.
@table @option
@item 0
@itemx standard
Standard A/D Converter (default)
@item 1
@itemx hdcd
HDCD A/D Converter
@end table
@end table
@subsection Other AC-3 Encoding Options
@table @option
@item -stereo_rematrixing @var{boolean}
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This
is an optional AC-3 feature that increases quality by selectively encoding
the left/right channels as mid/side. This option is enabled by default, and it
is highly recommended that it be left as enabled except for testing purposes.
@end table
@subsection Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not exist
for the fixed-point encoder due to the corresponding features not being
implemented in fixed-point.
@table @option
@item -channel_coupling @var{boolean}
Enables/Disables use of channel coupling, which is an optional AC-3 feature
that increases quality by combining high frequency information from multiple
channels into a single channel. The per-channel high frequency information is
sent with less accuracy in both the frequency and time domains. This allows
more bits to be used for lower frequencies while preserving enough information
to reconstruct the high frequencies. This option is enabled by default for the
floating-point encoder and should generally be left as enabled except for
testing purposes or to increase encoding speed.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@item 0
@itemx off
Disable Channel Coupling
@item 1
@itemx on
Enable Channel Coupling
@end table
@item -cpl_start_band @var{number}
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a
value higher than the bandwidth is used, it will be reduced to 1 less than the
coupling end band. If @var{auto} is used, the start band will be determined by
the encoder based on the bit rate, sample rate, and channel layout. This option
has no effect if channel coupling is disabled.
@table @option
@item -1
@itemx auto
Selected by Encoder (default)
@end table
@end table
@c man end AUDIO ENCODERS
@chapter Video Encoders
@c man begin VIDEO ENCODERS
A description of some of the currently available video encoders
follows.
@section libtheora
Theora format supported through libtheora.
Requires the presence of the libtheora headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libtheora}.
@subsection Options
The following global options are mapped to internal libtheora options
which affect the quality and the bitrate of the encoded stream.
@table @option
@item b
Set the video bitrate, only works if the @code{qscale} flag in
@option{flags} is not enabled.
@item flags
Used to enable constant quality mode encoding through the
@option{qscale} flag, and to enable the @code{pass1} and @code{pass2}
modes.
@item g
Set the GOP size.
@item global_quality
Set the global quality in lambda units, only works if the
@code{qscale} flag in @option{flags} is enabled. The value is clipped
in the [0 - 10*@code{FF_QP2LAMBDA}] range, and then multiplied for 6.3
to get a value in the native libtheora range [0-63]. A higher value
corresponds to a higher quality.
For example, to set maximum constant quality encoding with
@command{ffmpeg}:
@example
ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg
@end example
@end table
@section libvpx
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libvpx}.
@subsection Options
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
@table @option
@item threads
g_threads
@item profile
g_profile
@item vb
rc_target_bitrate
@item g
kf_max_dist
@item keyint_min
kf_min_dist
@item qmin
rc_min_quantizer
@item qmax
rc_max_quantizer
@item bufsize, vb
rc_buf_sz
@code{(bufsize * 1000 / vb)}
rc_buf_optimal_sz
@code{(bufsize * 1000 / vb * 5 / 6)}
@item rc_init_occupancy, vb
rc_buf_initial_sz
@code{(rc_init_occupancy * 1000 / vb)}
@item rc_buffer_aggressivity
rc_undershoot_pct
@item skip_threshold
rc_dropframe_thresh
@item qcomp
rc_2pass_vbr_bias_pct
@item maxrate, vb
rc_2pass_vbr_maxsection_pct
@code{(maxrate * 100 / vb)}
@item minrate, vb
rc_2pass_vbr_minsection_pct
@code{(minrate * 100 / vb)}
@item minrate, maxrate, vb
@code{VPX_CBR}
@code{(minrate == maxrate == vb)}
@item crf
@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
@item quality
@table @option
@item @var{best}
@code{VPX_DL_BEST_QUALITY}
@item @var{good}
@code{VPX_DL_GOOD_QUALITY}
@item @var{realtime}
@code{VPX_DL_REALTIME}
@end table
@item speed
@code{VP8E_SET_CPUUSED}
@item nr
@code{VP8E_SET_NOISE_SENSITIVITY}
@item mb_threshold
@code{VP8E_SET_STATIC_THRESHOLD}
@item slices
@code{VP8E_SET_TOKEN_PARTITIONS}
@item max-intra-rate
@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
@item force_key_frames
@code{VPX_EFLAG_FORCE_KF}
@item Alternate reference frame related
@table @option
@item vp8flags altref
@code{VP8E_SET_ENABLEAUTOALTREF}
@item @var{arnr_max_frames}
@code{VP8E_SET_ARNR_MAXFRAMES}
@item @var{arnr_type}
@code{VP8E_SET_ARNR_TYPE}
@item @var{arnr_strength}
@code{VP8E_SET_ARNR_STRENGTH}
@item @var{rc_lookahead}
g_lag_in_frames
@end table
@item vp8flags error_resilient
g_error_resilient
@end table
For more information about libvpx see:
@url{http://www.webmproject.org/}
@section libx264
H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through
libx264.
Requires the presence of the libx264 headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libx264}.
@subsection Options
@table @option
@item preset @var{preset_name}
Set the encoding preset.
@item tune @var{tune_name}
Tune the encoding params.
@item fastfirstpass @var{bool}
Use fast settings when encoding first pass, default value is 1.
@item profile @var{profile_name}
Set profile restrictions.
@item level @var{level}
Specify level (as defined by Annex A).
Deprecated in favor of @var{x264opts}.
@item passlogfile @var{filename}
Specify filename for 2 pass stats.
Deprecated in favor of @var{x264opts} (see @var{stats} libx264 option).
@item wpredp @var{wpred_type}
Specify Weighted prediction for P-frames.
Deprecated in favor of @var{x264opts} (see @var{weightp} libx264 option).
@item x264opts @var{options}
Allow to set any x264 option, see @code{x264 --fullhelp} for a list.
@var{options} is a list of @var{key}=@var{value} couples separated by
":". In @var{filter} and @var{psy-rd} options that use ":" as a separator
themselves, use "," instead. They accept it as well since long ago but this
is kept undocumented for some reason.
@end table
For example to specify libx264 encoding options with @command{ffmpeg}:
@example
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
@end example
For more information about libx264 and the supported options see:
@url{http://www.videolan.org/developers/x264.html}
@c man end VIDEO ENCODERS

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The following table lists most error codes found in various operating
systems supported by FFmpeg.
OS
Code Std F LBMWwb Text (YMMV)
E2BIG POSIX ++++++ Argument list too long
EACCES POSIX ++++++ Permission denied
EADDRINUSE POSIX +++..+ Address in use
EADDRNOTAVAIL POSIX +++..+ Cannot assign requested address
EADV +..... Advertise error
EAFNOSUPPORT POSIX +++..+ Address family not supported
EAGAIN POSIX + ++++++ Resource temporarily unavailable
EALREADY POSIX +++..+ Operation already in progress
EAUTH .++... Authentication error
EBADARCH ..+... Bad CPU type in executable
EBADE +..... Invalid exchange
EBADEXEC ..+... Bad executable
EBADF POSIX ++++++ Bad file descriptor
EBADFD +..... File descriptor in bad state
EBADMACHO ..+... Malformed Macho file
EBADMSG POSIX ++4... Bad message
EBADR +..... Invalid request descriptor
EBADRPC .++... RPC struct is bad
EBADRQC +..... Invalid request code
EBADSLT +..... Invalid slot
EBFONT +..... Bad font file format
EBUSY POSIX - ++++++ Device or resource busy
ECANCELED POSIX +++... Operation canceled
ECHILD POSIX ++++++ No child processes
ECHRNG +..... Channel number out of range
ECOMM +..... Communication error on send
ECONNABORTED POSIX +++..+ Software caused connection abort
ECONNREFUSED POSIX - +++ss+ Connection refused
ECONNRESET POSIX +++..+ Connection reset
EDEADLK POSIX ++++++ Resource deadlock avoided
EDEADLOCK +..++. File locking deadlock error
EDESTADDRREQ POSIX +++... Destination address required
EDEVERR ..+... Device error
EDOM C89 - ++++++ Numerical argument out of domain
EDOOFUS .F.... Programming error
EDOTDOT +..... RFS specific error
EDQUOT POSIX +++... Disc quota exceeded
EEXIST POSIX ++++++ File exists
EFAULT POSIX - ++++++ Bad address
EFBIG POSIX - ++++++ File too large
EFTYPE .++... Inappropriate file type or format
EHOSTDOWN +++... Host is down
EHOSTUNREACH POSIX +++..+ No route to host
EHWPOISON +..... Memory page has hardware error
EIDRM POSIX +++... Identifier removed
EILSEQ C99 ++++++ Illegal byte sequence
EINPROGRESS POSIX - +++ss+ Operation in progress
EINTR POSIX - ++++++ Interrupted system call
EINVAL POSIX + ++++++ Invalid argument
EIO POSIX + ++++++ I/O error
EISCONN POSIX +++..+ Socket is already connected
EISDIR POSIX ++++++ Is a directory
EISNAM +..... Is a named type file
EKEYEXPIRED +..... Key has expired
EKEYREJECTED +..... Key was rejected by service
EKEYREVOKED +..... Key has been revoked
EL2HLT +..... Level 2 halted
EL2NSYNC +..... Level 2 not synchronized
EL3HLT +..... Level 3 halted
EL3RST +..... Level 3 reset
ELIBACC +..... Can not access a needed shared library
ELIBBAD +..... Accessing a corrupted shared library
ELIBEXEC +..... Cannot exec a shared library directly
ELIBMAX +..... Too many shared libraries
ELIBSCN +..... .lib section in a.out corrupted
ELNRNG +..... Link number out of range
ELOOP POSIX +++..+ Too many levels of symbolic links
EMEDIUMTYPE +..... Wrong medium type
EMFILE POSIX ++++++ Too many open files
EMLINK POSIX ++++++ Too many links
EMSGSIZE POSIX +++..+ Message too long
EMULTIHOP POSIX ++4... Multihop attempted
ENAMETOOLONG POSIX - ++++++ Filen ame too long
ENAVAIL +..... No XENIX semaphores available
ENEEDAUTH .++... Need authenticator
ENETDOWN POSIX +++..+ Network is down
ENETRESET SUSv3 +++..+ Network dropped connection on reset
ENETUNREACH POSIX +++..+ Network unreachable
ENFILE POSIX ++++++ Too many open files in system
ENOANO +..... No anode
ENOATTR .++... Attribute not found
ENOBUFS POSIX - +++..+ No buffer space available
ENOCSI +..... No CSI structure available
ENODATA XSR +N4... No message available
ENODEV POSIX - ++++++ No such device
ENOENT POSIX - ++++++ No such file or directory
ENOEXEC POSIX ++++++ Exec format error
ENOFILE ...++. No such file or directory
ENOKEY +..... Required key not available
ENOLCK POSIX ++++++ No locks available
ENOLINK POSIX ++4... Link has been severed
ENOMEDIUM +..... No medium found
ENOMEM POSIX ++++++ Not enough space
ENOMSG POSIX +++..+ No message of desired type
ENONET +..... Machine is not on the network
ENOPKG +..... Package not installed
ENOPROTOOPT POSIX +++..+ Protocol not available
ENOSPC POSIX ++++++ No space left on device
ENOSR XSR +N4... No STREAM resources
ENOSTR XSR +N4... Not a STREAM
ENOSYS POSIX + ++++++ Function not implemented
ENOTBLK +++... Block device required
ENOTCONN POSIX +++..+ Socket is not connected
ENOTDIR POSIX ++++++ Not a directory
ENOTEMPTY POSIX ++++++ Directory not empty
ENOTNAM +..... Not a XENIX named type file
ENOTRECOVERABLE SUSv4 - +..... State not recoverable
ENOTSOCK POSIX +++..+ Socket operation on non-socket
ENOTSUP POSIX +++... Operation not supported
ENOTTY POSIX ++++++ Inappropriate I/O control operation
ENOTUNIQ +..... Name not unique on network
ENXIO POSIX ++++++ No such device or address
EOPNOTSUPP POSIX +++..+ Operation not supported (on socket)
EOVERFLOW POSIX +++..+ Value too large to be stored in data type
EOWNERDEAD SUSv4 +..... Owner died
EPERM POSIX - ++++++ Operation not permitted
EPFNOSUPPORT +++..+ Protocol family not supported
EPIPE POSIX - ++++++ Broken pipe
EPROCLIM .++... Too many processes
EPROCUNAVAIL .++... Bad procedure for program
EPROGMISMATCH .++... Program version wrong
EPROGUNAVAIL .++... RPC prog. not avail
EPROTO POSIX ++4... Protocol error
EPROTONOSUPPORT POSIX - +++ss+ Protocol not supported
EPROTOTYPE POSIX +++..+ Protocol wrong type for socket
EPWROFF ..+... Device power is off
ERANGE C89 - ++++++ Result too large
EREMCHG +..... Remote address changed
EREMOTE +++... Object is remote
EREMOTEIO +..... Remote I/O error
ERESTART +..... Interrupted system call should be restarted
ERFKILL +..... Operation not possible due to RF-kill
EROFS POSIX ++++++ Read-only file system
ERPCMISMATCH .++... RPC version wrong
ESHLIBVERS ..+... Shared library version mismatch
ESHUTDOWN +++..+ Cannot send after socket shutdown
ESOCKTNOSUPPORT +++... Socket type not supported
ESPIPE POSIX ++++++ Illegal seek
ESRCH POSIX ++++++ No such process
ESRMNT +..... Srmount error
ESTALE POSIX +++..+ Stale NFS file handle
ESTRPIPE +..... Streams pipe error
ETIME XSR +N4... Stream ioctl timeout
ETIMEDOUT POSIX - +++ss+ Connection timed out
ETOOMANYREFS +++... Too many references: cannot splice
ETXTBSY POSIX +++... Text file busy
EUCLEAN +..... Structure needs cleaning
EUNATCH +..... Protocol driver not attached
EUSERS +++... Too many users
EWOULDBLOCK POSIX +++..+ Operation would block
EXDEV POSIX ++++++ Cross-device link
EXFULL +..... Exchange full
Notations:
F: used in FFmpeg (-: a few times, +: a lot)
SUSv3: Single Unix Specification, version 3
SUSv4: Single Unix Specification, version 4
XSR: XSI STREAMS (obsolete)
OS: availability on some supported operating systems
L: GNU/Linux
B: BSD (F: FreeBSD, N: NetBSD)
M: MacOS X
W: Microsoft Windows (s: emulated with winsock, see libavformat/network.h)
w: Mingw32 (3.17) and Mingw64 (2.0.1)
b: BeOS

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@chapter Expression Evaluation
@c man begin EXPRESSION EVALUATION
When evaluating an arithmetic expression, FFmpeg uses an internal
formula evaluator, implemented through the @file{libavutil/eval.h}
interface.
An expression may contain unary, binary operators, constants, and
functions.
Two expressions @var{expr1} and @var{expr2} can be combined to form
another expression "@var{expr1};@var{expr2}".
@var{expr1} and @var{expr2} are evaluated in turn, and the new
expression evaluates to the value of @var{expr2}.
The following binary operators are available: @code{+}, @code{-},
@code{*}, @code{/}, @code{^}.
The following unary operators are available: @code{+}, @code{-}.
The following functions are available:
@table @option
@item sinh(x)
Compute hyperbolic sine of @var{x}.
@item cosh(x)
Compute hyperbolic cosine of @var{x}.
@item tanh(x)
Compute hyperbolic tangent of @var{x}.
@item sin(x)
Compute sine of @var{x}.
@item cos(x)
Compute cosine of @var{x}.
@item tan(x)
Compute tangent of @var{x}.
@item atan(x)
Compute arctangent of @var{x}.
@item asin(x)
Compute arcsine of @var{x}.
@item acos(x)
Compute arccosine of @var{x}.
@item exp(x)
Compute exponential of @var{x} (with base @code{e}, the Euler's number).
@item log(x)
Compute natural logarithm of @var{x}.
@item abs(x)
Compute absolute value of @var{x}.
@item squish(x)
Compute expression @code{1/(1 + exp(4*x))}.
@item gauss(x)
Compute Gauss function of @var{x}, corresponding to
@code{exp(-x*x/2) / sqrt(2*PI)}.
@item isinf(x)
Return 1.0 if @var{x} is +/-INFINITY, 0.0 otherwise.
@item isnan(x)
Return 1.0 if @var{x} is NAN, 0.0 otherwise.
@item mod(x, y)
Compute the remainder of division of @var{x} by @var{y}.
@item max(x, y)
Return the maximum between @var{x} and @var{y}.
@item min(x, y)
Return the maximum between @var{x} and @var{y}.
@item eq(x, y)
Return 1 if @var{x} and @var{y} are equivalent, 0 otherwise.
@item gte(x, y)
Return 1 if @var{x} is greater than or equal to @var{y}, 0 otherwise.
@item gt(x, y)
Return 1 if @var{x} is greater than @var{y}, 0 otherwise.
@item lte(x, y)
Return 1 if @var{x} is lesser than or equal to @var{y}, 0 otherwise.
@item lt(x, y)
Return 1 if @var{x} is lesser than @var{y}, 0 otherwise.
@item st(var, expr)
Allow to store the value of the expression @var{expr} in an internal
variable. @var{var} specifies the number of the variable where to
store the value, and it is a value ranging from 0 to 9. The function
returns the value stored in the internal variable.
Note, Variables are currently not shared between expressions.
@item ld(var)
Allow to load the value of the internal variable with number
@var{var}, which was previously stored with st(@var{var}, @var{expr}).
The function returns the loaded value.
@item while(cond, expr)
Evaluate expression @var{expr} while the expression @var{cond} is
non-zero, and returns the value of the last @var{expr} evaluation, or
NAN if @var{cond} was always false.
@item ceil(expr)
Round the value of expression @var{expr} upwards to the nearest
integer. For example, "ceil(1.5)" is "2.0".
@item floor(expr)
Round the value of expression @var{expr} downwards to the nearest
integer. For example, "floor(-1.5)" is "-2.0".
@item trunc(expr)
Round the value of expression @var{expr} towards zero to the nearest
integer. For example, "trunc(-1.5)" is "-1.0".
@item sqrt(expr)
Compute the square root of @var{expr}. This is equivalent to
"(@var{expr})^.5".
@item not(expr)
Return 1.0 if @var{expr} is zero, 0.0 otherwise.
@item pow(x, y)
Compute the power of @var{x} elevated @var{y}, it is equivalent to
"(@var{x})^(@var{y})".
@item random(x)
Return a pseudo random value between 0.0 and 1.0. @var{x} is the index of the
internal variable which will be used to save the seed/state.
@item hypot(x, y)
This function is similar to the C function with the same name; it returns
"sqrt(@var{x}*@var{x} + @var{y}*@var{y})", the length of the hypotenuse of a
right triangle with sides of length @var{x} and @var{y}, or the distance of the
point (@var{x}, @var{y}) from the origin.
@item gcd(x, y)
Return the greatest common divisor of @var{x} and @var{y}. If both @var{x} and
@var{y} are 0 or either or both are less than zero then behavior is undefined.
@item if(x, y)
Evaluate @var{x}, and if the result is non-zero return the result of
the evaluation of @var{y}, return 0 otherwise.
@item ifnot(x, y)
Evaluate @var{x}, and if the result is zero return the result of the
evaluation of @var{y}, return 0 otherwise.
@item taylor(expr, x) taylor(expr, x, id)
Evaluate a taylor series at x.
expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified
then 0 is assumed.
note, when you have the derivatives at y instead of 0
taylor(expr, x-y) can be used
When the series does not converge the results are undefined.
@item root(expr, max)
Finds x where f(x)=0 in the interval 0..max.
f() must be continuous or the result is undefined.
@end table
The following constants are available:
@table @option
@item PI
area of the unit disc, approximately 3.14
@item E
exp(1) (Euler's number), approximately 2.718
@item PHI
golden ratio (1+sqrt(5))/2, approximately 1.618
@end table
Assuming that an expression is considered "true" if it has a non-zero
value, note that:
@code{*} works like AND
@code{+} works like OR
and the construct:
@example
if A then B else C
@end example
is equivalent to
@example
if(A,B) + ifnot(A,C)
@end example
In your C code, you can extend the list of unary and binary functions,
and define recognized constants, so that they are available for your
expressions.
The evaluator also recognizes the International System number
postfixes. If 'i' is appended after the postfix, powers of 2 are used
instead of powers of 10. The 'B' postfix multiplies the value for 8,
and can be appended after another postfix or used alone. This allows
using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Follows the list of available International System postfixes, with
indication of the corresponding powers of 10 and of 2.
@table @option
@item y
-24 / -80
@item z
-21 / -70
@item a
-18 / -60
@item f
-15 / -50
@item p
-12 / -40
@item n
-9 / -30
@item u
-6 / -20
@item m
-3 / -10
@item c
-2
@item d
-1
@item h
2
@item k
3 / 10
@item K
3 / 10
@item M
6 / 20
@item G
9 / 30
@item T
12 / 40
@item P
15 / 40
@item E
18 / 50
@item Z
21 / 60
@item Y
24 / 70
@end table
@c man end

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# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -O2 -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= decoding_encoding \
demuxing \
filtering_video \
filtering_audio \
metadata \
muxing \
resampling_audio \
scaling_video \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
decoding_encoding: LDLIBS += -lm
muxing: LDLIBS += -lm
.phony: all clean-test clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

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FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
1) Build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
2) Build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then you can go into the
doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.

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/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavcodec API use example.
*
* Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the
* format handling
* @example doc/examples/decoding_encoding.c
*/
#include <math.h>
#include <libavutil/opt.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/imgutils.h>
#include <libavutil/mathematics.h>
#include <libavutil/samplefmt.h>
#define INBUF_SIZE 4096
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
best_samplerate = FFMAX(*p, best_samplerate);
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channells = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channells) {
best_ch_layout = *p;
best_nb_channells = nb_channels;
}
p++;
}
return best_ch_layout;
}
/*
* Audio encoding example
*/
static void audio_encode_example(const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket pkt;
int i, j, k, ret, got_output;
int buffer_size;
FILE *f;
uint16_t *samples;
float t, tincr;
printf("Encode audio file %s\n", filename);
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* frame containing input raw audio */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* the codec gives us the frame size, in samples,
* we calculate the size of the samples buffer in bytes */
buffer_size = av_samples_get_buffer_size(NULL, c->channels, c->frame_size,
c->sample_fmt, 0);
samples = av_malloc(buffer_size);
if (!samples) {
fprintf(stderr, "Could not allocate %d bytes for samples buffer\n",
buffer_size);
exit(1);
}
/* setup the data pointers in the AVFrame */
ret = avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(const uint8_t*)samples, buffer_size, 0);
if (ret < 0) {
fprintf(stderr, "Could not setup audio frame\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for(i=0;i<200;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
/* encode the samples */
ret = avcodec_encode_audio2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
ret = avcodec_encode_audio2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
fclose(f);
av_freep(&samples);
avcodec_free_frame(&frame);
avcodec_close(c);
av_free(c);
}
/*
* Audio decoding.
*/
static void audio_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int len;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
AVFrame *decoded_frame = NULL;
av_init_packet(&avpkt);
printf("Decode audio file %s to %s\n", filename, outfilename);
/* find the mpeg audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
avpkt.data = inbuf;
avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (avpkt.size > 0) {
int got_frame = 0;
if (!decoded_frame) {
if (!(decoded_frame = avcodec_alloc_frame())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
} else
avcodec_get_frame_defaults(decoded_frame);
len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
if (len < 0) {
fprintf(stderr, "Error while decoding\n");
exit(1);
}
if (got_frame) {
/* if a frame has been decoded, output it */
int data_size = av_samples_get_buffer_size(NULL, c->channels,
decoded_frame->nb_samples,
c->sample_fmt, 1);
fwrite(decoded_frame->data[0], 1, data_size, outfile);
}
avpkt.size -= len;
avpkt.data += len;
avpkt.dts =
avpkt.pts = AV_NOPTS_VALUE;
if (avpkt.size < AUDIO_REFILL_THRESH) {
/* Refill the input buffer, to avoid trying to decode
* incomplete frames. Instead of this, one could also use
* a parser, or use a proper container format through
* libavformat. */
memmove(inbuf, avpkt.data, avpkt.size);
avpkt.data = inbuf;
len = fread(avpkt.data + avpkt.size, 1,
AUDIO_INBUF_SIZE - avpkt.size, f);
if (len > 0)
avpkt.size += len;
}
}
fclose(outfile);
fclose(f);
avcodec_close(c);
av_free(c);
avcodec_free_frame(&decoded_frame);
}
/*
* Video encoding example
*/
static void video_encode_example(const char *filename, int codec_id)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y, got_output;
FILE *f;
AVFrame *frame;
AVPacket pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
printf("Encode video file %s\n", filename);
/* find the mpeg1 video encoder */
codec = avcodec_find_encoder(codec_id);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base= (AVRational){1,25};
c->gop_size = 10; /* emit one intra frame every ten frames */
c->max_b_frames=1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if(codec_id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
/* the image can be allocated by any means and av_image_alloc() is
* just the most convenient way if av_malloc() is to be used */
ret = av_image_alloc(frame->data, frame->linesize, c->width, c->height,
c->pix_fmt, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw picture buffer\n");
exit(1);
}
/* encode 1 second of video */
for(i=0;i<25;i++) {
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
fflush(stdout);
/* prepare a dummy image */
/* Y */
for(y=0;y<c->height;y++) {
for(x=0;x<c->width;x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for(y=0;y<c->height/2;y++) {
for(x=0;x<c->width/2;x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* get the delayed frames */
for (got_output = 1; got_output; i++) {
fflush(stdout);
ret = avcodec_encode_video2(c, &pkt, NULL, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding frame\n");
exit(1);
}
if (got_output) {
printf("Write frame %3d (size=%5d)\n", i, pkt.size);
fwrite(pkt.data, 1, pkt.size, f);
av_free_packet(&pkt);
}
}
/* add sequence end code to have a real mpeg file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_close(c);
av_free(c);
av_freep(&frame->data[0]);
avcodec_free_frame(&frame);
printf("\n");
}
/*
* Video decoding example
*/
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f=fopen(filename,"w");
fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
for(i=0;i<ysize;i++)
fwrite(buf + i * wrap,1,xsize,f);
fclose(f);
}
static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
AVFrame *frame, int *frame_count, AVPacket *pkt, int last)
{
int len, got_frame;
char buf[1024];
len = avcodec_decode_video2(avctx, frame, &got_frame, pkt);
if (len < 0) {
fprintf(stderr, "Error while decoding frame %d\n", *frame_count);
return len;
}
if (got_frame) {
printf("Saving %sframe %3d\n", last ? "last " : "", *frame_count);
fflush(stdout);
/* the picture is allocated by the decoder, no need to free it */
snprintf(buf, sizeof(buf), outfilename, *frame_count);
pgm_save(frame->data[0], frame->linesize[0],
avctx->width, avctx->height, buf);
(*frame_count)++;
}
if (pkt->data) {
pkt->size -= len;
pkt->data += len;
}
return 0;
}
static void video_decode_example(const char *outfilename, const char *filename)
{
AVCodec *codec;
AVCodecContext *c= NULL;
int frame_count;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
AVPacket avpkt;
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
printf("Decode video file %s to %s\n", filename, outfilename);
/* find the mpeg1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
if(codec->capabilities&CODEC_CAP_TRUNCATED)
c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame_count = 0;
for(;;) {
avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
if (avpkt.size == 0)
break;
/* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
and this is the only method to use them because you cannot
know the compressed data size before analysing it.
BUT some other codecs (msmpeg4, mpeg4) are inherently frame
based, so you must call them with all the data for one
frame exactly. You must also initialize 'width' and
'height' before initializing them. */
/* NOTE2: some codecs allow the raw parameters (frame size,
sample rate) to be changed at any frame. We handle this, so
you should also take care of it */
/* here, we use a stream based decoder (mpeg1video), so we
feed decoder and see if it could decode a frame */
avpkt.data = inbuf;
while (avpkt.size > 0)
if (decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 0) < 0)
exit(1);
}
/* some codecs, such as MPEG, transmit the I and P frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
avpkt.data = NULL;
avpkt.size = 0;
decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);
fclose(f);
avcodec_close(c);
av_free(c);
avcodec_free_frame(&frame);
printf("\n");
}
int main(int argc, char **argv)
{
const char *output_type;
/* register all the codecs */
avcodec_register_all();
if (argc < 2) {
printf("usage: %s output_type\n"
"API example program to decode/encode a media stream with libavcodec.\n"
"This program generates a synthetic stream and encodes it to a file\n"
"named test.h264, test.mp2 or test.mpg depending on output_type.\n"
"The encoded stream is then decoded and written to a raw data output.\n"
"output_type must be choosen between 'h264', 'mp2', 'mpg'.\n",
argv[0]);
return 1;
}
output_type = argv[1];
if (!strcmp(output_type, "h264")) {
video_encode_example("test.h264", AV_CODEC_ID_H264);
} else if (!strcmp(output_type, "mp2")) {
audio_encode_example("test.mp2");
audio_decode_example("test.sw", "test.mp2");
} else if (!strcmp(output_type, "mpg")) {
video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
video_decode_example("test%02d.pgm", "test.mpg");
} else {
fprintf(stderr, "Invalid output type '%s', choose between 'h264', 'mp2', or 'mpg'\n",
output_type);
return 1;
}
return 0;
}

340
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/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat demuxing API use example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example doc/examples/demuxing.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static uint8_t **audio_dst_data = NULL;
static int audio_dst_linesize;
static int audio_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int decode_packet(int *got_frame, int cached)
{
int ret = 0;
if (pkt.stream_index == video_stream_idx) {
/* decode video frame */
ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding video frame\n");
return ret;
}
if (*got_frame) {
printf("video_frame%s n:%d coded_n:%d pts:%s\n",
cached ? "(cached)" : "",
video_frame_count++, frame->coded_picture_number,
av_ts2timestr(frame->pts, &video_dec_ctx->time_base));
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
}
} else if (pkt.stream_index == audio_stream_idx) {
/* decode audio frame */
ret = avcodec_decode_audio4(audio_dec_ctx, frame, got_frame, &pkt);
if (ret < 0) {
fprintf(stderr, "Error decoding audio frame\n");
return ret;
}
if (*got_frame) {
printf("audio_frame%s n:%d nb_samples:%d pts:%s\n",
cached ? "(cached)" : "",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
ret = av_samples_alloc(audio_dst_data, &audio_dst_linesize, frame->channels,
frame->nb_samples, frame->format, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio buffer\n");
return AVERROR(ENOMEM);
}
/* TODO: extend return code of the av_samples_* functions so that this call is not needed */
audio_dst_bufsize =
av_samples_get_buffer_size(NULL, frame->channels,
frame->nb_samples, frame->format, 1);
/* copy audio data to destination buffer:
* this is required since rawaudio expects non aligned data */
av_samples_copy(audio_dst_data, frame->data, 0, 0,
frame->nb_samples, frame->channels, frame->format);
/* write to rawaudio file */
fwrite(audio_dst_data[0], 1, audio_dst_bufsize, audio_dst_file);
av_freep(&audio_dst_data[0]);
}
}
return ret;
}
static int open_codec_context(int *stream_idx,
AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
*stream_idx = ret;
st = fmt_ctx->streams[*stream_idx];
/* find decoder for the stream */
dec_ctx = st->codec;
dec = avcodec_find_decoder(dec_ctx->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return ret;
}
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0, got_frame;
if (argc != 4) {
fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n"
"\n", argv[0]);
exit(1);
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* register all formats and codecs */
av_register_all();
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = video_stream->codec;
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
ret = av_image_alloc(video_dst_data, video_dst_linesize,
video_dec_ctx->width, video_dec_ctx->height,
video_dec_ctx->pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
int nb_planes;
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dec_ctx = audio_stream->codec;
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
nb_planes = av_sample_fmt_is_planar(audio_dec_ctx->sample_fmt) ?
audio_dec_ctx->channels : 1;
audio_dst_data = av_mallocz(sizeof(uint8_t *) * nb_planes);
if (!audio_dst_data) {
fprintf(stderr, "Could not allocate audio data buffers\n");
ret = AVERROR(ENOMEM);
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0)
decode_packet(&got_frame, 0);
/* flush cached frames */
pkt.data = NULL;
pkt.size = 0;
do {
decode_packet(&got_frame, 1);
} while (got_frame);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
video_dst_filename);
}
if (audio_stream) {
const char *fmt;
if ((ret = get_format_from_sample_fmt(&fmt, audio_dec_ctx->sample_fmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, audio_dec_ctx->channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
if (video_dec_ctx)
avcodec_close(video_dec_ctx);
if (audio_dec_ctx)
avcodec_close(audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_free(frame);
av_free(video_dst_data[0]);
av_free(audio_dst_data);
return ret < 0;
}

241
project/jni/ffmpeg/doc/examples/filtering_audio.c Normal file
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@@ -0,0 +1,241 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example doc/examples/filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
const char *filter_descr = "aresample=8000,aconvert=s16:mono";
const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
dec_ctx = fmt_ctx->streams[audio_stream_index]->codec;
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
AVFilter *abuffersink = avfilter_get_by_name("ffabuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
AVABufferSinkParams *abuffersink_params;
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
return ret;
}
/* buffer audio sink: to terminate the filter chain. */
abuffersink_params = av_abuffersink_params_alloc();
abuffersink_params->sample_fmts = sample_fmts;
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, abuffersink_params, filter_graph);
av_free(abuffersink_params);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
return 0;
}
static void print_samplesref(AVFilterBufferRef *samplesref)
{
const AVFilterBufferRefAudioProps *props = samplesref->audio;
const int n = props->nb_samples * av_get_channel_layout_nb_channels(props->channel_layout);
const uint16_t *p = (uint16_t*)samplesref->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *samplesref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_audio4(dec_ctx, &frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding audio\n");
continue;
}
if (got_frame) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_buffer_ref(buffersink_ctx, &samplesref, 0);
if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if(ret < 0)
goto end;
if (samplesref) {
print_samplesref(samplesref);
avfilter_unref_bufferp(&samplesref);
}
}
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

248
project/jni/ffmpeg/doc/examples/filtering_video.c Normal file
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@@ -0,0 +1,248 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
* @example doc/examples/filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/avfiltergraph.h>
#include <libavfilter/avcodec.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
const char *filter_descr = "scale=78:24";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
dec_ctx = fmt_ctx->streams[video_stream_index]->codec;
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret;
AVFilter *buffersrc = avfilter_get_by_name("buffer");
AVFilter *buffersink = avfilter_get_by_name("ffbuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
AVBufferSinkParams *buffersink_params;
filter_graph = avfilter_graph_alloc();
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
return ret;
}
/* buffer video sink: to terminate the filter chain. */
buffersink_params = av_buffersink_params_alloc();
buffersink_params->pixel_fmts = pix_fmts;
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, buffersink_params, filter_graph);
av_free(buffersink_params);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
return ret;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
return ret;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
return ret;
return 0;
}
static void display_picref(AVFilterBufferRef *picref, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (picref->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(picref->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = picref->pts;
}
/* Trivial ASCII grayscale display. */
p0 = picref->data[0];
puts("\033c");
for (y = 0; y < picref->video->h; y++) {
p = p0;
for (x = 0; x < picref->video->w; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += picref->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame frame;
int got_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
avcodec_register_all();
av_register_all();
avfilter_register_all();
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
AVFilterBufferRef *picref;
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
avcodec_get_frame_defaults(&frame);
got_frame = 0;
ret = avcodec_decode_video2(dec_ctx, &frame, &got_frame, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error decoding video\n");
break;
}
if (got_frame) {
frame.pts = av_frame_get_best_effort_timestamp(&frame);
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame(buffersrc_ctx, &frame, 0) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered pictures from the filtergraph */
while (1) {
ret = av_buffersink_get_buffer_ref(buffersink_ctx, &picref, 0);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
if (picref) {
display_picref(picref, buffersink_ctx->inputs[0]->time_base);
avfilter_unref_bufferp(&picref);
}
}
}
}
av_free_packet(&packet);
}
end:
avfilter_graph_free(&filter_graph);
if (dec_ctx)
avcodec_close(dec_ctx);
avformat_close_input(&fmt_ctx);
if (ret < 0 && ret != AVERROR_EOF) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "Error occurred: %s\n", buf);
exit(1);
}
exit(0);
}

56
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/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example doc/examples/metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
av_register_all();
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}

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/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format.
* The default codecs are used.
* @example doc/examples/muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/mathematics.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
/* 5 seconds stream duration */
#define STREAM_DURATION 200.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_NB_FRAMES ((int)(STREAM_DURATION * STREAM_FRAME_RATE))
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
static int sws_flags = SWS_BICUBIC;
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;
/* Add an output stream. */
static AVStream *add_stream(AVFormatContext *oc, AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
AVStream *st;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
st = avformat_new_stream(oc, *codec);
if (!st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
st->id = oc->nb_streams-1;
c = st->codec;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
st->id = 1;
c->sample_fmt = AV_SAMPLE_FMT_S16;
c->bit_rate = 64000;
c->sample_rate = 44100;
c->channels = 2;
break;
case AVMEDIA_TYPE_VIDEO:
avcodec_get_context_defaults3(c, *codec);
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
c->time_base.den = STREAM_FRAME_RATE;
c->time_base.num = 1;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= CODEC_FLAG_GLOBAL_HEADER;
return st;
}
/**************************************************************/
/* audio output */
static float t, tincr, tincr2;
static int16_t *samples;
static int audio_input_frame_size;
static void open_audio(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
AVCodecContext *c;
int ret;
c = st->codec;
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
t = 0;
tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
audio_input_frame_size = 10000;
else
audio_input_frame_size = c->frame_size;
samples = av_malloc(audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels);
if (!samples) {
fprintf(stderr, "Could not allocate audio samples buffer\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static void get_audio_frame(int16_t *samples, int frame_size, int nb_channels)
{
int j, i, v;
int16_t *q;
q = samples;
for (j = 0; j < frame_size; j++) {
v = (int)(sin(t) * 10000);
for (i = 0; i < nb_channels; i++)
*q++ = v;
t += tincr;
tincr += tincr2;
}
}
static void write_audio_frame(AVFormatContext *oc, AVStream *st)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame = avcodec_alloc_frame();
int got_packet, ret;
av_init_packet(&pkt);
c = st->codec;
get_audio_frame(samples, audio_input_frame_size, c->channels);
frame->nb_samples = audio_input_frame_size;
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
(uint8_t *)samples,
audio_input_frame_size *
av_get_bytes_per_sample(c->sample_fmt) *
c->channels, 1);
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (!got_packet)
return;
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
if (ret != 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
avcodec_free_frame(&frame);
}
static void close_audio(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(samples);
}
/**************************************************************/
/* video output */
static AVFrame *frame;
static AVPicture src_picture, dst_picture;
static int frame_count;
static void open_video(AVFormatContext *oc, AVCodec *codec, AVStream *st)
{
int ret;
AVCodecContext *c = st->codec;
/* open the codec */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
frame = avcodec_alloc_frame();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* Allocate the encoded raw picture. */
ret = avpicture_alloc(&dst_picture, c->pix_fmt, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate picture: %s\n", av_err2str(ret));
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ret = avpicture_alloc(&src_picture, AV_PIX_FMT_YUV420P, c->width, c->height);
if (ret < 0) {
fprintf(stderr, "Could not allocate temporary picture: %s\n",
av_err2str(ret));
exit(1);
}
}
/* copy data and linesize picture pointers to frame */
*((AVPicture *)frame) = dst_picture;
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVPicture *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static void write_video_frame(AVFormatContext *oc, AVStream *st)
{
int ret;
static struct SwsContext *sws_ctx;
AVCodecContext *c = st->codec;
if (frame_count >= STREAM_NB_FRAMES) {
/* No more frames to compress. The codec has a latency of a few
* frames if using B-frames, so we get the last frames by
* passing the same picture again. */
} else {
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!sws_ctx) {
sws_ctx = sws_getContext(c->width, c->height, AV_PIX_FMT_YUV420P,
c->width, c->height, c->pix_fmt,
sws_flags, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(&src_picture, frame_count, c->width, c->height);
sws_scale(sws_ctx,
(const uint8_t * const *)src_picture.data, src_picture.linesize,
0, c->height, dst_picture.data, dst_picture.linesize);
} else {
fill_yuv_image(&dst_picture, frame_count, c->width, c->height);
}
}
if (oc->oformat->flags & AVFMT_RAWPICTURE) {
/* Raw video case - directly store the picture in the packet */
AVPacket pkt;
av_init_packet(&pkt);
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
pkt.data = dst_picture.data[0];
pkt.size = sizeof(AVPicture);
ret = av_interleaved_write_frame(oc, &pkt);
} else {
/* encode the image */
AVPacket pkt;
int got_output;
av_init_packet(&pkt);
pkt.data = NULL; // packet data will be allocated by the encoder
pkt.size = 0;
ret = avcodec_encode_video2(c, &pkt, frame, &got_output);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
/* If size is zero, it means the image was buffered. */
if (got_output) {
if (c->coded_frame->key_frame)
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
ret = av_interleaved_write_frame(oc, &pkt);
} else {
ret = 0;
}
}
if (ret != 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
frame_count++;
}
static void close_video(AVFormatContext *oc, AVStream *st)
{
avcodec_close(st->codec);
av_free(src_picture.data[0]);
av_free(dst_picture.data[0]);
av_free(frame);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVStream *audio_st, *video_st;
AVCodec *audio_codec, *video_codec;
double audio_pts, video_pts;
int ret, i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
if (argc != 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc) {
return 1;
}
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
video_st = NULL;
audio_st = NULL;
if (fmt->video_codec != AV_CODEC_ID_NONE) {
video_st = add_stream(oc, &video_codec, fmt->video_codec);
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
audio_st = add_stream(oc, &audio_codec, fmt->audio_codec);
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (video_st)
open_video(oc, video_codec, video_st);
if (audio_st)
open_audio(oc, audio_codec, audio_st);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
if (frame)
frame->pts = 0;
for (;;) {
/* Compute current audio and video time. */
if (audio_st)
audio_pts = (double)audio_st->pts.val * audio_st->time_base.num / audio_st->time_base.den;
else
audio_pts = 0.0;
if (video_st)
video_pts = (double)video_st->pts.val * video_st->time_base.num /
video_st->time_base.den;
else
video_pts = 0.0;
if ((!audio_st || audio_pts >= STREAM_DURATION) &&
(!video_st || video_pts >= STREAM_DURATION))
break;
/* write interleaved audio and video frames */
if (!video_st || (video_st && audio_st && audio_pts < video_pts)) {
write_audio_frame(oc, audio_st);
} else {
write_video_frame(oc, video_st);
frame->pts += av_rescale_q(1, video_st->codec->time_base, video_st->time_base);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (video_st)
close_video(oc, video_st);
if (audio_st)
close_audio(oc, audio_st);
/* Free the streams. */
for (i = 0; i < oc->nb_streams; i++) {
av_freep(&oc->streams[i]->codec);
av_freep(&oc->streams[i]);
}
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_close(oc->pb);
/* free the stream */
av_free(oc);
return 0;
}

223
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/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example doc/examples/resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int alloc_samples_array_and_data(uint8_t ***data, int *linesize, int nb_channels,
int nb_samples, enum AVSampleFormat sample_fmt, int align)
{
int nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
*data = av_malloc(sizeof(*data) * nb_planes);
if (!*data)
return AVERROR(ENOMEM);
return av_samples_alloc(*data, linesize, nb_channels,
nb_samples, sample_fmt, align);
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = alloc_samples_array_and_data(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = alloc_samples_array_and_data(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_free(dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
if (dst_file)
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

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/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example doc/examples/scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignmnet */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
if (dst_file)
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

554
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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg FAQ
@titlepage
@center @titlefont{FFmpeg FAQ}
@end titlepage
@top
@contents
@chapter General Questions
@section Why doesn't FFmpeg support feature [xyz]?
Because no one has taken on that task yet. FFmpeg development is
driven by the tasks that are important to the individual developers.
If there is a feature that is important to you, the best way to get
it implemented is to undertake the task yourself or sponsor a developer.
@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?
No. Windows DLLs are not portable, bloated and often slow.
Moreover FFmpeg strives to support all codecs natively.
A DLL loader is not conducive to that goal.
@section I cannot read this file although this format seems to be supported by ffmpeg.
Even if ffmpeg can read the container format, it may not support all its
codecs. Please consult the supported codec list in the ffmpeg
documentation.
@section Which codecs are supported by Windows?
Windows does not support standard formats like MPEG very well, unless you
install some additional codecs.
The following list of video codecs should work on most Windows systems:
@table @option
@item msmpeg4v2
.avi/.asf
@item msmpeg4
.asf only
@item wmv1
.asf only
@item wmv2
.asf only
@item mpeg4
Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
@item mpeg1video
.mpg only
@end table
Note, ASF files often have .wmv or .wma extensions in Windows. It should also
be mentioned that Microsoft claims a patent on the ASF format, and may sue
or threaten users who create ASF files with non-Microsoft software. It is
strongly advised to avoid ASF where possible.
The following list of audio codecs should work on most Windows systems:
@table @option
@item adpcm_ima_wav
@item adpcm_ms
@item pcm_s16le
always
@item libmp3lame
If some MP3 codec like LAME is installed.
@end table
@chapter Compilation
@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'}
This is a bug in gcc. Do not report it to us. Instead, please report it to
the gcc developers. Note that we will not add workarounds for gcc bugs.
Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
@url{http://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@chapter Usage
@section ffmpeg does not work; what is wrong?
Try a @code{make distclean} in the ffmpeg source directory before the build.
If this does not help see
(@url{http://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
First, rename your pictures to follow a numerical sequence.
For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
@example
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example
Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
Use the @option{-start_number} option to declare a starting number for
the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
example will start with @file{img100.jpg}:
@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
You can also use @command{cat} to pipe images to ffmpeg:
@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
Use:
@example
ffmpeg -i movie.mpg movie%d.jpg
@end example
The @file{movie.mpg} used as input will be converted to
@file{movie1.jpg}, @file{movie2.jpg}, etc...
Instead of relying on file format self-recognition, you may also use
@table @option
@item -c:v ppm
@item -c:v png
@item -c:v mjpeg
@end table
to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@section Why do I see a slight quality degradation with multithreaded MPEG* encoding?
For multithreaded MPEG* encoding, the encoded slices must be independent,
otherwise thread n would practically have to wait for n-1 to finish, so it's
quite logical that there is a small reduction of quality. This is not a bug.
@section How can I read from the standard input or write to the standard output?
Use @file{-} as file name.
@section -f jpeg doesn't work.
Try '-f image2 test%d.jpg'.
@section Why can I not change the frame rate?
Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v command line option.
@section How do I encode Xvid or DivX video with ffmpeg?
Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
standard (note that there are many other coding formats that use this
same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
force the fourcc 'xvid' to be stored as the video fourcc rather than the
default.
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
but beware the '-g 100' might cause problems with some decoders.
Things to try: '-bf 2', '-flags qprd', '-flags mv0', '-flags skiprd.
@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced
material, and try '-top 0/1' if the result looks really messed-up.
@section How can I read DirectShow files?
If you have built FFmpeg with @code{./configure --enable-avisynth}
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
@example
ffmpeg -i input.avs
@end example
For ANY other help on Avisynth, please visit the
@uref{http://www.avisynth.org/, Avisynth homepage}.
@section How can I join video files?
To "join" video files is quite ambiguous. The following list explains the
different kinds of "joining" and points out how those are addressed in
FFmpeg. To join video files may mean:
@itemize
@item
To put them one after the other: this is called to @emph{concatenate} them
(in short: concat) and is addressed
@ref{How can I concatenate video files, in this very faq}.
@item
To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
@emph{multiplex} them together (in short: mux), and is done by simply
invoking ffmpeg with several @option{-i} options.
@item
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{http://ffmpeg.org/ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{http://ffmpeg.org/ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@anchor{How can I concatenate video files}
@section How can I concatenate video files?
There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{http://www.ffmpeg.org/ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
video by merely concatenating the files them.
Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble @code{cat} command (or the
equally humble @code{copy} under Windows), and finally transcoding back to your
format of choice.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
Additionally, you can use the @code{concat} protocol instead of @code{cat} or
@code{copy} which will avoid creation of a potentially huge intermediate file.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
Note that you may need to escape the character "|" which is special for many
shells.
Another option is usage of named pipes, should your platform support it:
@example
mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -acodec libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
For example, let's say we want to concatenate two FLV files into an
output.flv file:
@example
mkfifo temp1.a
mkfifo temp1.v
mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -acodec pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -acodec pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
@end example
@section -profile option fails when encoding H.264 video with AAC audio
@command{ffmpeg} prints an error like
@example
Undefined constant or missing '(' in 'baseline'
Unable to parse option value "baseline"
Error setting option profile to value baseline.
@end example
Short answer: write @option{-profile:v} instead of @option{-profile}.
Long answer: this happens because the @option{-profile} option can apply to both
video and audio. Specifically the AAC encoder also defines some profiles, none
of which are named @var{baseline}.
The solution is to apply the @option{-profile} option to the video stream only
by using @url{http://ffmpeg.org/ffmpeg.html#Stream-specifiers-1, Stream specifiers}.
Appending @code{:v} to it will do exactly that.
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
lost.
Most likely, it is through @code{auto-inserted aconvert}. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as @option{-f lavfi} currently
only support packed S16.
Then insert the correct @code{aconvert} explicitly in the filter graph,
specifying the exact format.
@example
aconvert=s16:stereo:packed
@end example
@section Why does FFmpeg not see the subtitles in my VOB file?
VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initally detected.
Some applications, including the @code{ffmpeg} command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.
The size of the initial scan is controlled by two options: @code{probesize}
(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
The @option{-sameq} option meant "same quantizer", and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
"same quality" and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
Yes. Check the @file{doc/examples} directory in the source
repository, also available online at:
@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
Examples are also installed by default, usually in
@code{$PREFIX/share/ffmpeg/examples}.
Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
examine the source code for one of the many open source projects that
already incorporate FFmpeg at (@url{projects.html}).
@section Can you support my C compiler XXX?
It depends. If your compiler is C99-compliant, then patches to support
it are likely to be welcome if they do not pollute the source code
with @code{#ifdef}s related to the compiler.
@section Is Microsoft Visual C++ supported?
Yes. Please see the @uref{platform.html, Microsoft Visual C++}
section in the FFmpeg documentation.
@section Can you add automake, libtool or autoconf support?
No. These tools are too bloated and they complicate the build.
@section Why not rewrite FFmpeg in object-oriented C++?
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{http://www.tux.org/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?
The build process creates ffmpeg_g, ffplay_g, etc. which contain full debug
information. Those binaries are stripped to create ffmpeg, ffplay, etc. If
you need the debug information, use the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
Yes, as long as the code is optional and can easily and cleanly be placed
under #if CONFIG_GPL without breaking anything. So, for example, a new codec
or filter would be OK under GPL while a bug fix to LGPL code would not.
@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: @code{-lavdevice} must come before
@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.
An easy way to get the full list of required libraries in dependency order
is to use @code{pkg-config}.
@example
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example
See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
more details.
@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
FFmpeg is a pure C project, so to use the libraries within your C++ application
you need to explicitly state that you are using a C library. You can do this by
encompassing your FFmpeg includes using @code{extern "C"}.
See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
FFmpeg is a pure C project using C99 math features, in order to enable C++
to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
You have to create a custom AVIOContext using @code{avio_alloc_context},
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
@section Where can I find libav* headers for Pascal/Delphi?
see @url{http://www.iversenit.dk/dev/ffmpeg-headers/}
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
see @url{http://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
Even if peculiar since it is network oriented, RTP is a container like any
other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
In this specific case please look at RFC 4629 to see how it should be done.
@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
r_frame_rate is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then r_frame_rate
will be 150.
@section Why is @code{make fate} not running all tests?
Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
or @code{FATE_SAMPLES} environment variable or the @code{--samples}
@command{configure} option is set to the right path.
@section Why is @code{make fate} not finding the samples?
Do you happen to have a @code{~} character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace @code{~} by the full path.
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Automated Testing Environment
@titlepage
@center @titlefont{FFmpeg Automated Testing Environment}
@end titlepage
@node Top
@top
@contents
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with there recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@example
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
`--samples=<path to the samples directory>'. Afterwards you can
invoke the makefile targets without setting the SAMPLES makefile
variable. This is illustrated by the following commands:
@example
./configure --samples=fate-suite/
make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
@example
FATE_SAMPLES=fate-suite/ make fate
@end example
@float NOTE
Do not put a '~' character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
@end float
To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@example
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The `slot' configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
itself will be sourced in a shell script, therefore all shell features may
be used. This enables you to setup the environment as you need it for your
build.
For your first test runs the `fate_recv' variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@itemize
@item configure.log
@item compile.log
@item test.log
@item report
@item version
@end itemize
When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
b1:31:c8:79:3f:04:1d:f8:f2:23:26:5a:fd:55:fa:92
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter FATE makefile targets and variables
@section Makefile targets
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @option
@item V
Verbosity level, can be set to 0, 1 or 2.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@item 2: show everything
@end itemize
@item SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@item THREAD_TYPE
Specify which threading strategy test, either @var{slice} or @var{frame},
by default @var{slice+frame}
@item CPUFLAGS
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@end table
@section Examples
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
@end example

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slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
comment= # optional description
# the following are optional and map to configure options
arch=
cpu=
cross_prefix=
cc=
target_os=
sysroot=
target_exec=
target_path=
extra_cflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above
#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Bitstream Filters Documentation
@titlepage
@center @titlefont{FFmpeg Bitstream Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the bitstream filters provided by the
libavcodec library.
A bitstream filter operates on the encoded stream data, and performs
bitstream level modifications without performing decoding.
@c man end DESCRIPTION
@include bitstream_filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-bitstream-filters
@settitle FFmpeg bitstream filters
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Devices Documentation
@titlepage
@center @titlefont{FFmpeg Devices Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the input and output devices provided by the
libavdevice library.
@c man end DESCRIPTION
@chapter Device Options
@c man begin DEVICE OPTIONS
The libavdevice library provides the same interface as
libavformat. Namely, an input device is considered like a demuxer, and
an output device like a muxer, and the interface and generic device
options are the same provided by libavformat (see the ffmpeg-formats
manual).
In addition each input or output device may support so-called private
options, which are specific for that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the device
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
@c man end DEVICE OPTIONS
@include indevs.texi
@include outdevs.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavdevice(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-devices
@settitle FFmpeg devices
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Filters Documentation
@titlepage
@center @titlefont{FFmpeg Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes filters, sources, and sinks provided by the
libavfilter library.
@c man end DESCRIPTION
@include filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavfilter(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-filters
@settitle FFmpeg filters
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Formats Documentation
@titlepage
@center @titlefont{FFmpeg Formats Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the supported formats (muxers and demuxers)
provided by the libavformat library.
@c man end DESCRIPTION
@chapter Format Options
@c man begin FORMAT OPTIONS
The libavformat library provides some generic global options, which
can be set on all the muxers and demuxers. In addition each muxer or
demuxer may support so-called private options, which are specific for
that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
The list of supported options follows:
@table @option
@item avioflags @var{flags} (@emph{input/output})
Possible values:
@table @samp
@item direct
Reduce buffering.
@end table
@item probesize @var{integer} (@emph{input})
Set probing size in bytes, i.e. the size of the data to analyze to get
stream information. A higher value will allow to detect more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags} (@emph{input/output})
Set format flags.
Possible values:
@table @samp
@item ignidx
Ignore index.
@item genpts
Generate PTS.
@item nofillin
Do not fill in missing values that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item igndts
Ignore DTS.
@item discardcorrupt
Discard corrupted frames.
@item sortdts
Try to interleave output packets by DTS.
@item keepside
Do not merge side data.
@item latm
Enable RTP MP4A-LATM payload.
@item nobuffer
Reduce the latency introduced by optional buffering
@end table
@item analyzeduration @var{integer} (@emph{input})
Specify how many microseconds are analyzed to estimate duration.
@item cryptokey @var{hexadecimal string} (@emph{input})
Set decryption key.
@item indexmem @var{integer} (@emph{input})
Set max memory used for timestamp index (per stream).
@item rtbufsize @var{integer} (@emph{input})
Set max memory used for buffering real-time frames.
@item fdebug @var{flags} (@emph{input/output})
Print specific debug info.
Possible values:
@table @samp
@item ts
@end table
@item max_delay @var{integer} (@emph{input/output})
Set maximum muxing or demuxing delay in microseconds.
@item fpsprobesize @var{integer} (@emph{input})
Set number of frames used to probe fps.
@item audio_preload @var{integer} (@emph{output})
Set microseconds by which audio packets should be interleaved earlier.
@item chunk_duration @var{integer} (@emph{output})
Set microseconds for each chunk.
@item chunk_size @var{integer} (@emph{output})
Set size in bytes for each chunk.
@item err_detect, f_err_detect @var{flags} (@emph{input})
Set error detection flags. @code{f_err_detect} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item crccheck
Verify embedded CRCs.
@item bitstream
Detect bitstream specification deviations.
@item buffer
Detect improper bitstream length.
@item explode
Abort decoding on minor error detection.
@item careful
Consider things that violate the spec and have not been seen in the
wild as errors.
@item compliant
Consider all spec non compliancies as errors.
@item aggressive
Consider things that a sane encoder should not do as an error.
@end table
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
Use wallclock as timestamps.
@item avoid_negative_ts @var{integer} (@emph{output})
Shift timestamps to make them positive. 1 enables, 0 disables, default
of -1 enables when required by target format.
@end table
@c man end FORMAT OPTIONS
@include demuxers.texi
@include muxers.texi
@include metadata.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-formats
@settitle FFmpeg formats
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Protocols Documentation
@titlepage
@center @titlefont{FFmpeg Protocols Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the input and output protocols provided by the
libavformat library.
@c man end DESCRIPTION
@include protocols.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-protocols
@settitle FFmpeg protocols
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Resampler Documentation
@titlepage
@center @titlefont{FFmpeg Resampler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The FFmpeg resampler provides an high-level interface to the
libswresample library audio resampling utilities. In particular it
allows to perform audio resampling, audio channel layout rematrixing,
and convert audio format and packing layout.
@c man end DESCRIPTION
@chapter Resampler Options
@c man begin RESAMPLER OPTIONS
The audio resampler supports the following named options.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, @var{option}=@var{value} for the aresample filter,
by setting the value explicitly in the
@code{SwrContext} options or using the @file{libavutil/opt.h} API for
programmatic use.
@table @option
@item ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{in_channel_layout} is set.
@item och, out_channel_count
Set the number of output channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{out_channel_layout} is set.
@item uch, used_channel_count
Set the number of used channels. Default value is 0. This option is
only used for special remapping.
@item isr, in_sample_rate
Set the input sample rate. Default value is 0.
@item osr, out_sample_rate
Set the output sample rate. Default value is 0.
@item isf, in_sample_fmt
Specify the input sample format. It is set by default to @code{none}.
@item osf, out_sample_fmt
Specify the output sample format. It is set by default to @code{none}.
@item tsf, internal_sample_fmt
Set the internal sample format. Default value is @code{none}.
@item icl, in_channel_layout
Set the input channel layout.
@item ocl, out_channel_layout
Set the output channel layout.
@item clev, center_mix_level
Set center mix level. It is a value expressed in deciBel, and must be
inclusively included between -32 and +32.
@item slev, surround_mix_level
Set surround mix level. It is a value expressed in deciBel, and must
be inclusively included between -32 and +32.
@item lfe_mix_evel
Set LFE mix level.
@item rmvol, rematrix_volume
Set rematrix volume. Default value is 1.0.
@item flags, swr_flags
Set flags used by the converter. Default value is 0.
It supports the following individual flags:
@table @option
@item res
force resampling
@end table
@item dither_scale
Set the dither scale. Default value is 1.
@item dither_method
Set dither method. Default value is 0.
Supported values:
@table @samp
@item rectangular
select rectangular dither
@item triangular
select triangular dither
@item triangular_hp
select triangular dither with high pass
@end table
@item resampler
Set resampling engine. Default value is swr.
Supported values:
@table @samp
@item swr
select the native SW Resampler; filter options precision and cheby are not
applicable in this case.
@item soxr
select the SoX Resampler (where available); compensation, and filter options
filter_size, phase_shift, filter_type & kaiser_beta, are not applicable in this
case.
@end table
@item filter_size
For swr only, set resampling filter size, default value is 32.
@item phase_shift
For swr only, set resampling phase shift, default value is 10, must be included
between 0 and 30.
@item linear_interp
Use Linear Interpolation if set to 1, default value is 0.
@item cutoff
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
@item precision
For soxr only, the precision in bits to which the resampled signal will be
calculated. The default value of 20 (which, with suitable dithering, is
appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
value of 28 gives SoX's 'Very High Quality'.
@item cheby
For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
approximation for 'irrational' ratios. Default value is 0.
@item async
For swr only, simple 1 parameter audio sync to timestamps using stretching,
squeezing, filling and trimming. Setting this to 1 will enable filling and
trimming, larger values represent the maximum amount in samples that the data
may be stretched or squeezed for each second.
Default value is 0, thus no compensation is applied to make the samples match
the audio timestamps.
@item min_comp
For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger stretching/squeezing/filling or trimming of the
data to make it match the timestamps. The default is that
stretching/squeezing/filling and trimming is disabled
(@option{min_comp} = @code{FLT_MAX}).
@item min_hard_comp
For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger adding/dropping samples to make it match the
timestamps. This option effectively is a threshold to select between
hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
all compensation is by default disabled through @option{min_comp}.
The default is 0.1.
@item comp_duration
For swr only, set duration (in seconds) over which data is stretched/squeezed
to make it match the timestamps. Must be a non-negative double float value,
default value is 1.0.
@item max_soft_comp
For swr only, set maximum factor by which data is stretched/squeezed to make it
match the timestamps. Must be a non-negative double float value, default value
is 0.
@item matrix_encoding
Select matrixed stereo encoding.
It accepts the following values:
@table @samp
@item none
select none
@item dolby
select Dolby
@item dplii
select Dolby Pro Logic II
@end table
Default value is @code{none}.
@item filter_type
For swr only, select resampling filter type. This only affects resampling
operations.
It accepts the following values:
@table @samp
@item cubic
select cubic
@item blackman_nuttall
select Blackman Nuttall Windowed Sinc
@item kaiser
select Kaiser Windowed Sinc
@end table
@item kaiser_beta
For swr only, set Kaiser Window Beta value. Must be an integer included between
2 and 16, default value is 9.
@end table
@c man end RESAMPLER OPTIONS
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswresample(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-resampler
@settitle FFmpeg Resampler
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Scaler Documentation
@titlepage
@center @titlefont{FFmpeg Scaler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The FFmpeg rescaler provides an high-level interface to the libswscale
library image conversion utilities. In particular it allows to perform
image rescaling and pixel format conversion.
@c man end DESCRIPTION
@chapter Scaler Options
@c man begin SCALER OPTIONS
The video scaler supports the following named options.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools. For programmatic use, they can be set explicitly in the
@code{SwsContext} options or through the @file{libavutil/opt.h} API.
@table @option
@item sws_flags
Set the scaler flags. This is also used to set the scaling
algorithm. Only a single algorithm should be selected.
It accepts the following values:
@table @samp
@item fast_bilinear
Select fast bilinear scaling algorithm.
@item bilinear
Select bilinear scaling algorithm.
@item bicubic
Select bicubic scaling algorithm.
@item experimental
Select experimental scaling algorithm.
@item neighbor
Select nearest neighbor rescaling algorithm.
@item area
Select averaging area rescaling algorithm.
@item bicubiclin
Select bicubic scaling algorithm for the luma component, bilinear for
chroma components.
@item gauss
Select Gaussian rescaling algorithm.
@item sinc
Select sinc rescaling algorithm.
@item lanczos
Select lanczos rescaling algorithm.
@item spline
Select natural bicubic spline rescaling algorithm.
@item print_info
Enable printing/debug logging.
@item accurate_rnd
Enable accurate rounding.
@item full_chroma_int
Enable full chroma interpolation.
@item full_chroma_inp
Select full chroma input.
@item bitexact
Enable bitexact output.
@end table
@item srcw
Set source width.
@item srch
Set source height.
@item dstw
Set destination width.
@item dsth
Set destination height.
@item src_format
Set source pixel format (must be expressed as an integer).
@item dst_format
Set destination pixel format (must be expressed as an integer).
@item src_range
Select source range.
@item dst_range
Select destination range.
@item param0, param1
Set scaling algorithm parameters. The specified values are specific of
some scaling algorithms and ignored by others. The specified values
are floating point number values.
@end table
@c man end SCALER OPTIONS
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libswscale(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-scaler
@settitle FFmpeg video scaling and pixel format converter
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle FFmpeg Utilities Documentation
@titlepage
@center @titlefont{FFmpeg Utilities Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes some generic features and utilities provided
by the libavutil library.
@c man end DESCRIPTION
@include syntax.texi
@include eval.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-utils
@settitle FFmpeg utilities
@end ignore
@bye

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47
project/jni/ffmpeg/doc/ffmpeg.txt Normal file
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:
ffmpeg.c : libav*
======== : ======
:
:
--------------------------------:---> AVStream...
InputStream input_streams[] / :
/ :
InputFile input_files[] +==========================+ / ^ :
------> 0 | : st ---:-----------:--/ : :
^ +------+-----------+-----+ / +--------------------------+ : :
: | :ist_index--:-----:---------/ 1 | : st : | : :
: +------+-----------+-----+ +==========================+ : :
nb_input_files : | :ist_index--:-----:------------------> 2 | : st : | : :
: +------+-----------+-----+ +--------------------------+ : nb_input_streams :
: | :ist_index : | 3 | ... | : :
v +------+-----------+-----+ +--------------------------+ : :
--> 4 | | : :
| +--------------------------+ : :
| 5 | | : :
| +==========================+ v :
| :
| :
| :
| :
--------- --------------------------------:---> AVStream...
\ / :
OutputStream output_streams[] / :
\ / :
+======\======================/======+ ^ :
------> 0 | : source_index : st-:--- | : :
OutputFile output_files[] / +------------------------------------+ : :
/ 1 | : : : | : :
^ +------+------------+-----+ / +------------------------------------+ : :
: | : ost_index -:-----:------/ 2 | : : : | : :
nb_output_files : +------+------------+-----+ +====================================+ : :
: | : ost_index -:-----|-----------------> 3 | : : : | : :
: +------+------------+-----+ +------------------------------------+ : nb_output_streams :
: | : : | 4 | | : :
: +------+------------+-----+ +------------------------------------+ : :
: | : : | 5 | | : :
v +------+------------+-----+ +------------------------------------+ : :
6 | | : :
+------------------------------------+ : :
7 | | : :
+====================================+ v :
:

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\input texinfo @c -*- texinfo -*-
@settitle ffplay Documentation
@titlepage
@center @titlefont{ffplay Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffplay [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
FFplay is a very simple and portable media player using the FFmpeg
libraries and the SDL library. It is mostly used as a testbed for the
various FFmpeg APIs.
@c man end
@chapter Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@item -x @var{width}
Force displayed width.
@item -y @var{height}
Force displayed height.
@item -s @var{size}
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -an
Disable audio.
@item -vn
Disable video.
@item -ss @var{pos}
Seek to a given position in seconds.
@item -t @var{duration}
play <duration> seconds of audio/video
@item -bytes
Seek by bytes.
@item -nodisp
Disable graphical display.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
Set the show mode to use.
Available values for @var{mode} are:
@table @samp
@item 0, video
show video
@item 1, waves
show audio waves
@item 2, rdft
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
@end table
Default value is "video", if video is not present or cannot be played
"rdft" is automatically selected.
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@item -vf @var{filter_graph}
@var{filter_graph} is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
also sources and sinks).
@item -i @var{input_file}
Read @var{input_file}.
@end table
@section Advanced options
@table @option
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Show the stream duration, the codec parameters, the current position in
the stream and the audio/video synchronisation drift.
@item -bug
Work around bugs.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
Generate pts.
@item -rtp_tcp
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
if you are streaming with the RTSP protocol.
@item -sync @var{type}
Set the master clock to audio (@code{type=audio}), video
(@code{type=video}) or external (@code{type=ext}). Default is audio. The
master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
@item -threads @var{count}
Set the thread count.
@item -ast @var{audio_stream_number}
Select the desired audio stream number, counting from 0. The number
refers to the list of all the input audio streams. If it is greater
than the number of audio streams minus one, then the last one is
selected, if it is negative the audio playback is disabled.
@item -vst @var{video_stream_number}
Select the desired video stream number, counting from 0. The number
refers to the list of all the input video streams. If it is greater
than the number of video streams minus one, then the last one is
selected, if it is negative the video playback is disabled.
@item -sst @var{subtitle_stream_number}
Select the desired subtitle stream number, counting from 0. The number
refers to the list of all the input subtitle streams. If it is greater
than the number of subtitle streams minus one, then the last one is
selected, if it is negative the subtitle rendering is disabled.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{media_specifier} @var{codec_name}
Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), and @code{s} subtitle.
@item -acodec @var{codec_name}
Force a specific audio decoder.
@item -vcodec @var{codec_name}
Force a specific video decoder.
@item -scodec @var{codec_name}
Force a specific subtitle decoder.
@end table
@section While playing
@table @key
@item q, ESC
Quit.
@item f
Toggle full screen.
@item p, SPC
Pause.
@item a
Cycle audio channel.
@item v
Cycle video channel.
@item t
Cycle subtitle channel.
@item w
Show audio waves.
@item left/right
Seek backward/forward 10 seconds.
@item down/up
Seek backward/forward 1 minute.
@item page down/page up
Seek backward/forward 10 minutes.
@item mouse click
Seek to percentage in file corresponding to fraction of width.
@end table
@c man end
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
ffmpeg(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle ffprobe Documentation
@titlepage
@center @titlefont{ffprobe Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffprobe [@var{options}] [@file{input_file}]
@chapter Description
@c man begin DESCRIPTION
ffprobe gathers information from multimedia streams and prints it in
human- and machine-readable fashion.
For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a filename is specified in input, ffprobe will try to open and
probe the file content. If the file cannot be opened or recognized as
a multimedia file, a positive exit code is returned.
ffprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by ffprobe or
for specifying which information to display, and for setting how
ffprobe will show it.
ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{print_format} option.
Sections may contain other nested sections, and are identified by a
name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT" or "STREAM" section.
@c man end
@chapter Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@item -f @var{format}
Force format to use.
@item -unit
Show the unit of the displayed values.
@item -prefix
Use SI prefixes for the displayed values.
Unless the "-byte_binary_prefix" option is used all the prefixes
are decimal.
@item -byte_binary_prefix
Force the use of binary prefixes for byte values.
@item -sexagesimal
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
@item -pretty
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -of, -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@var{writer_options} specifies the options to be passed to the writer.
For example for printing the output in JSON format, specify:
@example
-print_format json
@end example
For more details on the available output printing formats, see the
Writers section below.
@item -sections
Print sections structure and section information, and exit. The output
is not meant to be parsed by a machine.
@item -select_streams @var{stream_specifier}
Select only the streams specified by @var{stream_specifier}. This
option affects only the options related to streams
(e.g. @code{show_streams}, @code{show_packets}, etc.).
For example to show only audio streams, you can use the command:
@example
ffprobe -show_streams -select_streams a INPUT
@end example
To show only video packets belonging to the video stream with index 1:
@example
ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as an hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.
The dump is printed as the "data" field. It may contain newlines.
@item -show_error
Show information about the error found when trying to probe the input.
The error information is printed within a section with name "ERROR".
@item -show_format
Show information about the container format of the input multimedia
stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_format_entry @var{name}
Like @option{-show_format}, but only prints the specified entry of the
container format information, rather than all. This option may be given more
than once, then all specified entries will be shown.
This option is deprecated, use @code{show_entries} instead.
@item -show_entries @var{section_entries}
Set list of entries to show.
Entries are specified according to the following
syntax. @var{section_entries} contains a list of section entries
separated by @code{:}. Each section entry is composed by a section
name (or unique name), optionally followed by a list of entries local
to that section, separated by @code{,}.
If section name is specified but is followed by no @code{=}, all
entries are printed to output, together with all the contained
sections. Otherwise only the entries specified in the local section
entries list are printed. In particular, if @code{=} is specified but
the list of local entries is empty, then no entries will be shown for
that section.
Note that the order of specification of the local section entries is
not honored in the output, and the usual display order will be
retained.
The formal syntax is given by:
@example
@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
@end example
For example, to show only the index and type of each stream, and the PTS
time, duration time, and stream index of the packets, you can specify
the argument:
@example
packet=pts_time,duration_time,stream_index : stream=index,codec_type
@end example
To show all the entries in the section "format", but only the codec
type in the section "stream", specify the argument:
@example
format : stream=codec_type
@end example
To show all the tags in the stream and format sections:
@example
format_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
sections:
@example
stream_tags=title
@end example
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame contained in the input multimedia
stream.
The information for each single frame is printed within a dedicated
section with name "FRAME".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -count_frames
Count the number of frames per stream and report it in the
corresponding stream section.
@item -count_packets
Count the number of packets per stream and report it in the
corresponding stream section.
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
This option is enabled by default, but you may need to disable it
for specific uses, for example when creating XSD-compliant XML output.
@item -show_program_version
Show information related to program version.
Version information is printed within a section with name
"PROGRAM_VERSION".
@item -show_library_versions
Show information related to library versions.
Version information for each library is printed within a section with
name "LIBRARY_VERSION".
@item -show_versions
Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@item -i @var{input_file}
Read @var{input_file}.
@end table
@c man end
@chapter Writers
@c man begin WRITERS
A writer defines the output format adopted by @command{ffprobe}, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options
to adopt. The options are specified as a list of @var{key}=@var{value}
pairs, separated by ":".
A description of the currently available writers follows.
@section default
Default format.
Print each section in the form:
@example
[SECTION]
key1=val1
...
keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT or
STREAM section, and are prefixed by the string "TAG:".
A description of the accepted options follows.
@table @option
@item nokey, nk
If set to 1 specify not to print the key of each field. Default value
is 0.
@item noprint_wrappers, nw
If set to 1 specify not to print the section header and footer.
Default value is 0.
@end table
@section compact, csv
Compact and CSV format.
The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
Each section is printed on a single line.
If no option is specifid, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
Metadata tags are printed in the corresponding "format" or "stream"
section. A metadata tag key, if printed, is prefixed by the string
"tag:".
The description of the accepted options follows.
@table @option
@item item_sep, s
Specify the character to use for separating fields in the output line.
It must be a single printable character, it is "|" by default ("," for
the @code{csv} writer).
@item nokey, nk
If set to 1 specify not to print the key of each field. Its default
value is 0 (1 for the @code{csv} writer).
@item escape, e
Set the escape mode to use, default to "c" ("csv" for the @code{csv}
writer).
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline ('\n'), carriage
return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
escaping, so that a newline is converted to the sequence "\n", a
carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
converted to "\@var{SEP}".
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline ('\n'), a carriage return ('\r'), a double quote
('"'), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@end table
@item print_section, p
Print the section name at the begin of each line if the value is
@code{1}, disable it with value set to @code{0}. Default value is
@code{1}.
@end table
@section flat
Flat format.
A free-form output where each line contains an explicit key=value, such as
"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
directly embedded in sh scripts as long as the separator character is an
alphanumeric character or an underscore (see @var{sep_char} option).
The description of the accepted options follows.
@table @option
@item sep_char, s
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is '.'.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
@section ini
INI format output.
Print output in an INI based format.
The following conventions are adopted:
@itemize
@item
all key and values are UTF-8
@item
'.' is the subgroup separator
@item
newline, '\t', '\f', '\b' and the following characters are escaped
@item
'\' is the escape character
@item
'#' is the comment indicator
@item
'=' is the key/value separator
@item
':' is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by ":".
The description of the accepted options follows.
@table @option
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
@section json
JSON based format.
Each section is printed using JSON notation.
The description of the accepted options follows.
@table @option
@item compact, c
If set to 1 enable compact output, that is each section will be
printed on a single line. Default value is 0.
@end table
For more information about JSON, see @url{http://www.json.org/}.
@section xml
XML based format.
The XML output is described in the XML schema description file
@file{ffprobe.xsd} installed in the FFmpeg datadir.
An updated version of the schema can be retrieved at the url
@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
latest schema committed into the FFmpeg development source code tree.
Note that the output issued will be compliant to the
@file{ffprobe.xsd} schema only when no special global output options
(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.
The description of the accepted options follows.
@table @option
@item fully_qualified, q
If set to 1 specify if the output should be fully qualified. Default
value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_compliant, x
If set to 1 perform more checks for ensuring that the output is XSD
compliant. Default value is 0.
This option automatically sets @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{http://www.w3.org/XML/}.
@c man end WRITERS
@chapter Timecode
@c man begin TIMECODE
@command{ffprobe} supports Timecode extraction:
@itemize
@item
MPEG1/2 timecode is extracted from the GOP, and is available in the video
stream details (@option{-show_streams}, see @var{timecode}).
@item
MOV timecode is extracted from tmcd track, so is available in the tmcd
stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
@item
DV, GXF and AVI timecodes are available in format metadata
(@option{-show_format}, see @var{TAG:timecode}).
@end itemize
@c man end TIMECODE
@chapter See Also
@ifhtml
@url{ffplay.html,ffmpeg}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffserver(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffprobe
@settitle ffprobe media prober
@end ignore
@bye

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project/jni/ffmpeg/doc/ffprobe.xsd Normal file
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<?xml version="1.0" encoding="UTF-8"?>
<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="convergence_duration" type="xsd:long" />
<xsd:attribute name="convergence_duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pts" type="xsd:long" />
<xsd:attribute name="pkt_pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="pkt_duration" type="xsd:long" />
<xsd:attribute name="pkt_duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<xsd:attribute name="channels" type="xsd:int" />
<xsd:attribute name="channel_layout" type="xsd:string"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="coded_picture_number" type="xsd:long" />
<xsd:attribute name="display_picture_number" type="xsd:long" />
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="reference" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string" use="required"/>
<xsd:attribute name="build_time" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_type" type="xsd:string" use="required"/>
<xsd:attribute name="compiler_version" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

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project/jni/ffmpeg/doc/ffserver.conf Normal file
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# 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 HTTP connections that can be handled. It has
# to be defined *before* the MaxClients parameter, since it defines the
# MaxClients maximum limit.
MaxHTTPConnections 2000
# Number of simultaneous requests that can be handled. Since FFServer
# is very fast, it is more likely that you will want to leave this high
# and use MaxBandwidth, below.
MaxClients 1000
# This the maximum amount of kbit/sec that you are prepared to
# consume when streaming to clients.
MaxBandwidth 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, where zero means unlimited. Default:
# File=/tmp/feed_name.ffm FileMaxSize=5M
File /tmp/feed1.ffm
FileMaxSize 200K
# You could specify
# ReadOnlyFile /saved/specialvideo.ffm
# This marks the file as readonly and it will not be deleted or updated.
# Specify launch in order to start ffmpeg automatically.
# First ffmpeg must be defined with an appropriate path if needed,
# after that options can follow, but avoid adding the http:// field
#Launch ffmpeg
# Only allow connections from localhost to the feed.
ACL allow 127.0.0.1
</Feed>
##################################################################
# Now you can define each stream which will be generated from the
# original audio and video stream. Each format has a filename (here
# 'test1.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 : MPEG-1 multiplexed video and audio
# mpegvideo : only MPEG-1 video
# mp2 : MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec)
# ogg : Ogg format (Vorbis audio codec)
# rm : RealNetworks-compatible stream. Multiplexed audio and video.
# ra : RealNetworks-compatible stream. Audio only.
# mpjpeg : Multipart JPEG (works with Netscape without any plugin)
# jpeg : Generate a single JPEG image.
# asf : ASF compatible streaming (Windows Media Player format).
# swf : Macromedia Flash compatible stream
# avi : AVI format (MPEG-4 video, MPEG audio sound)
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
# Ratecontrol buffer size
VideoBufferSize 40
# Number of frames per second
VideoFrameRate 3
# Size of the video frame: WxH (default: 160x128)
# The following abbreviations are defined: sqcif, qcif, cif, 4cif, qqvga,
# qvga, vga, svga, xga, uxga, qxga, sxga, qsxga, hsxga, wvga, wxga, wsxga,
# wuxga, woxga, wqsxga, wquxga, whsxga, whuxga, cga, ega, hd480, hd720,
# hd1080
VideoSize 160x128
# Transmit only intra frames (useful for low bitrates, but kills frame rate).
#VideoIntraOnly
# If non-intra only, an intra frame is transmitted every VideoGopSize
# frames. Video synchronization can only begin at an intra frame.
VideoGopSize 12
# More MPEG-4 parameters
# VideoHighQuality
# Video4MotionVector
# Choose your codecs:
#AudioCodec mp2
#VideoCodec mpeg1video
# Suppress audio
#NoAudio
# Suppress video
#NoVideo
#VideoQMin 3
#VideoQMax 31
# Set this to the number of seconds backwards in time to start. Note that
# most players will buffer 5-10 seconds of video, and also you need to allow
# for a keyframe to appear in the data stream.
#Preroll 15
# ACL:
# You can allow ranges of addresses (or single addresses)
#ACL ALLOW <first address> <last address>
# You can deny ranges of addresses (or single addresses)
#ACL DENY <first address> <last address>
# You can repeat the ACL allow/deny as often as you like. It is on a per
# stream basis. The first match defines the action. If there are no matches,
# then the default is the inverse of the last ACL statement.
#
# Thus 'ACL allow localhost' only allows access from localhost.
# 'ACL deny 1.0.0.0 1.255.255.255' would deny the whole of network 1 and
# allow everybody else.
</Stream>
##################################################################
# Example streams
# Multipart JPEG
#<Stream test.mjpg>
#Feed feed1.ffm
#Format mpjpeg
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#Strict -1
#</Stream>
# Single JPEG
#<Stream test.jpg>
#Feed feed1.ffm
#Format jpeg
#VideoFrameRate 2
#VideoIntraOnly
##VideoSize 352x240
#NoAudio
#Strict -1
#</Stream>
# Flash
#<Stream test.swf>
#Feed feed1.ffm
#Format swf
#VideoFrameRate 2
#VideoIntraOnly
#NoAudio
#</Stream>
# ASF compatible
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
# MP3 audio
#<Stream test.mp3>
#Feed feed1.ffm
#Format mp2
#AudioCodec mp3
#AudioBitRate 64
#AudioChannels 1
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Ogg Vorbis audio
#<Stream test.ogg>
#Feed feed1.ffm
#Title "Stream title"
#AudioBitRate 64
#AudioChannels 2
#AudioSampleRate 44100
#NoVideo
#</Stream>
# Real with audio only at 32 kbits
#<Stream test.ra>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#NoVideo
#NoAudio
#</Stream>
# Real with audio and video at 64 kbits
#<Stream test.rm>
#Feed feed1.ffm
#Format rm
#AudioBitRate 32
#VideoBitRate 128
#VideoFrameRate 25
#VideoGopSize 25
#NoAudio
#</Stream>
##################################################################
# A stream coming from a file: you only need to set the input
# filename and optionally a new format. Supported conversions:
# AVI -> ASF
#<Stream file.rm>
#File "/usr/local/httpd/htdocs/tlive.rm"
#NoAudio
#</Stream>
#<Stream file.asf>
#File "/usr/local/httpd/htdocs/test.asf"
#NoAudio
#Author "Me"
#Copyright "Super MegaCorp"
#Title "Test stream from disk"
#Comment "Test comment"
#</Stream>
##################################################################
# RTSP examples
#
# You can access this stream with the RTSP URL:
# rtsp://localhost:5454/test1-rtsp.mpg
#
# A non-standard RTSP redirector is also created. Its URL is:
# http://localhost:8090/test1-rtsp.rtsp
#<Stream test1-rtsp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#</Stream>
# Transcode an incoming live feed to another live feed,
# using libx264 and video presets
#<Stream live.h264>
#Format rtp
#Feed feed1.ffm
#VideoCodec libx264
#VideoFrameRate 24
#VideoBitRate 100
#VideoSize 480x272
#AVPresetVideo default
#AVPresetVideo baseline
#AVOptionVideo flags +global_header
#
#AudioCodec libfaac
#AudioBitRate 32
#AudioChannels 2
#AudioSampleRate 22050
#AVOptionAudio flags +global_header
#</Stream>
##################################################################
# SDP/multicast examples
#
# If you want to send your stream in multicast, you must set the
# multicast address with MulticastAddress. The port and the TTL can
# also be set.
#
# An SDP file is automatically generated by ffserver by adding the
# 'sdp' extension to the stream name (here
# http://localhost:8090/test1-sdp.sdp). You should usually give this
# file to your player to play the stream.
#
# The 'NoLoop' option can be used to avoid looping when the stream is
# terminated.
#<Stream test1-sdp.mpg>
#Format rtp
#File "/usr/local/httpd/htdocs/test1.mpg"
#MulticastAddress 224.124.0.1
#MulticastPort 5000
#MulticastTTL 16
#NoLoop
#</Stream>
##################################################################
# Special streams
# Server status
<Stream stat.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
#FaviconURL http://pond1.gladstonefamily.net:8080/favicon.ico
</Stream>
# Redirect index.html to the appropriate site
<Redirect index.html>
URL http://www.ffmpeg.org/
</Redirect>

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\input texinfo @c -*- texinfo -*-
@settitle ffserver Documentation
@titlepage
@center @titlefont{ffserver Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffserver [@var{options}]
@chapter Description
@c man begin DESCRIPTION
ffserver is a streaming server for both audio and video. It supports
several live feeds, streaming from files and time shifting on live feeds
(you can seek to positions in the past on each live feed, provided you
specify a big enough feed storage in ffserver.conf).
This documentation covers only the streaming aspects of ffserver /
ffmpeg. All questions about parameters for ffmpeg, codec questions,
etc. are not covered here. Read @file{ffmpeg.html} for more
information.
@section How does it work?
ffserver receives prerecorded files or FFM streams from some ffmpeg
instance as input, then streams them over RTP/RTSP/HTTP.
An ffserver instance will listen on some port as specified in the
configuration file. You can launch one or more instances of ffmpeg and
send one or more FFM streams to the port where ffserver is expecting
to receive them. Alternately, you can make ffserver launch such ffmpeg
instances at startup.
Input streams are called feeds, and each one is specified by a <Feed>
section in the configuration file.
For each feed you can have different output streams in various
formats, each one specified by a <Stream> section in the configuration
file.
@section Status stream
ffserver supports an HTTP interface which exposes the current status
of the server.
Simply point your browser to the address of the special status stream
specified in the configuration file.
For example if you have:
@example
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
@end example
then the server will post a page with the status information when
the special stream @file{status.html} is requested.
@section What can this do?
When properly configured and running, you can capture video and audio in real
time from a suitable capture card, and stream it out over the Internet to
either Windows Media Player or RealAudio player (with some restrictions).
It can also stream from files, though that is currently broken. Very often, a
web server can be used to serve up the files just as well.
It can stream prerecorded video from .ffm files, though it is somewhat tricky
to make it work correctly.
@section What do I need?
I use Linux on a 900 MHz Duron with a cheap Bt848 based TV capture card. I'm
using stock Linux 2.4.17 with the stock drivers. [Actually that isn't true,
I needed some special drivers for my motherboard-based sound card.]
I understand that FreeBSD systems work just fine as well.
@section How do I make it work?
First, build the kit. It *really* helps to have installed LAME first. Then when
you run the ffserver ./configure, make sure that you have the
@code{--enable-libmp3lame} flag turned on.
LAME is important as it allows for streaming audio to Windows Media Player.
Don't ask why the other audio types do not work.
As a simple test, just run the following two command lines where INPUTFILE
is some file which you can decode with ffmpeg:
@example
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
@end example
At this point you should be able to go to your Windows machine and fire up
Windows Media Player (WMP). Go to Open URL and enter
@example
http://<linuxbox>:8090/test.asf
@end example
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
transfer the entire file before starting to play.
The same is true of AVI files.
@section What happens next?
You should edit the ffserver.conf file to suit your needs (in terms of
frame rates etc). Then install ffserver and ffmpeg, write a script to start
them up, and off you go.
@section Troubleshooting
@subsection I don't hear any audio, but video is fine.
Maybe you didn't install LAME, or got your ./configure statement wrong. Check
the ffmpeg output to see if a line referring to MP3 is present. If not, then
your configuration was incorrect. If it is, then maybe your wiring is not
set up correctly. Maybe the sound card is not getting data from the right
input source. Maybe you have a really awful audio interface (like I do)
that only captures in stereo and also requires that one channel be flipped.
If you are one of these people, then export 'AUDIO_FLIP_LEFT=1' before
starting ffmpeg.
@subsection The audio and video lose sync after a while.
Yes, they do.
@subsection After a long while, the video update rate goes way down in WMP.
Yes, it does. Who knows why?
@subsection WMP 6.4 behaves differently to WMP 7.
Yes, it does. Any thoughts on this would be gratefully received. These
differences extend to embedding WMP into a web page. [There are two
object IDs that you can use: The old one, which does not play well, and
the new one, which does (both tested on the same system). However,
I suspect that the new one is not available unless you have installed WMP 7].
@section What else can it do?
You can replay video from .ffm files that was recorded earlier.
However, there are a number of caveats, including the fact that the
ffserver parameters must match the original parameters used to record the
file. If they do not, then ffserver deletes the file before recording into it.
(Now that I write this, it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and
there are a bunch more parameters that you cannot control. Post a message
to the mailing list if there are some 'must have' parameters. Look in
ffserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used
in browsers. These files are actually redirections to the underlying ASF
or RM file. The reason for this is that the browser often fetches the
entire file before starting up the external viewer. The redirection files
are very small and can be transferred quickly. [The stream itself is
often 'infinite' and thus the browser tries to download it and never
finishes.]
@section Tips
* When you connect to a live stream, most players (WMP, RA, etc) want to
buffer a certain number of seconds of material so that they can display the
signal continuously. However, ffserver (by default) starts sending data
in realtime. This means that there is a pause of a few seconds while the
buffering is being done by the player. The good news is that this can be
cured by adding a '?buffer=5' to the end of the URL. This means that the
stream should start 5 seconds in the past -- and so the first 5 seconds
of the stream are sent as fast as the network will allow. It will then
slow down to real time. This noticeably improves the startup experience.
You can also add a 'Preroll 15' statement into the ffserver.conf that will
add the 15 second prebuffering on all requests that do not otherwise
specify a time. In addition, ffserver will skip frames until a key_frame
is found. This further reduces the startup delay by not transferring data
that will be discarded.
* You may want to adjust the MaxBandwidth in the ffserver.conf to limit
the amount of bandwidth consumed by live streams.
@section Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully
grabbed is marginally less than the number that ought to be grabbed. This
means that the timestamp in the encoded data stream gets behind realtime.
This means that if you say 'Preroll 10', then when the stream gets 10
or more seconds behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are
handled.
@section Does the @code{?date=} stuff work.
Yes (subject to the limitation outlined above). Also note that whenever you
start ffserver, it deletes the ffm file (if any parameters have changed),
thus wiping out what you had recorded before.
The format of the @code{?date=xxxxxx} is fairly flexible. You should use one
of the following formats (the 'T' is literal):
@example
* YYYY-MM-DDTHH:MM:SS (localtime)
* YYYY-MM-DDTHH:MM:SSZ (UTC)
@end example
You can omit the YYYY-MM-DD, and then it refers to the current day. However
note that @samp{?date=16:00:00} refers to 16:00 on the current day -- this
may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream.
For example: @samp{http://localhost:8080/test.asf?date=2002-07-26T23:05:00}.
@c man end
@section What is FFM, FFM2
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of
video and audio streams and encoding options, and can store a moving time segment
of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files
generated by one version of ffmpeg/ffserver and another version of
ffmpeg/ffserver. It may work but its not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between
differing versions of tools. FFM2 is the default.
@chapter Options
@c man begin OPTIONS
@include avtools-common-opts.texi
@section Main options
@table @option
@item -f @var{configfile}
Use @file{configfile} instead of @file{/etc/ffserver.conf}.
@item -n
Enable no-launch mode. This option disables all the Launch directives
within the various <Stream> sections. Since ffserver will not launch
any ffmpeg instances, you will have to launch them manually.
@item -d
Enable debug mode. This option increases log verbosity, directs log
messages to stdout.
@end table
@c man end
@chapter See Also
@ifhtml
The @file{doc/ffserver.conf} example,
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
The @file{doc/ffserver.conf} example, ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffserver
@settitle ffserver video server
@end ignore
@bye

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Filter design
=============
This document explains guidelines that should be observed (or ignored with
good reason) when writing filters for libavfilter.
In this document, the word “frame” indicates either a video frame or a group
of audio samples, as stored in an AVFilterBuffer structure.
Format negotiation
==================
The query_formats method should set, for each input and each output links,
the list of supported formats.
For video links, that means pixel format. For audio links, that means
channel layout, and sample format (the sample packing is implied by the
sample format).
The lists are not just lists, they are references to shared objects. When
the negotiation mechanism computes the intersection of the formats
supported at each ends of a link, all references to both lists are
replaced with a reference to the intersection. And when a single format is
eventually chosen for a link amongst the remaining list, again, all
references to the list are updated.
That means that if a filter requires that its input and output have the
same format amongst a supported list, all it has to do is use a reference
to the same list of formats.
Buffer references ownership and permissions
===========================================
Principle
---------
Audio and video data are voluminous; the buffer and buffer reference
mechanism is intended to avoid, as much as possible, expensive copies of
that data while still allowing the filters to produce correct results.
The data is stored in buffers represented by AVFilterBuffer structures.
They must not be accessed directly, but through references stored in
AVFilterBufferRef structures. Several references can point to the
same buffer; the buffer is automatically deallocated once all
corresponding references have been destroyed.
The characteristics of the data (resolution, sample rate, etc.) are
stored in the reference; different references for the same buffer can
show different characteristics. In particular, a video reference can
point to only a part of a video buffer.
A reference is usually obtained as input to the start_frame or
filter_frame method or requested using the ff_get_video_buffer or
ff_get_audio_buffer functions. A new reference on an existing buffer can
be created with the avfilter_ref_buffer. A reference is destroyed using
the avfilter_unref_bufferp function.
Reference ownership
-------------------
At any time, a reference “belongs” to a particular piece of code,
usually a filter. With a few caveats that will be explained below, only
that piece of code is allowed to access it. It is also responsible for
destroying it, although this is sometimes done automatically (see the
section on link reference fields).
Here are the (fairly obvious) rules for reference ownership:
* A reference received by the start_frame or filter_frame method
belong to the corresponding filter.
Special exception: for video references: the reference may be used
internally for automatic copying and must not be destroyed before
end_frame; it can be given away to ff_start_frame.
* A reference passed to ff_start_frame or ff_filter_frame is given
away and must no longer be used.
* A reference created with avfilter_ref_buffer belongs to the code that
created it.
* A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
belongs to the code that requested it.
* A reference given as return value by the get_video_buffer or
get_audio_buffer method is given away and must no longer be used.
Link reference fields
---------------------
The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
the rules to handle them:
* cur_buf is set before the start_frame and filter_frame methods to
the same reference given as argument to the methods and belongs to the
destination filter of the link. If it has not been cleared after
end_frame or filter_frame, libavfilter will automatically destroy
the reference; therefore, any filter that needs to keep the reference
for longer must set cur_buf to NULL.
* out_buf belongs to the source filter of the link and can be used to
store a reference to the buffer that has been sent to the destination.
If it is not NULL after end_frame or filter_frame, libavfilter will
automatically destroy the reference.
If a video input pad does not have a start_frame method, the default
method will request a buffer on the first output of the filter, store
the reference in out_buf and push a second reference to the output.
* src_buf, cur_buf_copy and partial_buf are used by libavfilter
internally and must not be accessed by filters.
Reference permissions
---------------------
The AVFilterBufferRef structure has a perms field that describes what
the code that owns the reference is allowed to do to the buffer data.
Different references for the same buffer can have different permissions.
For video filters, the permissions only apply to the parts of the buffer
that have already been covered by the draw_slice method.
The value is a binary OR of the following constants:
* AV_PERM_READ: the owner can read the buffer data; this is essentially
always true and is there for self-documentation.
* AV_PERM_WRITE: the owner can modify the buffer data.
* AV_PERM_PRESERVE: the owner can rely on the fact that the buffer data
will not be modified by previous filters.
* AV_PERM_REUSE: the owner can output the buffer several times, without
modifying the data in between.
* AV_PERM_REUSE2: the owner can output the buffer several times and
modify the data in between (useless without the WRITE permissions).
* AV_PERM_ALIGN: the owner can access the data using fast operations
that require data alignment.
The READ, WRITE and PRESERVE permissions are about sharing the same
buffer between several filters to avoid expensive copies without them
doing conflicting changes on the data.
The REUSE and REUSE2 permissions are about special memory for direct
rendering. For example a buffer directly allocated in video memory must
not modified once it is displayed on screen, or it will cause tearing;
it will therefore not have the REUSE2 permission.
The ALIGN permission is about extracting part of the buffer, for
copy-less padding or cropping for example.
References received on input pads are guaranteed to have all the
permissions stated in the min_perms field and none of the permissions
stated in the rej_perms.
References obtained by ff_get_video_buffer and ff_get_audio_buffer are
guaranteed to have at least all the permissions requested as argument.
References created by avfilter_ref_buffer have the same permissions as
the original reference minus the ones explicitly masked; the mask is
usually ~0 to keep the same permissions.
Filters should remove permissions on reference they give to output
whenever necessary. It can be automatically done by setting the
rej_perms field on the output pad.
Here are a few guidelines corresponding to common situations:
* Filters that modify and forward their frame (like drawtext) need the
WRITE permission.
* Filters that read their input to produce a new frame on output (like
scale) need the READ permission on input and and must request a buffer
with the WRITE permission.
* Filters that intend to keep a reference after the filtering process
is finished (after end_frame or filter_frame returns) must have the
PRESERVE permission on it and remove the WRITE permission if they
create a new reference to give it away.
* Filters that intend to modify a reference they have kept after the end
of the filtering process need the REUSE2 permission and must remove
the PRESERVE permission if they create a new reference to give it
away.
Frame scheduling
================
The purpose of these rules is to ensure that frames flow in the filter
graph without getting stuck and accumulating somewhere.
Simple filters that output one frame for each input frame should not have
to worry about it.
start_frame / filter_frame
----------------------------
These methods are called when a frame is pushed to the filter's input.
They can be called at any time except in a reentrant way.
If the input frame is enough to produce output, then the filter should
push the output frames on the output link immediately.
As an exception to the previous rule, if the input frame is enough to
produce several output frames, then the filter needs output only at
least one per link. The additional frames can be left buffered in the
filter; these buffered frames must be flushed immediately if a new input
produces new output.
(Example: framerate-doubling filter: start_frame must (1) flush the
second copy of the previous frame, if it is still there, (2) push the
first copy of the incoming frame, (3) keep the second copy for later.)
If the input frame is not enough to produce output, the filter must not
call request_frame to get more. It must just process the frame or queue
it. The task of requesting more frames is left to the filter's
request_frame method or the application.
If a filter has several inputs, the filter must be ready for frames
arriving randomly on any input. Therefore, any filter with several inputs
will most likely require some kind of queuing mechanism. It is perfectly
acceptable to have a limited queue and to drop frames when the inputs
are too unbalanced.
request_frame
-------------
This method is called when a frame is wanted on an output.
For an input, it should directly call start_frame or filter_frame on
the corresponding output.
For a filter, if there are queued frames already ready, one of these
frames should be pushed. If not, the filter should request a frame on
one of its inputs, repeatedly until at least one frame has been pushed.
Return values:
if request_frame could produce a frame, it should return 0;
if it could not for temporary reasons, it should return AVERROR(EAGAIN);
if it could not because there are no more frames, it should return
AVERROR_EOF.
The typical implementation of request_frame for a filter with several
inputs will look like that:
if (frames_queued) {
push_one_frame();
return 0;
}
while (!frame_pushed) {
input = input_where_a_frame_is_most_needed();
ret = avfilter_request_frame(input);
if (ret == AVERROR_EOF) {
process_eof_on_input();
} else if (ret < 0) {
return ret;
}
}
return 0;
Note that, except for filters that can have queued frames, request_frame
does not push frames: it requests them to its input, and as a reaction,
the start_frame / filter_frame method will be called and do the work.

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\input texinfo @c -*- texinfo -*-
@settitle Using git to develop FFmpeg
@titlepage
@center @titlefont{Using git to develop FFmpeg}
@end titlepage
@top
@contents
@chapter Introduction
This document aims in giving some quick references on a set of useful git
commands. You should always use the extensive and detailed documentation
provided directly by git:
@example
git --help
man git
@end example
shows you the available subcommands,
@example
git <command> --help
man git-<command>
@end example
shows information about the subcommand <command>.
Additional information could be found on the
@url{http://gitref.org, Git Reference} website
For more information about the Git project, visit the
@url{http://git-scm.com/, Git website}
Consult these resources whenever you have problems, they are quite exhaustive.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project
@chapter Basics Usage
@section Get GIT
You can get git from @url{http://git-scm.com/}
Most distribution and operating system provide a package for it.
@section Cloning the source tree
@example
git clone git://source.ffmpeg.org/ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>}.
@example
git clone git@@source.ffmpeg.org:ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
@example
git config --global core.autocrlf false
@end example
@section Updating the source tree to the latest revision
@example
git pull (--rebase)
@end example
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
@float IMPORTANT
@command{--rebase} (see below) is recommended.
@end float
@section Rebasing your local branches
@example
git pull --rebase
@end example
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
@section Adding/removing files/directories
@example
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
@end example
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
@section Showing modifications
@example
git diff <filename(s)>
@end example
will show all local modifications in your working directory as unified diff.
@section Inspecting the changelog
@example
git log <filename(s)>
@end example
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org/
@section Checking source tree status
@example
git status
@end example
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
@section Committing
@example
git diff --check
@end example
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
@example
git diff -w -b <filename(s)>
@end example
Also check the output of
@example
git status
@end example
to make sure you don't have untracked files or deletions.
@example
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
@end example
Use @var{--global} to set the global configuration for all your git checkouts.
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
@example
git commit
@end example
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
@example
git config --global core.editor
@end example
or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
@section Preparing a patchset
@example
git format-patch <commit> [-o directory]
@end example
will generate a set of patches for each commit between @var{<commit>} and
current @var{HEAD}. E.g.
@example
git format-patch origin/master
@end example
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
@example
git format-patch -n
@end example
which will generate patches from last @var{n} commits.
By default the patches are created in the current directory.
@section Sending patches for review
@example
git send-email <commit list|directory>
@end example
will send the patches created by @command{git format-patch} or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. @var{git-email}
package on Debian-based distros).
@section Renaming/moving/copying files or contents of files
Git automatically tracks such changes, making those normal commits.
@example
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
@end example
@chapter Git configuration
In order to simplify a few workflows, it is advisable to configure both
your personal Git installation and your local FFmpeg repository.
@section Personal Git installation
Add the following to your @file{~/.gitconfig} to help @command{git send-email}
and @command{git format-patch} detect renames:
@example
[diff]
renames = copy
@end example
@section Repository configuration
In order to have @command{git send-email} automatically send patches
to the ffmpeg-devel mailing list, add the following stanza
to @file{/path/to/ffmpeg/repository/.git/config}:
@example
[sendemail]
to = ffmpeg-devel@@ffmpeg.org
@end example
@chapter FFmpeg specific
@section Reverting broken commits
@example
git reset <commit>
@end example
@command{git reset} will uncommit the changes till @var{<commit>} rewriting
the current branch history.
@example
git commit --amend
@end example
allows to amend the last commit details quickly.
@example
git rebase -i origin/master
@end example
will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
@float NOTE
@command{git reset}, @command{git commit --amend} and @command{git rebase}
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
@end float
@example
git revert <commit>
@end example
@command{git revert} will generate a revert commit. This will not make the
faulty commit disappear from the history.
@section Pushing changes to remote trees
@example
git push
@end example
Will push the changes to the default remote (@var{origin}).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to and to sync the local tree.
@example
git remote add <name> <url>
@end example
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
@example
git push <remote> <refspec>
@end example
Will push the changes to the @var{<remote>} repository.
Omitting @var{<refspec>} makes @command{git push} update all the remote
branches matching the local ones.
@section Finding a specific svn revision
Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
based on a regular expression. see man gitrevisions
@example
git show :/'as revision 23456'
@end example
will show the svn changeset @var{r23456}. With older git versions searching in
the @command{git log} output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
@example
git checkout -b svn_23456 :/'as revision 23456'
@end example
or for git < 1.7.1 with
@example
git checkout -b svn_23456 $SHA1
@end example
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter pre-push checklist
Once you have a set of commits that you feel are ready for pushing,
work through the following checklist to doublecheck everything is in
proper order. This list tries to be exhaustive. In case you are just
pushing a typo in a comment, some of the steps may be unnecessary.
Apply your common sense, but if in doubt, err on the side of caution.
First, make sure that the commits and branches you are going to push
match what you want pushed and that nothing is missing, extraneous or
wrong. You can see what will be pushed by running the git push command
with --dry-run first. And then inspecting the commits listed with
@command{git log -p 1234567..987654}. The @command{git status} command
may help in finding local changes that have been forgotten to be added.
Next let the code pass through a full run of our testsuite.
@itemize
@item @command{make distclean}
@item @command{/path/to/ffmpeg/configure}
@item @command{make check}
@item if fate fails due to missing samples run @command{make fate-rsync} and retry
@end itemize
Make sure all your changes have been checked before pushing them, the
testsuite only checks against regressions and that only to some extend. It does
obviously not check newly added features/code to be working unless you have
added a test for that (which is recommended).
Also note that every single commit should pass the test suite, not just
the result of a series of patches.
Once everything passed, push the changes to your public ffmpeg clone and post a
merge request to ffmpeg-devel. You can also push them directly but this is not
recommended.
@chapter Server Issues
Contact the project admins @email{root@@ffmpeg.org} if you have technical
problems with the GIT server.

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About Git write access:
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Before everything else, you should know how to use GIT properly.
Luckily Git comes with excellent documentation.
git --help
man git
shows you the available subcommands,
git <command> --help
man git-<command>
shows information about the subcommand <command>.
The most comprehensive manual is the website Git Reference
http://gitref.org/
For more information about the Git project, visit
http://git-scm.com/
Consult these resources whenever you have problems, they are quite exhaustive.
You do not need a special username or password.
All you need is to provide a ssh public key to the Git server admin.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines. Read it at least once, if you are granted commit privileges to the
FFmpeg project you are expected to be familiar with these rules.
I. BASICS:
==========
0. Get GIT:
Most distributions have a git package, if not
You can get git from http://git-scm.com/
1. Cloning the source tree:
git clone git://source.ffmpeg.org/ffmpeg <target>
This will put the FFmpeg sources into the directory <target>.
git clone git@source.ffmpeg.org:ffmpeg <target>
This will put the FFmpeg sources into the directory <target> and let
you push back your changes to the remote repository.
2. Updating the source tree to the latest revision:
git pull (--ff-only)
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
Caveat: Since merge commits are forbidden at least for the initial
months of git --ff-only or --rebase (see below) are recommended.
--ff-only will fail and not create merge commits if your branch
has diverged (has a different history) from the tracked branch.
2.a Rebasing your local branches:
git pull --rebase
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
3. Adding/removing files/directories:
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
GIT needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
4. Showing modifications:
git diff <filename(s)>
will show all local modifications in your working directory as unified diff.
5. Inspecting the changelog:
git log <filename(s)>
You may also use the graphical tools like gitview or gitk or the web
interface available at http://source.ffmpeg.org
6. Checking source tree status:
git status
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
7. Committing:
git diff --check
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
git diff -w -b <filename(s)>
Also check the output of
git status
to make sure you don't have untracked files or deletions.
git add [-i|-p|-A] <filenames/dirnames>
Make sure you have told git your name and email address, e.g. by running
git config --global user.name "My Name"
git config --global user.email my@email.invalid
(--global to set the global configuration for all your git checkouts).
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
git commit
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
git config core.editor
or set by one of the following environment variables:
GIT_EDITOR, VISUAL or EDITOR.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by git format-patch.
8. Renaming/moving/copying files or contents of files:
Git automatically tracks such changes, making those normal commits.
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
Do not move, rename or copy files of which you are not the maintainer without
discussing it on the mailing list first!
9. Reverting broken commits
git revert <commit>
git revert will generate a revert commit. This will not make the faulty
commit disappear from the history.
git reset <commit>
git reset will uncommit the changes till <commit> rewriting the current
branch history.
git commit --amend
allows to amend the last commit details quickly.
git rebase -i origin/master
will replay local commits over the main repository allowing to edit,
merge or remove some of them in the process.
Note that the reset, commit --amend and rebase rewrite history, so you
should use them ONLY on your local or topic branches.
The main repository will reject those changes.
10. Preparing a patchset.
git format-patch <commit> [-o directory]
will generate a set of patches for each commit between <commit> and
current HEAD. E.g.
git format-patch origin/master
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
git format-patch -n
which will generate patches from last n commits.
By default the patches are created in the current directory.
11. Sending patches for review
git send-email <commit list|directory>
will send the patches created by git format-patch or directly generates
them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. git-email
package on Debian-based distros).
12. Pushing changes to remote trees
git push
Will push the changes to the default remote (origin).
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to 2 and 2.a to sync the local tree.
git remote add <name> <url>
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
git push <remote> <refspec>
Will push the changes to the remote repository. Omitting refspec makes git
push update all the remote branches matching the local ones.
13. Finding a specific svn revision
Since version 1.7.1 git supports ':/foo' syntax for specifying commits
based on a regular expression. see man gitrevisions
git show :/'as revision 23456'
will show the svn changeset r23456. With older git versions searching in
the git log output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
git checkout -b svn_23456 :/'as revision 23456'
or for git < 1.7.1 with
git checkout -b svn_23456 $SHA1
where $SHA1 is the commit SHA1 from the 'git log' output.
Contact the project admins <root at ffmpeg dot org> if you have technical
problems with the GIT server.

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@chapter Input Devices
@c man begin INPUT DEVICES
Input devices are configured elements in FFmpeg which allow to access
the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices
are enabled by default. You can list all available ones using the
configure option "--list-indevs".
You can disable all the input devices using the configure option
"--disable-indevs", and selectively enable an input device using the
option "--enable-indev=@var{INDEV}", or you can disable a particular
input device using the option "--disable-indev=@var{INDEV}".
The option "-formats" of the ff* tools will display the list of
supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound
installed on your system.
This device allows capturing from an ALSA device. The name of the
device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
@example
hw:@var{CARD}[,@var{DEV}[,@var{SUBDEV}]]
@end example
where the @var{DEV} and @var{SUBDEV} components are optional.
The three arguments (in order: @var{CARD},@var{DEV},@var{SUBDEV})
specify card number or identifier, device number and subdevice number
(-1 means any).
To see the list of cards currently recognized by your system check the
files @file{/proc/asound/cards} and @file{/proc/asound/devices}.
For example to capture with @command{ffmpeg} from an ALSA device with
card id 0, you may run the command:
@example
ffmpeg -f alsa -i hw:0 alsaout.wav
@end example
For more information see:
@url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
@section bktr
BSD video input device.
@section dshow
Windows DirectShow input device.
DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.
Currently only audio and video devices are supported.
Multiple devices may be opened as separate inputs, but they may also be
opened on the same input, which should improve synchronism between them.
The input name should be in the format:
@example
@var{TYPE}=@var{NAME}[:@var{TYPE}=@var{NAME}]
@end example
where @var{TYPE} can be either @var{audio} or @var{video},
and @var{NAME} is the device's name.
@subsection Options
If no options are specified, the device's defaults are used.
If the device does not support the requested options, it will
fail to open.
@table @option
@item video_size
Set the video size in the captured video.
@item framerate
Set the framerate in the captured video.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
@item sample_size
Set the sample size (in bits) of the captured audio.
@item channels
Set the number of channels in the captured audio.
@item list_devices
If set to @option{true}, print a list of devices and exit.
@item list_options
If set to @option{true}, print a list of selected device's options
and exit.
@item video_device_number
Set video device number for devices with same name (starts at 0,
defaults to 0).
@item audio_device_number
Set audio device number for devices with same name (starts at 0,
defaults to 0).
@item pixel_format
Select pixel format to be used by DirectShow. This may only be set when
the video codec is not set or set to rawvideo.
@item audio_buffer_size
Set audio device buffer size in milliseconds (which can directly
impact latency, depending on the device).
Defaults to using the audio device's
default buffer size (typically some multiple of 500ms).
Setting this value too low can degrade performance.
See also
@url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
@end table
@subsection Examples
@itemize
@item
Print the list of DirectShow supported devices and exit:
@example
$ ffmpeg -list_devices true -f dshow -i dummy
@end example
@item
Open video device @var{Camera}:
@example
$ ffmpeg -f dshow -i video="Camera"
@end example
@item
Open second video device with name @var{Camera}:
@example
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
@end example
@item
Open video device @var{Camera} and audio device @var{Microphone}:
@example
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
@end example
@item
Print the list of supported options in selected device and exit:
@example
$ ffmpeg -list_options true -f dshow -i video="Camera"
@end example
@end itemize
@section dv1394
Linux DV 1394 input device.
@section fbdev
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction
layer to show graphics on a computer monitor, typically on the
console. It is accessed through a file device node, usually
@file{/dev/fb0}.
For more detailed information read the file
Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device @file{/dev/fb0} with
@command{ffmpeg}:
@example
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
@end example
You can take a single screenshot image with the command:
@example
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
@end example
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section iec61883
FireWire DV/HDV input device using libiec61883.
To enable this input device, you need libiec61883, libraw1394 and
libavc1394 installed on your system. Use the configure option
@code{--enable-libiec61883} to compile with the device enabled.
The iec61883 capture device supports capturing from a video device
connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
FireWire stack (juju). This is the default DV/HDV input method in Linux
Kernel 2.6.37 and later, since the old FireWire stack was removed.
Specify the FireWire port to be used as input file, or "auto"
to choose the first port connected.
@subsection Options
@table @option
@item dvtype
Override autodetection of DV/HDV. This should only be used if auto
detection does not work, or if usage of a different device type
should be prohibited. Treating a DV device as HDV (or vice versa) will
not work and result in undefined behavior.
The values @option{auto}, @option{dv} and @option{hdv} are supported.
@item dvbuffer
Set maxiumum size of buffer for incoming data, in frames. For DV, this
is an exact value. For HDV, it is not frame exact, since HDV does
not have a fixed frame size.
@item dvguid
Select the capture device by specifying it's GUID. Capturing will only
be performed from the specified device and fails if no device with the
given GUID is found. This is useful to select the input if multiple
devices are connected at the same time.
Look at /sys/bus/firewire/devices to find out the GUIDs.
@end table
@subsection Examples
@itemize
@item
Grab and show the input of a FireWire DV/HDV device.
@example
ffplay -f iec61883 -i auto
@end example
@item
Grab and record the input of a FireWire DV/HDV device,
using a packet buffer of 100000 packets if the source is HDV.
@example
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
@end example
@end itemize
@section jack
JACK input device.
To enable this input device during configuration you need libjack
installed on your system.
A JACK input device creates one or more JACK writable clients, one for
each audio channel, with name @var{client_name}:input_@var{N}, where
@var{client_name} is the name provided by the application, and @var{N}
is a number which identifies the channel.
Each writable client will send the acquired data to the FFmpeg input
device.
Once you have created one or more JACK readable clients, you need to
connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the @command{jack_connect}
and @command{jack_disconnect} programs, or do it through a graphical interface,
for example with @command{qjackctl}.
To list the JACK clients and their properties you can invoke the command
@command{jack_lsp}.
Follows an example which shows how to capture a JACK readable client
with @command{ffmpeg}.
@example
# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav
# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000
# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm
# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1
@end example
For more information read:
@url{http://jackaudio.org/}
@section lavfi
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter
filtergraph.
For each filtergraph open output, the input device will create a
corresponding stream which is mapped to the generated output. Currently
only video data is supported. The filtergraph is specified through the
option @option{graph}.
@subsection Options
@table @option
@item graph
Specify the filtergraph to use as input. Each video open output must be
labelled by a unique string of the form "out@var{N}", where @var{N} is a
number starting from 0 corresponding to the mapped input stream
generated by the device.
The first unlabelled output is automatically assigned to the "out0"
label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input
device.
@item graph_file
Set the filename of the filtergraph to be read and sent to the other
filters. Syntax of the filtergraph is the same as the one specified by
the option @var{graph}.
@end table
@subsection Examples
@itemize
@item
Create a color video stream and play it back with @command{ffplay}:
@example
ffplay -f lavfi -graph "color=c=pink [out0]" dummy
@end example
@item
As the previous example, but use filename for specifying the graph
description, and omit the "out0" label:
@example
ffplay -f lavfi color=c=pink
@end example
@item
Create three different video test filtered sources and play them:
@example
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
@end example
@item
Read an audio stream from a file using the amovie source and play it
back with @command{ffplay}:
@example
ffplay -f lavfi "amovie=test.wav"
@end example
@item
Read an audio stream and a video stream and play it back with
@command{ffplay}:
@example
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
@end example
@end itemize
@section libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
@section openal
The OpenAL input device provides audio capture on all systems with a
working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with @code{--enable-openal}.
OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on your
installation you may need to specify additional flags via the
@code{--extra-cflags} and @code{--extra-ldflags} for allowing the build
system to locate the OpenAL headers and libraries.
An incomplete list of OpenAL implementations follows:
@table @strong
@item Creative
The official Windows implementation, providing hardware acceleration
with supported devices and software fallback.
See @url{http://openal.org/}.
@item OpenAL Soft
Portable, open source (LGPL) software implementation. Includes
backends for the most common sound APIs on the Windows, Linux,
Solaris, and BSD operating systems.
See @url{http://kcat.strangesoft.net/openal.html}.
@item Apple
OpenAL is part of Core Audio, the official Mac OS X Audio interface.
See @url{http://developer.apple.com/technologies/mac/audio-and-video.html}
@end table
This device allows to capture from an audio input device handled
through OpenAL.
You need to specify the name of the device to capture in the provided
filename. If the empty string is provided, the device will
automatically select the default device. You can get the list of the
supported devices by using the option @var{list_devices}.
@subsection Options
@table @option
@item channels
Set the number of channels in the captured audio. Only the values
@option{1} (monaural) and @option{2} (stereo) are currently supported.
Defaults to @option{2}.
@item sample_size
Set the sample size (in bits) of the captured audio. Only the values
@option{8} and @option{16} are currently supported. Defaults to
@option{16}.
@item sample_rate
Set the sample rate (in Hz) of the captured audio.
Defaults to @option{44.1k}.
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}.
@end table
@subsection Examples
Print the list of OpenAL supported devices and exit:
@example
$ ffmpeg -list_devices true -f openal -i dummy out.ogg
@end example
Capture from the OpenAL device @file{DR-BT101 via PulseAudio}:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
@end example
Capture from the default device (note the empty string '' as filename):
@example
$ ffmpeg -f openal -i '' out.ogg
@end example
Capture from two devices simultaneously, writing to two different files,
within the same @command{ffmpeg} command:
@example
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
@end example
Note: not all OpenAL implementations support multiple simultaneous capture -
try the latest OpenAL Soft if the above does not work.
@section oss
Open Sound System input device.
The filename to provide to the input device is the device node
representing the OSS input device, and is usually set to
@file{/dev/dsp}.
For example to grab from @file{/dev/dsp} using @command{ffmpeg} use the
command:
@example
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
@end example
For more information about OSS see:
@url{http://manuals.opensound.com/usersguide/dsp.html}
@section pulse
pulseaudio input device.
To enable this input device during configuration you need libpulse-simple
installed in your system.
The filename to provide to the input device is a source device or the
string "default"
To list the pulse source devices and their properties you can invoke
the command @command{pactl list sources}.
@example
ffmpeg -f pulse -i default /tmp/pulse.wav
@end example
@subsection @var{server} AVOption
The syntax is:
@example
-server @var{server name}
@end example
Connects to a specific server.
@subsection @var{name} AVOption
The syntax is:
@example
-name @var{application name}
@end example
Specify the application name pulse will use when showing active clients,
by default it is the LIBAVFORMAT_IDENT string
@subsection @var{stream_name} AVOption
The syntax is:
@example
-stream_name @var{stream name}
@end example
Specify the stream name pulse will use when showing active streams,
by default it is "record"
@subsection @var{sample_rate} AVOption
The syntax is:
@example
-sample_rate @var{samplerate}
@end example
Specify the samplerate in Hz, by default 48kHz is used.
@subsection @var{channels} AVOption
The syntax is:
@example
-channels @var{N}
@end example
Specify the channels in use, by default 2 (stereo) is set.
@subsection @var{frame_size} AVOption
The syntax is:
@example
-frame_size @var{bytes}
@end example
Specify the number of byte per frame, by default it is set to 1024.
@subsection @var{fragment_size} AVOption
The syntax is:
@example
-fragment_size @var{bytes}
@end example
Specify the minimal buffering fragment in pulseaudio, it will affect the
audio latency. By default it is unset.
@section sndio
sndio input device.
To enable this input device during configuration you need libsndio
installed on your system.
The filename to provide to the input device is the device node
representing the sndio input device, and is usually set to
@file{/dev/audio0}.
For example to grab from @file{/dev/audio0} using @command{ffmpeg} use the
command:
@example
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
@end example
@section video4linux2
Video4Linux2 input video device.
The name of the device to grab is a file device node, usually Linux
systems tend to automatically create such nodes when the device
(e.g. an USB webcam) is plugged into the system, and has a name of the
kind @file{/dev/video@var{N}}, where @var{N} is a number associated to
the device.
Video4Linux2 devices usually support a limited set of
@var{width}x@var{height} sizes and framerates. You can check which are
supported using @command{-list_formats all} for Video4Linux2 devices.
Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
The time base for the timestamps is 1 microsecond. Depending on the kernel
version and configuration, the timestamps may be derived from the real time
clock (origin at the Unix Epoch) or the monotonic clock (origin usually at
boot time, unaffected by NTP or manual changes to the clock). The
@option{-timestamps abs} or @option{-ts abs} option can be used to force
conversion into the real time clock.
Note that if FFmpeg is build with v4l-utils support ("--enable-libv4l2"
option), it will always be used.
@example
# Grab and show the input of a video4linux2 device.
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
# Grab and record the input of a video4linux2 device, leave the
framerate and size as previously set.
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
@end example
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and
"video4linux2".
@section vfwcap
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from
0 to 9. You may use "list" as filename to print a list of drivers. Any
other filename will be interpreted as device number 0.
@section x11grab
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
@example
[@var{hostname}]:@var{display_number}.@var{screen_number}[+@var{x_offset},@var{y_offset}]
@end example
@var{hostname}:@var{display_number}.@var{screen_number} specifies the
X11 display name of the screen to grab from. @var{hostname} can be
omitted, and defaults to "localhost". The environment variable
@env{DISPLAY} contains the default display name.
@var{x_offset} and @var{y_offset} specify the offsets of the grabbed
area with respect to the top-left border of the X11 screen. They
default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the @command{dpyinfo} program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from @file{:0.0} using @command{ffmpeg}:
@example
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg
@end example
Grab at position @code{10,20}:
@example
ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg
@end example
@subsection Options
@table @option
@item draw_mouse
Specify whether to draw the mouse pointer. A value of @code{0} specify
not to draw the pointer. Default value is @code{1}.
@item follow_mouse
Make the grabbed area follow the mouse. The argument can be
@code{centered} or a number of pixels @var{PIXELS}.
When it is specified with "centered", the grabbing region follows the mouse
pointer and keeps the pointer at the center of region; otherwise, the region
follows only when the mouse pointer reaches within @var{PIXELS} (greater than
zero) to the edge of region.
For example:
@example
ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
@end example
To follow only when the mouse pointer reaches within 100 pixels to edge:
@example
ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg
@end example
@item framerate
Set the grabbing frame rate. Default value is @code{ntsc},
corresponding to a framerate of @code{30000/1001}.
@item show_region
Show grabbed region on screen.
If @var{show_region} is specified with @code{1}, then the grabbing
region will be indicated on screen. With this option, it is easy to
know what is being grabbed if only a portion of the screen is grabbed.
For example:
@example
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
@end example
With @var{follow_mouse}:
@example
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg
@end example
@item video_size
Set the video frame size. Default value is @code{vga}.
@end table
@c man end INPUT DEVICES

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FFmpeg's bug/patch/feature request tracker manual
=================================================
NOTE: This is a draft.
Overview:
---------
FFmpeg uses Trac for tracking issues, new issues and changes to
existing issues can be done through a web interface.
Issues can be different kinds of things we want to keep track of
but that do not belong into the source tree itself. This includes
bug reports, patches, feature requests and license violations. We
might add more items to this list in the future, so feel free to
propose a new `type of issue' on the ffmpeg-devel mailing list if
you feel it is worth tracking.
It is possible to subscribe to individual issues by adding yourself to the
Cc list or to subscribe to the ffmpeg-trac mailing list which receives
a mail for every change to every issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
The URL of the webinterface of the tracker is:
http(s)://ffmpeg.org/trac/ffmpeg
Type:
-----
bug / defect
An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
prevents it from behaving as intended.
feature request / enhancement
Request of support for encoding or decoding of a new codec, container
or variant.
Request of support for more, less or plain different output or behavior
where the current implementation cannot be considered wrong.
license violation
ticket to keep track of (L)GPL violations of ffmpeg by others
patch
A patch as generated by diff which conforms to the patch submission and
development policy.
Priority:
---------
critical
Bugs and patches which deal with data loss and security issues.
No feature request can be critical.
important
Bugs which make FFmpeg unusable for a significant number of users, and
patches fixing them.
Examples here might be completely broken MPEG-4 decoding or a build issue
on Linux.
While broken 4xm decoding or a broken OS/2 build would not be important,
the separation to normal is somewhat fuzzy.
For feature requests this priority would be used for things many people
want.
Regressions also should be marked as important, regressions are bugs that
don't exist in a past revision or another branch.
normal
minor
Bugs and patches about things like spelling errors, "mp2" instead of
"mp3" being shown and such.
Feature requests about things few people want or which do not make a big
difference.
wish
Something that is desirable to have but that there is no urgency at
all to implement, e.g. something completely cosmetic like a website
restyle or a personalized doxy template or the FFmpeg logo.
This priority is not valid for bugs.
Status:
-------
new
initial state
open
intermediate states
closed
final state
Analyzed flag:
--------------
Bugs which have been analyzed and where it is understood what causes them
and which exact chain of events triggers them. This analysis should be
available as a message in the bug report.
Note, do not change the status to analyzed without also providing a clear
and understandable analysis.
This state implicates that the bug either has been reproduced or that
reproduction is not needed as the bug is already understood.
Type/Status/Substatus:
----------
*/new/new
Initial state of new bugs, patches and feature requests submitted by
users.
*/open/open
Issues which have been briefly looked at and which did not look outright
invalid.
This implicates that no real more detailed state applies yet. Conversely,
the more detailed states below implicate that the issue has been briefly
looked at.
*/closed/duplicate
Bugs, patches or feature requests which are duplicates.
Note that patches dealing with the same thing in a different way are not
duplicates.
Note, if you mark something as duplicate, do not forget setting the
superseder so bug reports are properly linked.
*/closed/invalid
Bugs caused by user errors, random ineligible or otherwise nonsense stuff.
*/closed/needs_more_info
Issues for which some information has been requested by the developers,
but which has not been provided by anyone within reasonable time.
bug/closed/fixed
Bugs which have to the best of our knowledge been fixed.
bug/closed/wont_fix
Bugs which we will not fix. Possible reasons include legality, high
complexity for the sake of supporting obscure corner cases, speed loss
for similarly esoteric purposes, et cetera.
This also means that we would reject a patch.
If we are just too lazy to fix a bug then the correct state is open
and unassigned. Closed means that the case is closed which is not
the case if we are just waiting for a patch.
bug/closed/works_for_me
Bugs for which sufficient information was provided to reproduce but
reproduction failed - that is the code seems to work correctly to the
best of our knowledge.
patch/open/approved
Patches which have been reviewed and approved by a developer.
Such patches can be applied anytime by any other developer after some
reasonable testing (compile + regression tests + does the patch do
what the author claimed).
patch/open/needs_changes
Patches which have been reviewed and need changes to be accepted.
patch/closed/applied
Patches which have been applied.
patch/closed/rejected
Patches which have been rejected.
feature_request/closed/implemented
Feature requests which have been implemented.
feature_request/closed/wont_implement
Feature requests which will not be implemented. The reasons here could
be legal, philosophical or others.
Note, please do not use type-status-substatus combinations other than the
above without asking on ffmpeg-dev first!
Note2, if you provide the requested info do not forget to remove the
needs_more_info substatus.
Component:
----------
avcodec
issues in libavcodec/*
avformat
issues in libavformat/*
avutil
issues in libavutil/*
regression test
issues in tests/*
ffmpeg
issues in or related to ffmpeg.c
ffplay
issues in or related to ffplay.c
ffprobe
issues in or related to ffprobe.c
ffserver
issues in or related to ffserver.c
build system
issues in or related to configure/Makefile
regression
bugs which were not present in a past revision
trac
issues related to our issue tracker

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\input texinfo @c -*- texinfo -*-
@settitle Libavcodec Documentation
@titlepage
@center @titlefont{Libavcodec Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavcodec library provides a generic encoding/decoding framework
and contains multiple decoders and encoders for audio, video and
subtitle streams, and several bitstream filters.
The shared architecture provides various services ranging from bit
stream I/O to DSP optimizations, and makes it suitable for
implementing robust and fast codecs as well as for experimentation.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavcodec
@settitle media streams decoding and encoding library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Libavdevice Documentation
@titlepage
@center @titlefont{Libavdevice Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavdevice library provides a generic framework for grabbing from
and rendering to many common multimedia input/output devices, and
supports several input and output devices, including Video4Linux2,
VfW, DShow, and ALSA.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-devices(1),
libavutil(3), libavcodec(3), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavdevice
@settitle multimedia device handling library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Libavfilter Documentation
@titlepage
@center @titlefont{Libavfilter Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavfilter library provides a generic audio/video filtering
framework containing several filters, sources and sinks.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-filters.html,ffmpeg-filters},
@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-filters(1),
libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavfilter
@settitle multimedia filtering library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Libavformat Documentation
@titlepage
@center @titlefont{Libavformat Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavformat library provides a generic framework for multiplexing
and demultiplexing (muxing and demuxing) audio, video and subtitle
streams. It encompasses multiple muxers and demuxers for multimedia
container formats.
It also supports several input and output protocols to access a media
resource.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-formats(1), ffmpeg-protocols(1),
libavutil(3), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavformat
@settitle multimedia muxing and demuxing library
@end ignore
@bye

44
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\input texinfo @c -*- texinfo -*-
@settitle Libavutil Documentation
@titlepage
@center @titlefont{Libavutil Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats).
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-utils.html,ffmpeg-utils}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-utils(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavutil
@settitle multimedia-biased utility library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Libswresample Documentation
@titlepage
@center @titlefont{Libswresample Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libswresample library performs highly optimized audio resampling,
rematrixing and sample format conversion operations.
Specifically, this library performs the following conversions:
@itemize
@item
@emph{Resampling}: is the process of changing the audio rate, for
example from an high sample rate of 44100Hz to 8000Hz. Audio
conversion from high to low sample rate is a lossy process. Several
resampling options and algorithms are available.
@item
@emph{Format conversion}: is the process of converting the type of
samples, for example from 16-bit signed samples to unsigned 8-bit or
float samples. It also handles packing conversion, when passing from
packed layout (all samples belonging to distinct channels interleaved
in the same buffer), to planar layout (all samples belonging to the
same channel stored in a dedicated buffer or "plane").
@item
@emph{Rematrixing}: is the process of changing the channel layout, for
example from stereo to mono. When the input channels cannot be mapped
to the output streams, the process is lossy, since it involves
different gain factors and mixing.
@end itemize
Various other audio conversions (e.g. stretching and padding) are
enabled through dedicated options.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-resampler(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libswresample
@settitle audio resampling library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@settitle Libswscale Documentation
@titlepage
@center @titlefont{Libswscale Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libswscale library performs highly optimized image scaling and
colorspace and pixel format conversion operations.
Specifically, this library performs the following conversions:
@itemize
@item
@emph{Rescaling}: is the process of changing the video size. Several
rescaling options and algorithms are available. This is usually a
lossy process.
@item
@emph{Pixel format conversion}: is the process of converting the image
format and colorspace of the image, for example from planar YUV420P to
RGB24 packed. It also handles packing conversion, that is converts
from packed layout (all pixels belonging to distinct planes
interleaved in the same buffer), to planar layout (all samples
belonging to the same plane stored in a dedicated buffer or "plane").
This is usually a lossy process in case the source and destination
colorspaces differ.
@end itemize
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe}, @url{ffserver.html,ffserver},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1),
ffmpeg-scaler(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libswscale
@settitle video scaling and pixel format conversion library
@end ignore
@bye

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@chapter Metadata
@c man begin METADATA
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded
INI-like text file and then load it back using the metadata muxer/demuxer.
The file format is as follows:
@enumerate
@item
A file consists of a header and a number of metadata tags divided into sections,
each on its own line.
@item
The header is a ';FFMETADATA' string, followed by a version number (now 1).
@item
Metadata tags are of the form 'key=value'
@item
Immediately after header follows global metadata
@item
After global metadata there may be sections with per-stream/per-chapter
metadata.
@item
A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
brackets ('[', ']') and ends with next section or end of file.
@item
At the beginning of a chapter section there may be an optional timebase to be
used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
den are integers. If the timebase is missing then start/end times are assumed to
be in milliseconds.
Next a chapter section must contain chapter start and end times in form
'START=num', 'END=num', where num is a positive integer.
@item
Empty lines and lines starting with ';' or '#' are ignored.
@item
Metadata keys or values containing special characters ('=', ';', '#', '\' and a
newline) must be escaped with a backslash '\'.
@item
Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
the tag (in the example above key is 'foo ', value is ' bar').
@end enumerate
A ffmetadata file might look like this:
@example
;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team
[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line
@end example
@c man end METADATA

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MIPS optimizations info
===============================================
MIPS optimizations of codecs are targeting MIPS 74k family of
CPUs. Some of these optimizations are relying more on properties of
this architecture and some are relying less (and can be used on most
MIPS architectures without degradation in performance).
Along with FFMPEG copyright notice, there is MIPS copyright notice in
all the files that are created by people from MIPS Technologies.
Example of copyright notice:
===============================================
/*
* Copyright (c) 2012
* MIPS Technologies, Inc., California.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* Author: Author Name (author_name@@mips.com)
*/
Files that have MIPS copyright notice in them:
===============================================
* libavutil/mips/
float_dsp_mips.c
libm_mips.h
* libavcodec/mips/
ac3dsp_mips.c
acelp_filters_mips.c
acelp_vectors_mips.c
amrwbdec_mips.c
amrwbdec_mips.h
celp_filters_mips.c
celp_math_mips.c
compute_antialias_fixed.h
compute_antialias_float.h
lsp_mips.h
dsputil_mips.c
fft_mips.c
fft_table.h
fft_init_table.c
fmtconvert_mips.c
mpegaudiodsp_mips_fixed.c
mpegaudiodsp_mips_float.c

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FFmpeg multithreading methods
==============================================
FFmpeg provides two methods for multithreading codecs.
Slice threading decodes multiple parts of a frame at the same time, using
AVCodecContext execute() and execute2().
Frame threading decodes multiple frames at the same time.
It accepts N future frames and delays decoded pictures by N-1 frames.
The later frames are decoded in separate threads while the user is
displaying the current one.
Restrictions on clients
==============================================
Slice threading -
* The client's draw_horiz_band() must be thread-safe according to the comment
in avcodec.h.
Frame threading -
* Restrictions with slice threading also apply.
* For best performance, the client should set thread_safe_callbacks if it
provides a thread-safe get_buffer() callback.
* There is one frame of delay added for every thread beyond the first one.
Clients must be able to handle this; the pkt_dts and pkt_pts fields in
AVFrame will work as usual.
Restrictions on codec implementations
==============================================
Slice threading -
None except that there must be something worth executing in parallel.
Frame threading -
* Codecs can only accept entire pictures per packet.
* Codecs similar to ffv1, whose streams don't reset across frames,
will not work because their bitstreams cannot be decoded in parallel.
* The contents of buffers must not be read before ff_thread_await_progress()
has been called on them. reget_buffer() and buffer age optimizations no longer work.
* The contents of buffers must not be written to after ff_thread_report_progress()
has been called on them. This includes draw_edges().
Porting codecs to frame threading
==============================================
Find all context variables that are needed by the next frame. Move all
code changing them, as well as code calling get_buffer(), up to before
the decode process starts. Call ff_thread_finish_setup() afterwards. If
some code can't be moved, have update_thread_context() run it in the next
thread.
If the codec allocates writable tables in its init(), add an init_thread_copy()
which re-allocates them for other threads.
Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
speed gain at this point but it should work.
Call ff_thread_report_progress() after some part of the current picture has decoded.
A good place to put this is where draw_horiz_band() is called - add this if it isn't
called anywhere, as it's useful too and the implementation is trivial when you're
doing this. Note that draw_edges() needs to be called before reporting progress.
Before accessing a reference frame or its MVs, call ff_thread_await_progress().

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@chapter Muxers
@c man begin MUXERS
Muxers are configured elements in FFmpeg which allow writing
multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option @code{--list-muxers}.
You can disable all the muxers with the configure option
@code{--disable-muxers} and selectively enable / disable single muxers
with the options @code{--enable-muxer=@var{MUXER}} /
@code{--disable-muxer=@var{MUXER}}.
The option @code{-formats} of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
@anchor{crc}
@section crc
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a single line of the form:
CRC=0x@var{CRC}, where @var{CRC} is a hexadecimal number 0-padded to
8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file
@file{out.crc}:
@example
ffmpeg -i INPUT -f crc out.crc
@end example
You can print the CRC to stdout with the command:
@example
ffmpeg -i INPUT -f crc -
@end example
You can select the output format of each frame with @command{ffmpeg} by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -
@end example
See also the @ref{framecrc} muxer.
@anchor{framecrc}
@section framecrc
Per-packet CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each audio
and video packet. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
CRC.
The output of the muxer consists of a line for each audio and video
packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, 0x@var{CRC}
@end example
@var{CRC} is a hexadecimal number 0-padded to 8 digits containing the
CRC of the packet.
For example to compute the CRC of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.crc}:
@example
ffmpeg -i INPUT -f framecrc out.crc
@end example
To print the information to stdout, use the command:
@example
ffmpeg -i INPUT -f framecrc -
@end example
With @command{ffmpeg}, you can select the output format to which the
audio and video frames are encoded before computing the CRC for each
packet by specifying the audio and video codec. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
@example
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -
@end example
See also the @ref{crc} muxer.
@anchor{framemd5}
@section framemd5
Per-packet MD5 testing format.
This muxer computes and prints the MD5 hash for each audio
and video packet. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
hash.
The output of the muxer consists of a line for each audio and video
packet of the form:
@example
@var{stream_index}, @var{packet_dts}, @var{packet_pts}, @var{packet_duration}, @var{packet_size}, @var{MD5}
@end example
@var{MD5} is a hexadecimal number representing the computed MD5 hash
for the packet.
For example to compute the MD5 of the audio and video frames in
@file{INPUT}, converted to raw audio and video packets, and store it
in the file @file{out.md5}:
@example
ffmpeg -i INPUT -f framemd5 out.md5
@end example
To print the information to stdout, use the command:
@example
ffmpeg -i INPUT -f framemd5 -
@end example
See also the @ref{md5} muxer.
@anchor{hls}
@section hls
Apple HTTP Live Streaming muxer that segments MPEG-TS according to
the HTTP Live Streaming specification.
It creates a playlist file and numbered segment files. The output
filename specifies the playlist filename; the segment filenames
receive the same basename as the playlist, a sequential number and
a .ts extension.
@example
ffmpeg -i in.nut out.m3u8
@end example
@table @option
@item -hls_time @var{seconds}
Set the segment length in seconds.
@item -hls_list_size @var{size}
Set the maximum number of playlist entries.
@item -hls_wrap @var{wrap}
Set the number after which index wraps.
@item -start_number @var{number}
Start the sequence from @var{number}.
@end table
@anchor{ico}
@section ico
ICO file muxer.
Microsoft's icon file format (ICO) has some strict limitations that should be noted:
@itemize
@item
Size cannot exceed 256 pixels in any dimension
@item
Only BMP and PNG images can be stored
@item
If a BMP image is used, it must be one of the following pixel formats:
@example
BMP Bit Depth FFmpeg Pixel Format
1bit pal8
4bit pal8
8bit pal8
16bit rgb555le
24bit bgr24
32bit bgra
@end example
@item
If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
@item
If a PNG image is used, it must use the rgba pixel format
@end itemize
@anchor{image2}
@section image2
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to
produce sequentially numbered series of files.
The pattern may contain the string "%d" or "%0@var{N}d", this string
specifies the position of the characters representing a numbering in
the filenames. If the form "%0@var{N}d" is used, the string
representing the number in each filename is 0-padded to @var{N}
digits. The literal character '%' can be specified in the pattern with
the string "%%".
If the pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified will contain the number 1, all the following
numbers will be sequential.
The pattern may contain a suffix which is used to automatically
determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.
The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
form @file{img%-1.jpg}, @file{img%-2.jpg}, ..., @file{img%-10.jpg},
etc.
The following example shows how to use @command{ffmpeg} for creating a
sequence of files @file{img-001.jpeg}, @file{img-002.jpeg}, ...,
taking one image every second from the input video:
@example
ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'
@end example
Note that with @command{ffmpeg}, if the format is not specified with the
@code{-f} option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
@example
ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'
@end example
Note also that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to create a single image file
@file{img.jpeg} from the input video you can employ the command:
@example
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
@end example
@table @option
@item -start_number @var{number}
Start the sequence from @var{number}.
@end table
The image muxer supports the .Y.U.V image file format. This format is
special in that that each image frame consists of three files, for
each of the YUV420P components. To read or write this image file format,
specify the name of the '.Y' file. The muxer will automatically open the
'.U' and '.V' files as required.
@anchor{md5}
@section md5
MD5 testing format.
This muxer computes and prints the MD5 hash of all the input audio
and video frames. By default audio frames are converted to signed
16-bit raw audio and video frames to raw video before computing the
hash.
The output of the muxer consists of a single line of the form:
MD5=@var{MD5}, where @var{MD5} is a hexadecimal number representing
the computed MD5 hash.
For example to compute the MD5 hash of the input converted to raw
audio and video, and store it in the file @file{out.md5}:
@example
ffmpeg -i INPUT -f md5 out.md5
@end example
You can print the MD5 to stdout with the command:
@example
ffmpeg -i INPUT -f md5 -
@end example
See also the @ref{framemd5} muxer.
@section MOV/MP4/ISMV
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
file has all the metadata about all packets stored in one location
(written at the end of the file, it can be moved to the start for
better playback by adding @var{faststart} to the @var{movflags}, or
using the @command{qt-faststart} tool). A fragmented
file consists of a number of fragments, where packets and metadata
about these packets are stored together. Writing a fragmented
file has the advantage that the file is decodable even if the
writing is interrupted (while a normal MOV/MP4 is undecodable if
it is not properly finished), and it requires less memory when writing
very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.
Fragmentation is enabled by setting one of the AVOptions that define
how to cut the file into fragments:
@table @option
@item -moov_size @var{bytes}
Reserves space for the moov atom at the beginning of the file instead of placing the
moov atom at the end. If the space reserved is insufficient, muxing will fail.
@item -movflags frag_keyframe
Start a new fragment at each video keyframe.
@item -frag_duration @var{duration}
Create fragments that are @var{duration} microseconds long.
@item -frag_size @var{size}
Create fragments that contain up to @var{size} bytes of payload data.
@item -movflags frag_custom
Allow the caller to manually choose when to cut fragments, by
calling @code{av_write_frame(ctx, NULL)} to write a fragment with
the packets written so far. (This is only useful with other
applications integrating libavformat, not from @command{ffmpeg}.)
@item -min_frag_duration @var{duration}
Don't create fragments that are shorter than @var{duration} microseconds long.
@end table
If more than one condition is specified, fragments are cut when
one of the specified conditions is fulfilled. The exception to this is
@code{-min_frag_duration}, which has to be fulfilled for any of the other
conditions to apply.
Additionally, the way the output file is written can be adjusted
through a few other options:
@table @option
@item -movflags empty_moov
Write an initial moov atom directly at the start of the file, without
describing any samples in it. Generally, an mdat/moov pair is written
at the start of the file, as a normal MOV/MP4 file, containing only
a short portion of the file. With this option set, there is no initial
mdat atom, and the moov atom only describes the tracks but has
a zero duration.
Files written with this option set do not work in QuickTime.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags separate_moof
Write a separate moof (movie fragment) atom for each track. Normally,
packets for all tracks are written in a moof atom (which is slightly
more efficient), but with this option set, the muxer writes one moof/mdat
pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
@item -movflags faststart
Run a second pass moving the moov atom on top of the file. This
operation can take a while, and will not work in various situations such
as fragmented output, thus it is not enabled by default.
@end table
Smooth Streaming content can be pushed in real time to a publishing
point on IIS with this muxer. Example:
@example
ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
@end example
@section mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
@table @option
@item -mpegts_original_network_id @var{number}
Set the original_network_id (default 0x0001). This is unique identifier
of a network in DVB. Its main use is in the unique identification of a
service through the path Original_Network_ID, Transport_Stream_ID.
@item -mpegts_transport_stream_id @var{number}
Set the transport_stream_id (default 0x0001). This identifies a
transponder in DVB.
@item -mpegts_service_id @var{number}
Set the service_id (default 0x0001) also known as program in DVB.
@item -mpegts_pmt_start_pid @var{number}
Set the first PID for PMT (default 0x1000, max 0x1f00).
@item -mpegts_start_pid @var{number}
Set the first PID for data packets (default 0x0100, max 0x0f00).
@end table
The recognized metadata settings in mpegts muxer are @code{service_provider}
and @code{service_name}. If they are not set the default for
@code{service_provider} is "FFmpeg" and the default for
@code{service_name} is "Service01".
@example
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
@end example
@section null
Null muxer.
This muxer does not generate any output file, it is mainly useful for
testing or benchmarking purposes.
For example to benchmark decoding with @command{ffmpeg} you can use the
command:
@example
ffmpeg -benchmark -i INPUT -f null out.null
@end example
Note that the above command does not read or write the @file{out.null}
file, but specifying the output file is required by the @command{ffmpeg}
syntax.
Alternatively you can write the command as:
@example
ffmpeg -benchmark -i INPUT -f null -
@end example
@section matroska
Matroska container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
@table @option
@item title=@var{title name}
Name provided to a single track
@end table
@table @option
@item language=@var{language name}
Specifies the language of the track in the Matroska languages form
@end table
@table @option
@item stereo_mode=@var{mode}
Stereo 3D video layout of two views in a single video track
@table @option
@item mono
video is not stereo
@item left_right
Both views are arranged side by side, Left-eye view is on the left
@item bottom_top
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
@item top_bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
@item checkerboard_rl
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
@item checkerboard_lr
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
@item row_interleaved_rl
Each view is constituted by a row based interleaving, Right-eye view is first row
@item row_interleaved_lr
Each view is constituted by a row based interleaving, Left-eye view is first row
@item col_interleaved_rl
Both views are arranged in a column based interleaving manner, Right-eye view is first column
@item col_interleaved_lr
Both views are arranged in a column based interleaving manner, Left-eye view is first column
@item anaglyph_cyan_red
All frames are in anaglyph format viewable through red-cyan filters
@item right_left
Both views are arranged side by side, Right-eye view is on the left
@item anaglyph_green_magenta
All frames are in anaglyph format viewable through green-magenta filters
@item block_lr
Both eyes laced in one Block, Left-eye view is first
@item block_rl
Both eyes laced in one Block, Right-eye view is first
@end table
@end table
For example a 3D WebM clip can be created using the following command line:
@example
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm
@end example
@section segment, stream_segment, ssegment
Basic stream segmenter.
The segmenter muxer outputs streams to a number of separate files of nearly
fixed duration. Output filename pattern can be set in a fashion similar to
@ref{image2}.
@code{stream_segment} is a variant of the muxer used to write to
streaming output formats, i.e. which do not require global headers,
and is recommended for outputting e.g. to MPEG transport stream segments.
@code{ssegment} is a shorter alias for @code{stream_segment}.
Every segment starts with a keyframe of the selected reference stream,
which is set through the @option{reference_stream} option.
Note that if you want accurate splitting for a video file, you need to
make the input key frames correspond to the exact splitting times
expected by the segmenter, or the segment muxer will start the new
segment with the key frame found next after the specified start
time.
The segment muxer works best with a single constant frame rate video.
Optionally it can generate a list of the created segments, by setting
the option @var{segment_list}. The list type is specified by the
@var{segment_list_type} option.
The segment muxer supports the following options:
@table @option
@item reference_stream @var{specifier}
Set the reference stream, as specified by the string @var{specifier}.
If @var{specifier} is set to @code{auto}, the reference is choosen
automatically. Otherwise it must be a stream specifier (see the ``Stream
specifiers'' chapter in the ffmpeg manual) which specifies the
reference stream. The default value is ``auto''.
@item segment_format @var{format}
Override the inner container format, by default it is guessed by the filename
extension.
@item segment_list @var{name}
Generate also a listfile named @var{name}. If not specified no
listfile is generated.
@item segment_list_flags @var{flags}
Set flags affecting the segment list generation.
It currently supports the following flags:
@table @var
@item cache
Allow caching (only affects M3U8 list files).
@item live
Allow live-friendly file generation.
This currently only affects M3U8 lists. In particular, write a fake
EXT-X-TARGETDURATION duration field at the top of the file, based on
the specified @var{segment_time}.
@end table
Default value is @code{cache}.
@item segment_list_size @var{size}
Overwrite the listfile once it reaches @var{size} entries. If 0
the listfile is never overwritten. Default value is 0.
@item segment_list type @var{type}
Specify the format for the segment list file.
The following values are recognized:
@table @option
@item flat
Generate a flat list for the created segments, one segment per line.
@item csv, ext
Generate a list for the created segments, one segment per line,
each line matching the format (comma-separated values):
@example
@var{segment_filename},@var{segment_start_time},@var{segment_end_time}
@end example
@var{segment_filename} is the name of the output file generated by the
muxer according to the provided pattern. CSV escaping (according to
RFC4180) is applied if required.
@var{segment_start_time} and @var{segment_end_time} specify
the segment start and end time expressed in seconds.
A list file with the suffix @code{".csv"} or @code{".ext"} will
auto-select this format.
@code{ext} is deprecated in favor or @code{csv}.
@item m3u8
Generate an extended M3U8 file, version 4, compliant with
@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt}.
A list file with the suffix @code{".m3u8"} will auto-select this format.
@end table
If not specified the type is guessed from the list file name suffix.
@item segment_time @var{time}
Set segment duration to @var{time}. Default value is "2".
@item segment_time_delta @var{delta}
Specify the accuracy time when selecting the start time for a
segment. Default value is "0".
When delta is specified a key-frame will start a new segment if its
PTS satisfies the relation:
@example
PTS >= start_time - time_delta
@end example
This option is useful when splitting video content, which is always
split at GOP boundaries, in case a key frame is found just before the
specified split time.
In particular may be used in combination with the @file{ffmpeg} option
@var{force_key_frames}. The key frame times specified by
@var{force_key_frames} may not be set accurately because of rounding
issues, with the consequence that a key frame time may result set just
before the specified time. For constant frame rate videos a value of
1/2*@var{frame_rate} should address the worst case mismatch between
the specified time and the time set by @var{force_key_frames}.
@item segment_times @var{times}
Specify a list of split points. @var{times} contains a list of comma
separated duration specifications, in increasing order.
@item segment_frames @var{frames}
Specify a list of split video frame numbers. @var{frames} contains a
list of comma separated integer numbers, in increasing order.
This option specifies to start a new segment whenever a reference
stream key frame is found and the sequential number (starting from 0)
of the frame is greater or equal to the next value in the list.
@item segment_wrap @var{limit}
Wrap around segment index once it reaches @var{limit}.
@item segment_start_number @var{number}
Set the sequence number of the first segment. Defaults to @code{0}.
@item reset_timestamps @var{1|0}
Reset timestamps at the begin of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the playback
of the generated segments. May not work with some combinations of
muxers/codecs. It is set to @code{0} by default.
@end table
@section Examples
@itemize
@item
To remux the content of file @file{in.mkv} to a list of segments
@file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
generated segments to @file{out.list}:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
@end example
@item
As the example above, but segment the input file according to the split
points specified by the @var{segment_times} option:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
@end example
@item
As the example above, but use the @code{ffmpeg} @var{force_key_frames}
option to force key frames in the input at the specified location, together
with the segment option @var{segment_time_delta} to account for
possible roundings operated when setting key frame times.
@example
ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
-f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut
@end example
In order to force key frames on the input file, transcoding is
required.
@item
Segment the input file by splitting the input file according to the
frame numbers sequence specified with the @var{segment_frames} option:
@example
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
@end example
@item
To convert the @file{in.mkv} to TS segments using the @code{libx264}
and @code{libfaac} encoders:
@example
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
@end example
@item
Segment the input file, and create an M3U8 live playlist (can be used
as live HLS source):
@example
ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
-segment_list_flags +live -segment_time 10 out%03d.mkv
@end example
@end itemize
@section mp3
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
@code{id3v2_version} option controls which one is used. The legacy ID3v1 tag is
not written by default, but may be enabled with the @code{write_id3v1} option.
For seekable output the muxer also writes a Xing frame at the beginning, which
contains the number of frames in the file. It is useful for computing duration
of VBR files.
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
are supplied to the muxer in form of a video stream with a single packet. There
can be any number of those streams, each will correspond to a single APIC frame.
The stream metadata tags @var{title} and @var{comment} map to APIC
@var{description} and @var{picture type} respectively. See
@url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
Note that the APIC frames must be written at the beginning, so the muxer will
buffer the audio frames until it gets all the pictures. It is therefore advised
to provide the pictures as soon as possible to avoid excessive buffering.
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@example
ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3
@end example
To attach a picture to an mp3 file select both the audio and the picture stream
with @code{map}:
@example
ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
-metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3
@end example
@c man end MUXERS

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\input texinfo @c -*- texinfo -*-
@settitle NUT
@titlepage
@center @titlefont{NUT}
@end titlepage
@top
@contents
@chapter Description
NUT is a low overhead generic container format. It stores audio, video,
subtitle and user-defined streams in a simple, yet efficient, way.
It was created by a group of FFmpeg and MPlayer developers in 2003
and was finalized in 2008.
The official nut specification is at svn://svn.mplayerhq.hu/nut
In case of any differences between this text and the official specification,
the official specification shall prevail.
@chapter Container-specific codec tags
@section Generic raw YUVA formats
Since many exotic planar YUVA pixel formats are not considered by
the AVI/QuickTime FourCC lists, the following scheme is adopted for
representing them.
The first two bytes can contain the values:
Y1 = only Y
Y2 = Y+A
Y3 = YUV
Y4 = YUVA
The third byte represents the width and height chroma subsampling
values for the UV planes, that is the amount to shift the luma
width/height right to find the chroma width/height.
The fourth byte is the number of bits used (8, 16, ...).
If the order of bytes is inverted, that means that each component has
to be read big-endian.
@section Raw Audio
@multitable @columnfractions .4 .4
@item ALAW @tab A-LAW
@item ULAW @tab MU-LAW
@item P<type><interleaving><bits> @tab little-endian PCM
@item <bits><interleaving><type>P @tab big-endian PCM
@end multitable
<type> is S for signed integer, U for unsigned integer, F for IEEE float
<interleaving> is D for default, P is for planar.
<bits> is 8/16/24/32
@example
PFD[32] would for example be signed 32 bit little-endian IEEE float
@end example
@section Subtitles
@multitable @columnfractions .4 .4
@item UTF8 @tab Raw UTF-8
@item SSA[0] @tab SubStation Alpha
@item DVDS @tab DVD subtitles
@item DVBS @tab DVB subtitles
@end multitable
@section Raw Data
@multitable @columnfractions .4 .4
@item UTF8 @tab Raw UTF-8
@end multitable
@section Codecs
@multitable @columnfractions .4 .4
@item 3IV1 @tab non-compliant MPEG-4 generated by old 3ivx
@item ASV1 @tab Asus Video
@item ASV2 @tab Asus Video 2
@item CVID @tab Cinepak
@item CYUV @tab Creative YUV
@item DIVX @tab non-compliant MPEG-4 generated by old DivX
@item DUCK @tab Truemotion 1
@item FFV1 @tab FFmpeg video 1
@item FFVH @tab FFmpeg Huffyuv
@item H261 @tab ITU H.261
@item H262 @tab ITU H.262
@item H263 @tab ITU H.263
@item H264 @tab ITU H.264
@item HFYU @tab Huffyuv
@item I263 @tab Intel H.263
@item IV31 @tab Indeo 3.1
@item IV32 @tab Indeo 3.2
@item IV50 @tab Indeo 5.0
@item LJPG @tab ITU JPEG (lossless)
@item MJLS @tab ITU JPEG-LS
@item MJPG @tab ITU JPEG
@item MPG4 @tab MS MPEG-4v1 (not ISO MPEG-4)
@item MP42 @tab MS MPEG-4v2
@item MP43 @tab MS MPEG-4v3
@item MP4V @tab ISO MPEG-4 Part 2 Video (from old encoders)
@item mpg1 @tab ISO MPEG-1 Video
@item mpg2 @tab ISO MPEG-2 Video
@item MRLE @tab MS RLE
@item MSVC @tab MS Video 1
@item RT21 @tab Indeo 2.1
@item RV10 @tab RealVideo 1.0
@item RV20 @tab RealVideo 2.0
@item RV30 @tab RealVideo 3.0
@item RV40 @tab RealVideo 4.0
@item SNOW @tab FFmpeg Snow
@item SVQ1 @tab Sorenson Video 1
@item SVQ3 @tab Sorenson Video 3
@item theo @tab Xiph Theora
@item TM20 @tab Truemotion 2.0
@item UMP4 @tab non-compliant MPEG-4 generated by UB Video MPEG-4
@item VCR1 @tab ATI VCR1
@item VP30 @tab VP 3.0
@item VP31 @tab VP 3.1
@item VP50 @tab VP 5.0
@item VP60 @tab VP 6.0
@item VP61 @tab VP 6.1
@item VP62 @tab VP 6.2
@item VP70 @tab VP 7.0
@item WMV1 @tab MS WMV7
@item WMV2 @tab MS WMV8
@item WMV3 @tab MS WMV9
@item WV1F @tab non-compliant MPEG-4 generated by ?
@item WVC1 @tab VC-1
@item XVID @tab non-compliant MPEG-4 generated by old Xvid
@item XVIX @tab non-compliant MPEG-4 generated by old Xvid with interlacing bug
@end multitable

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optimization Tips (for libavcodec):
===================================
What to optimize:
-----------------
If you plan to do non-x86 architecture specific optimizations (SIMD normally),
then take a look in the x86/ directory, as most important functions are
already optimized for MMX.
If you want to do x86 optimizations then you can either try to finetune the
stuff in the x86 directory or find some other functions in the C source to
optimize, but there aren't many left.
Understanding these overoptimized functions:
--------------------------------------------
As many functions tend to be a bit difficult to understand because
of optimizations, it can be hard to optimize them further, or write
architecture-specific versions. It is recommended to look at older
revisions of the interesting files (web frontends for the various FFmpeg
branches are listed at http://ffmpeg.org/download.html).
Alternatively, look into the other architecture-specific versions in
the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
comprehend the instructions, it could help understanding the functions
and how they can be optimized.
NOTE: If you still don't understand some function, ask at our mailing list!!!
(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel)
When is an optimization justified?
----------------------------------
Normally, clean and simple optimizations for widely used codecs are
justified even if they only achieve an overall speedup of 0.1%. These
speedups accumulate and can make a big difference after awhile. Also, if
none of the following factors get worse due to an optimization -- speed,
binary code size, source size, source readability -- and at least one
factor improves, then an optimization is always a good idea even if the
overall gain is less than 0.1%. For obscure codecs that are not often
used, the goal is more toward keeping the code clean, small, and
readable instead of making it 1% faster.
WTF is that function good for ....:
-----------------------------------
The primary purpose of this list is to avoid wasting time optimizing functions
which are rarely used.
put(_no_rnd)_pixels{,_x2,_y2,_xy2}
Used in motion compensation (en/decoding).
avg_pixels{,_x2,_y2,_xy2}
Used in motion compensation of B-frames.
These are less important than the put*pixels functions.
avg_no_rnd_pixels*
unused
pix_abs16x16{,_x2,_y2,_xy2}
Used in motion estimation (encoding) with SAD.
pix_abs8x8{,_x2,_y2,_xy2}
Used in motion estimation (encoding) with SAD of MPEG-4 4MV only.
These are less important than the pix_abs16x16* functions.
put_mspel8_mc* / wmv2_mspel8*
Used only in WMV2.
it is not recommended that you waste your time with these, as WMV2
is an ugly and relatively useless codec.
mpeg4_qpel* / *qpel_mc*
Used in MPEG-4 qpel motion compensation (encoding & decoding).
The qpel8 functions are used only for 4mv,
the avg_* functions are used only for B-frames.
Optimizing them should have a significant impact on qpel
encoding & decoding.
qpel{8,16}_mc??_old_c / *pixels{8,16}_l4
Just used to work around a bug in an old libavcodec encoder version.
Don't optimize them.
tpel_mc_func {put,avg}_tpel_pixels_tab
Used only for SVQ3, so only optimize them if you need fast SVQ3 decoding.
add_bytes/diff_bytes
For huffyuv only, optimize if you want a faster ffhuffyuv codec.
get_pixels / diff_pixels
Used for encoding, easy.
clear_blocks
easiest to optimize
gmc
Used for MPEG-4 gmc.
Optimizing this should have a significant effect on the gmc decoding
speed.
gmc1
Used for chroma blocks in MPEG-4 gmc with 1 warp point
(there are 4 luma & 2 chroma blocks per macroblock, so
only 1/3 of the gmc blocks use this, the other 2/3
use the normal put_pixel* code, but only if there is
just 1 warp point).
Note: DivX5 gmc always uses just 1 warp point.
pix_sum
Used for encoding.
hadamard8_diff / sse / sad == pix_norm1 / dct_sad / quant_psnr / rd / bit
Specific compare functions used in encoding, it depends upon the
command line switches which of these are used.
Don't waste your time with dct_sad & quant_psnr, they aren't
really useful.
put_pixels_clamped / add_pixels_clamped
Used for en/decoding in the IDCT, easy.
Note, some optimized IDCTs have the add/put clamped code included and
then put_pixels_clamped / add_pixels_clamped will be unused.
idct/fdct
idct (encoding & decoding)
fdct (encoding)
difficult to optimize
dct_quantize_trellis
Used for encoding with trellis quantization.
difficult to optimize
dct_quantize
Used for encoding.
dct_unquantize_mpeg1
Used in MPEG-1 en/decoding.
dct_unquantize_mpeg2
Used in MPEG-2 en/decoding.
dct_unquantize_h263
Used in MPEG-4/H.263 en/decoding.
FIXME remaining functions?
BTW, most of these functions are in dsputil.c/.h, some are in mpegvideo.c/.h.
Alignment:
Some instructions on some architectures have strict alignment restrictions,
for example most SSE/SSE2 instructions on x86.
The minimum guaranteed alignment is written in the .h files, for example:
void (*put_pixels_clamped)(const DCTELEM *block/*align 16*/, UINT8 *pixels/*align 8*/, int line_size);
General Tips:
-------------
Use asm loops like:
__asm__(
"1: ....
...
"jump_instruction ....
Do not use C loops:
do{
__asm__(
...
}while()
For x86, mark registers that are clobbered in your asm. This means both
general x86 registers (e.g. eax) as well as XMM registers. This last one is
particularly important on Win64, where xmm6-15 are callee-save, and not
restoring their contents leads to undefined results. In external asm (e.g.
yasm), you do this by using:
cglobal functon_name, num_args, num_regs, num_xmm_regs
In inline asm, you specify clobbered registers at the end of your asm:
__asm__(".." ::: "%eax").
If gcc is not set to support sse (-msse) it will not accept xmm registers
in the clobber list. For that we use two macros to declare the clobbers.
XMM_CLOBBERS should be used when there are other clobbers, for example:
__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax");
and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers:
__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0"));
Do not expect a compiler to maintain values in your registers between separate
(inline) asm code blocks. It is not required to. For example, this is bad:
__asm__("movdqa %0, %%xmm7" : src);
/* do something */
__asm__("movdqa %%xmm7, %1" : dst);
- first of all, you're assuming that the compiler will not use xmm7 in
between the two asm blocks. It probably won't when you test it, but it's
a poor assumption that will break at some point for some --cpu compiler flag
- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would
have restored the original value of xmm7 after the first asm block, thus
rendering the combination of the two blocks of code invalid
Code that depends on data in registries being untouched, should be written as
a single __asm__() statement. Ideally, a single function contains only one
__asm__() block.
Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
The latter requires a good optimizing compiler which gcc is not.
Inline asm vs. external asm
---------------------------
Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
and external asm (.s or .asm files, handled by an assembler such as yasm/nasm)
are accepted in FFmpeg. Which one to use differs per specific case.
- if your code is intended to be inlined in a C function, inline asm is always
better, because external asm cannot be inlined
- if your code calls external functions, yasm is always better
- if your code takes huge and complex structs as function arguments (e.g.
MpegEncContext; note that this is not ideal and is discouraged if there
are alternatives), then inline asm is always better, because predicting
member offsets in complex structs is almost impossible. It's safest to let
the compiler take care of that
- in many cases, both can be used and it just depends on the preference of the
person writing the asm. For new asm, the choice is up to you. For existing
asm, you'll likely want to maintain whatever form it is currently in unless
there is a good reason to change it.
- if, for some reason, you believe that a particular chunk of existing external
asm could be improved upon further if written in inline asm (or the other
way around), then please make the move from external asm <-> inline asm a
separate patch before your patches that actually improve the asm.
Links:
======
http://www.aggregate.org/MAGIC/
x86-specific:
-------------
http://developer.intel.com/design/pentium4/manuals/248966.htm
The IA-32 Intel Architecture Software Developer's Manual, Volume 2:
Instruction Set Reference
http://developer.intel.com/design/pentium4/manuals/245471.htm
http://www.agner.org/assem/
AMD Athlon Processor x86 Code Optimization Guide:
http://www.amd.com/us-en/assets/content_type/white_papers_and_tech_docs/22007.pdf
ARM-specific:
-------------
ARM Architecture Reference Manual (up to ARMv5TE):
http://www.arm.com/community/university/eulaarmarm.html
Procedure Call Standard for the ARM Architecture:
http://www.arm.com/pdfs/aapcs.pdf
Optimization guide for ARM9E (used in Nokia 770 Internet Tablet):
http://infocenter.arm.com/help/topic/com.arm.doc.ddi0240b/DDI0240A.pdf
Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
http://download.intel.com/design/intelxscale/27347302.pdf
Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual
http://download.intel.com/design/intelxscale/31451001.pdf
PowerPC-specific:
-----------------
PowerPC32/AltiVec PIM:
www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPEM.pdf
PowerPC32/AltiVec PEM:
www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPIM.pdf
CELL/SPU:
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
SPARC-specific:
---------------
SPARC Joint Programming Specification (JPS1): Commonality
http://www.fujitsu.com/downloads/PRMPWR/JPS1-R1.0.4-Common-pub.pdf
UltraSPARC III Processor User's Manual (contains instruction timings)
http://www.sun.com/processors/manuals/USIIIv2.pdf
VIS Whitepaper (contains optimization guidelines)
http://www.sun.com/processors/vis/download/vis/vis_whitepaper.pdf
GCC asm links:
--------------
official doc but quite ugly
http://gcc.gnu.org/onlinedocs/gcc/Extended-Asm.html
a bit old (note "+" is valid for input-output, even though the next disagrees)
http://www.cs.virginia.edu/~clc5q/gcc-inline-asm.pdf

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@chapter Output Devices
@c man begin OUTPUT DEVICES
Output devices are configured elements in FFmpeg which allow to write
multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices
are enabled by default. You can list all available ones using the
configure option "--list-outdevs".
You can disable all the output devices using the configure option
"--disable-outdevs", and selectively enable an output device using the
option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
input device using the option "--disable-outdev=@var{OUTDEV}".
The option "-formats" of the ff* tools will display the list of
enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) output device.
@section caca
CACA output device.
This output devices allows to show a video stream in CACA window.
Only one CACA window is allowed per application, so you can
have only one instance of this output device in an application.
To enable this output device you need to configure FFmpeg with
@code{--enable-libcaca}.
libcaca is a graphics library that outputs text instead of pixels.
For more information about libcaca, check:
@url{http://caca.zoy.org/wiki/libcaca}
@subsection Options
@table @option
@item window_title
Set the CACA window title, if not specified default to the filename
specified for the output device.
@item window_size
Set the CACA window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video.
@item driver
Set display driver.
@item algorithm
Set dithering algorithm. Dithering is necessary
because the picture being rendered has usually far more colours than
the available palette.
The accepted values are listed with @code{-list_dither algorithms}.
@item antialias
Set antialias method. Antialiasing smoothens the rendered
image and avoids the commonly seen staircase effect.
The accepted values are listed with @code{-list_dither antialiases}.
@item charset
Set which characters are going to be used when rendering text.
The accepted values are listed with @code{-list_dither charsets}.
@item color
Set color to be used when rendering text.
The accepted values are listed with @code{-list_dither colors}.
@item list_drivers
If set to @option{true}, print a list of available drivers and exit.
@item list_dither
List available dither options related to the argument.
The argument must be one of @code{algorithms}, @code{antialiases},
@code{charsets}, @code{colors}.
@end table
@subsection Examples
@itemize
@item
The following command shows the @command{ffmpeg} output is an
CACA window, forcing its size to 80x25:
@example
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
@end example
@item
Show the list of available drivers and exit:
@example
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
@end example
@item
Show the list of available dither colors and exit:
@example
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize
@section oss
OSS (Open Sound System) output device.
@section sdl
SDL (Simple DirectMedia Layer) output device.
This output devices allows to show a video stream in an SDL
window. Only one SDL window is allowed per application, so you can
have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system
when configuring your build.
For more information about SDL, check:
@url{http://www.libsdl.org/}
@subsection Options
@table @option
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item icon_title
Set the name of the iconified SDL window, if not specified it is set
to the same value of @var{window_title}.
@item window_size
Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video,
downscaled according to the aspect ratio.
@end table
@subsection Examples
The following command shows the @command{ffmpeg} output is an
SDL window, forcing its size to the qcif format:
@example
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
@end example
@section sndio
sndio audio output device.
@c man end OUTPUT DEVICES

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\input texinfo @c -*- texinfo -*-
@settitle Platform Specific Information
@titlepage
@center @titlefont{Platform Specific Information}
@end titlepage
@top
@contents
@chapter Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@command{gmake}).
@section (Open)Solaris
GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@anchor{Darwin}
@section Darwin (Mac OS X, iPhone)
The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{http://github.com/yuvi/gas-preprocessor} to build the optimized
assembler functions. Just download the Perl script and put it somewhere
in your PATH, FFmpeg's configure will pick it up automatically.
Mac OS X on amd64 and x86 requires @command{yasm} to build most of the
optimized assembler functions. @uref{http://www.finkproject.org/, Fink},
@uref{http://www.gentoo.org/proj/en/gentoo-alt/prefix/bootstrap-macos.xml, Gentoo Prefix},
@uref{http://mxcl.github.com/homebrew/, Homebrew}
or @uref{http://www.macports.org, MacPorts} can easily provide it.
@chapter DOS
Using a cross-compiler is preferred for various reasons.
@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html}
@chapter OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
@section Native Windows compilation using MinGW or MinGW-w64
FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
You can find detailed installation instructions in the download section and
the FAQ.
Notes:
@itemize
@item Building natively using MSYS can be sped up by disabling implicit rules
in the Makefile by calling @code{make -r} instead of plain @code{make}. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example during
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
@item By using @code{./configure --enable-shared} when configuring FFmpeg,
you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
libavformat) as DLLs.
@end itemize
@section Microsoft Visual C++
FFmpeg can be built with MSVC using a C99-to-C89 conversion utility and
wrapper.
You will need the following prerequisites:
@itemize
@item @uref{http://download.videolan.org/pub/contrib/c99-to-c89/, C99-to-C89 Converter & Wrapper}
@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
@item @uref{http://www.mingw.org/, MSYS}
@item @uref{http://yasm.tortall.net/, YASM}
@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
you want to run @uref{fate.html, FATE}.
@end itemize
To set up a proper MSVC environment in MSYS, you simply need to run
@code{msys.bat} from the Visual Studio command prompt.
Place @code{makedef}, @code{c99wrap.exe}, @code{c99conv.exe}, and @code{yasm.exe}
somewhere in your @code{PATH}.
Next, make sure @code{inttypes.h} and any other headers and libs you want to use
are located in a spot that MSVC can see. Do so by modifying the @code{LIB} and
@code{INCLUDE} environment variables to include the @strong{Windows} paths to
these directories. Alternatively, you can try and use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.
Finally, run:
@example
./configure --toolchain=msvc
make
make install
@end example
If you wish to compile shared libraries, add @code{--enable-shared} to your
configure options. Note that due to the way MSVC handles DLL imports and
exports, you cannot compile static and shared libraries at the same time, and
enabling shared libraries will automatically disable the static ones.
Notes:
@itemize
@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
You can find out by running @code{which link} to see which @code{link.exe} you
are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
@item If you wish to build with zlib support, you will have to grab a compatible
zlib binary from somewhere, with an MSVC import lib, or if you wish to link
statically, you can follow the instructions below to build a compatible
@code{zlib.lib} with MSVC. Regardless of which method you use, you must still
follow step 3, or compilation will fail.
@enumerate
@item Grab the @uref{http://zlib.net/, zlib sources}.
@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since
this is how FFmpeg is built as well.
@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets
erroneously included when building FFmpeg.
@item Run @code{nmake -f win32/Makefile.msc}.
@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC
can see.
@end enumerate
@item FFmpeg has been tested with Visual Studio 2010 and 2012, Pro and Express.
Anything else is not officially supported.
@end itemize
@subsection Linking to FFmpeg with Microsoft Visual C++
If you plan to link with MSVC-built static libraries, you will need
to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.
FFmpeg headers do not declare global data for Windows DLLs through the usual
dllexport/dllimport interface. Such data will be exported properly while
building, but to use them in your MSVC code you will have to edit the
appropriate headers and mark the data as dllimport. For example, in
libavutil/pixdesc.h you should have:
@example
extern __declspec(dllimport) const AVPixFmtDescriptor av_pix_fmt_descriptors[];
@end example
You will also need to define @code{inline} to something MSVC understands:
@example
#define inline __inline
@end example
Also note, that as stated in @strong{Microsoft Visual C++}, you will need
an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}.
If you plan on using import libraries created by dlltool, you must
set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization
settings, otherwise the resulting binaries will fail during runtime.
This is not required when using import libraries generated by @code{lib.exe}.
This issue is reported upstream at
@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
To create import libraries that work with the @code{/OPT:REF} option
(which is enabled by default in Release mode), follow these steps:
@enumerate
@item Open the @emph{Visual Studio Command Prompt}.
Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
which sets up the environment variables for the Visual C++ tools
(the standard location for this file is something like
@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}).
@item Enter the @file{bin} directory where the created LIB and DLL files
are stored.
@item Generate new import libraries with @command{lib.exe}:
@example
lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib
@end example
Replace @code{foo-version} and @code{foo} with the respective library names.
@end enumerate
@anchor{Cross compilation for Windows with Linux}
@section Cross compilation for Windows with Linux
You must use the MinGW cross compilation tools available at
@url{http://www.mingw.org/}.
Then configure FFmpeg with the following options:
@example
./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
@end example
(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}.
@section Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, git, mingw-runtime, texi2html
@end example
In order to run FATE you will also need the following "Utils" packages:
@example
bc, diffutils
@end example
If you want to build FFmpeg with additional libraries, download Cygwin
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
libogg-devel, libvorbis-devel
@end example
These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
yasm, libSDL-devel, libfaac-devel, libaacplus-devel, libgsm-devel, libmp3lame-devel,
libschroedinger1.0-devel, speex-devel, libtheora-devel, libxvidcore-devel
@end example
The recommendation for x264 is to build it from source, as it evolves too
quickly for Cygwin Ports to be up to date.
@section Crosscompilation for Windows under Cygwin
With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
Just install your Cygwin as explained before, plus these additional
"Devel" packages:
@example
gcc-mingw-core, mingw-runtime, mingw-zlib
@end example
and add some special flags to your configure invocation.
For a static build run
@example
./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
and for a build with shared libraries
@example
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@chapter Plan 9
The native @uref{http://plan9.bell-labs.com/plan9/, Plan 9} compiler
does not implement all the C99 features needed by FFmpeg so the gcc
port must be used. Furthermore, a few items missing from the C
library and shell environment need to be fixed.
@itemize
@item GNU awk, grep, make, and sed
Working packages of these tools can be found at
@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}.
They can be installed with @uref{http://9front.org/, 9front's} @code{pkg}
utility by setting @code{pkgpath} to
@code{http://ports2plan9.googlecode.com/files/}.
@item Missing/broken @code{head} and @code{printf} commands
Replacements adequate for building FFmpeg can be found in the
@code{compat/plan9} directory. Place these somewhere they will be
found by the shell. These are not full implementations of the
commands and are @emph{not} suitable for general use.
@item Missing C99 @code{stdint.h} and @code{inttypes.h}
Replacement headers are available from
@url{http://code.google.com/p/plan9front/issues/detail?id=152}.
@item Missing or non-standard library functions
Some functions in the C library are missing or incomplete. The
@code{@uref{http://ports2plan9.googlecode.com/files/gcc-apelibs-1207.tbz,
gcc-apelibs-1207}} package from
@uref{http://code.google.com/p/ports2plan9/downloads/list, ports2plan9}
includes an updated C library, but installing the full package gives
unusable executables. Instead, keep the files from @code{gccbin.tgz}
under @code{/386/lib/gnu}. From the @code{libc.a} archive in the
@code{gcc-apelibs-1207} package, extract the following object files and
turn them into a library:
@itemize
@item @code{strerror.o}
@item @code{strtoll.o}
@item @code{snprintf.o}
@item @code{vsnprintf.o}
@item @code{vfprintf.o}
@item @code{_IO_getc.o}
@item @code{_IO_putc.o}
@end itemize
Use the @code{--extra-libs} option of @code{configure} to inform the
build system of this library.
@item FPU exceptions enabled by default
Unlike most other systems, Plan 9 enables FPU exceptions by default.
These must be disabled before calling any FFmpeg functions. While the
included tools will do this automatically, other users of the
libraries must do it themselves.
@end itemize
@bye

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project/jni/ffmpeg/doc/print_options.c Normal file
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/*
* Copyright (c) 2012 Anton Khirnov
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/*
* generate texinfo manpages for avoptions
*/
#include <stddef.h>
#include <string.h>
#include <float.h>
#include "libavformat/avformat.h"
#include "libavcodec/avcodec.h"
#include "libavutil/opt.h"
static void print_usage(void)
{
fprintf(stderr, "Usage: enum_options type\n"
"type: format codec\n");
exit(1);
}
static void print_option(const AVOption *opts, const AVOption *o, int per_stream)
{
printf("@item -%s%s @var{", o->name, per_stream ? "[:stream_specifier]" : "");
switch (o->type) {
case AV_OPT_TYPE_BINARY: printf("hexadecimal string"); break;
case AV_OPT_TYPE_STRING: printf("string"); break;
case AV_OPT_TYPE_INT:
case AV_OPT_TYPE_INT64: printf("integer"); break;
case AV_OPT_TYPE_FLOAT:
case AV_OPT_TYPE_DOUBLE: printf("float"); break;
case AV_OPT_TYPE_RATIONAL: printf("rational number"); break;
case AV_OPT_TYPE_FLAGS: printf("flags"); break;
default: printf("value"); break;
}
printf("} (@emph{");
if (o->flags & AV_OPT_FLAG_DECODING_PARAM) {
printf("input");
if (o->flags & AV_OPT_FLAG_ENCODING_PARAM)
printf("/");
}
if (o->flags & AV_OPT_FLAG_ENCODING_PARAM) printf("output");
if (o->flags & AV_OPT_FLAG_AUDIO_PARAM) printf(",audio");
if (o->flags & AV_OPT_FLAG_VIDEO_PARAM) printf(",video");
if (o->flags & AV_OPT_FLAG_SUBTITLE_PARAM) printf(",subtitles");
printf("})\n");
if (o->help)
printf("%s\n", o->help);
if (o->unit) {
const AVOption *u;
printf("\nPossible values:\n@table @samp\n");
for (u = opts; u->name; u++) {
if (u->type == AV_OPT_TYPE_CONST && u->unit && !strcmp(u->unit, o->unit))
printf("@item %s\n%s\n", u->name, u->help ? u->help : "");
}
printf("@end table\n");
}
}
static void show_opts(const AVOption *opts, int per_stream)
{
const AVOption *o;
printf("@table @option\n");
for (o = opts; o->name; o++) {
if (o->type != AV_OPT_TYPE_CONST)
print_option(opts, o, per_stream);
}
printf("@end table\n");
}
static void show_format_opts(void)
{
#include "libavformat/options_table.h"
printf("@section Format AVOptions\n");
show_opts(options, 0);
}
static void show_codec_opts(void)
{
#include "libavcodec/options_table.h"
printf("@section Codec AVOptions\n");
show_opts(options, 1);
}
int main(int argc, char **argv)
{
if (argc < 2)
print_usage();
printf("@c DO NOT EDIT THIS FILE!\n"
"@c It was generated by print_options.\n\n");
if (!strcmp(argv[1], "format"))
show_format_opts();
else if (!strcmp(argv[1], "codec"))
show_codec_opts();
else
print_usage();
return 0;
}

733
project/jni/ffmpeg/doc/protocols.texi Normal file
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@chapter Protocols
@c man begin PROTOCOLS
Protocols are configured elements in FFmpeg which allow to access
resources which require the use of a particular protocol.
When you configure your FFmpeg build, all the supported protocols are
enabled by default. You can list all available ones using the
configure option "--list-protocols".
You can disable all the protocols using the configure option
"--disable-protocols", and selectively enable a protocol using the
option "--enable-protocol=@var{PROTOCOL}", or you can disable a
particular protocol using the option
"--disable-protocol=@var{PROTOCOL}".
The option "-protocols" of the ff* tools will display the list of
supported protocols.
A description of the currently available protocols follows.
@section bluray
Read BluRay playlist.
The accepted options are:
@table @option
@item angle
BluRay angle
@item chapter
Start chapter (1...N)
@item playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)
@end table
Examples:
Read longest playlist from BluRay mounted to /mnt/bluray:
@example
bluray:/mnt/bluray
@end example
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
@example
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
@end example
@section concat
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were
a unique resource.
A URL accepted by this protocol has the syntax:
@example
concat:@var{URL1}|@var{URL2}|...|@var{URLN}
@end example
where @var{URL1}, @var{URL2}, ..., @var{URLN} are the urls of the
resource to be concatenated, each one possibly specifying a distinct
protocol.
For example to read a sequence of files @file{split1.mpeg},
@file{split2.mpeg}, @file{split3.mpeg} with @command{ffplay} use the
command:
@example
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
@end example
Note that you may need to escape the character "|" which is special for
many shells.
@section data
Data in-line in the URI. See @url{http://en.wikipedia.org/wiki/Data_URI_scheme}.
For example, to convert a GIF file given inline with @command{ffmpeg}:
@example
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
@end example
@section file
File access protocol.
Allow to read from or read to a file.
For example to read from a file @file{input.mpeg} with @command{ffmpeg}
use the command:
@example
ffmpeg -i file:input.mpeg output.mpeg
@end example
The ff* tools default to the file protocol, that is a resource
specified with the name "FILE.mpeg" is interpreted as the URL
"file:FILE.mpeg".
@section gopher
Gopher protocol.
@section hls
Read Apple HTTP Live Streaming compliant segmented stream as
a uniform one. The M3U8 playlists describing the segments can be
remote HTTP resources or local files, accessed using the standard
file protocol.
The nested protocol is declared by specifying
"+@var{proto}" after the hls URI scheme name, where @var{proto}
is either "file" or "http".
@example
hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8
@end example
Using this protocol is discouraged - the hls demuxer should work
just as well (if not, please report the issues) and is more complete.
To use the hls demuxer instead, simply use the direct URLs to the
m3u8 files.
@section http
HTTP (Hyper Text Transfer Protocol).
@section mmst
MMS (Microsoft Media Server) protocol over TCP.
@section mmsh
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
@example
mmsh://@var{server}[:@var{port}][/@var{app}][/@var{playpath}]
@end example
@section md5
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes
this to the designated output or stdout if none is specified. It can
be used to test muxers without writing an actual file.
Some examples follow.
@example
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:
@end example
Note that some formats (typically MOV) require the output protocol to
be seekable, so they will fail with the MD5 output protocol.
@section pipe
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
@example
pipe:[@var{number}]
@end example
@var{number} is the number corresponding to the file descriptor of the
pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If @var{number}
is not specified, by default the stdout file descriptor will be used
for writing, stdin for reading.
For example to read from stdin with @command{ffmpeg}:
@example
cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
@end example
For writing to stdout with @command{ffmpeg}:
@example
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
@end example
Note that some formats (typically MOV), require the output protocol to
be seekable, so they will fail with the pipe output protocol.
@section rtmp
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia
content across a TCP/IP network.
The required syntax is:
@example
rtmp://@var{server}[:@var{port}][/@var{app}][/@var{instance}][/@var{playpath}]
@end example
The accepted parameters are:
@table @option
@item server
The address of the RTMP server.
@item port
The number of the TCP port to use (by default is 1935).
@item app
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
(e.g. @file{/ondemand/}, @file{/flash/live/}, etc.). You can override
the value parsed from the URI through the @code{rtmp_app} option, too.
@item playpath
It is the path or name of the resource to play with reference to the
application specified in @var{app}, may be prefixed by "mp4:". You
can override the value parsed from the URI through the @code{rtmp_playpath}
option, too.
@item listen
Act as a server, listening for an incoming connection.
@item timeout
Maximum time to wait for the incoming connection. Implies listen.
@end table
Additionally, the following parameters can be set via command line options
(or in code via @code{AVOption}s):
@table @option
@item rtmp_app
Name of application to connect on the RTMP server. This option
overrides the parameter specified in the URI.
@item rtmp_buffer
Set the client buffer time in milliseconds. The default is 3000.
@item rtmp_conn
Extra arbitrary AMF connection parameters, parsed from a string,
e.g. like @code{B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0}.
Each value is prefixed by a single character denoting the type,
B for Boolean, N for number, S for string, O for object, or Z for null,
followed by a colon. For Booleans the data must be either 0 or 1 for
FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
1 to end or begin an object, respectively. Data items in subobjects may
be named, by prefixing the type with 'N' and specifying the name before
the value (i.e. @code{NB:myFlag:1}). This option may be used multiple
times to construct arbitrary AMF sequences.
@item rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default
is LNX 9,0,124,2.
@item rtmp_flush_interval
Number of packets flushed in the same request (RTMPT only). The default
is 10.
@item rtmp_live
Specify that the media is a live stream. No resuming or seeking in
live streams is possible. The default value is @code{any}, which means the
subscriber first tries to play the live stream specified in the
playpath. If a live stream of that name is not found, it plays the
recorded stream. The other possible values are @code{live} and
@code{recorded}.
@item rtmp_pageurl
URL of the web page in which the media was embedded. By default no
value will be sent.
@item rtmp_playpath
Stream identifier to play or to publish. This option overrides the
parameter specified in the URI.
@item rtmp_subscribe
Name of live stream to subscribe to. By default no value will be sent.
It is only sent if the option is specified or if rtmp_live
is set to live.
@item rtmp_swfhash
SHA256 hash of the decompressed SWF file (32 bytes).
@item rtmp_swfsize
Size of the decompressed SWF file, required for SWFVerification.
@item rtmp_swfurl
URL of the SWF player for the media. By default no value will be sent.
@item rtmp_swfverify
URL to player swf file, compute hash/size automatically.
@item rtmp_tcurl
URL of the target stream. Defaults to proto://host[:port]/app.
@end table
For example to read with @command{ffplay} a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
@example
ffplay rtmp://myserver/vod/sample
@end example
@section rtmpe
Encrypted Real-Time Messaging Protocol.
The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
streaming multimedia content within standard cryptographic primitives,
consisting of Diffie-Hellman key exchange and HMACSHA256, generating
a pair of RC4 keys.
@section rtmps
Real-Time Messaging Protocol over a secure SSL connection.
The Real-Time Messaging Protocol (RTMPS) is used for streaming
multimedia content across an encrypted connection.
@section rtmpt
Real-Time Messaging Protocol tunneled through HTTP.
The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
for streaming multimedia content within HTTP requests to traverse
firewalls.
@section rtmpte
Encrypted Real-Time Messaging Protocol tunneled through HTTP.
The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE)
is used for streaming multimedia content within HTTP requests to traverse
firewalls.
@section rtmpts
Real-Time Messaging Protocol tunneled through HTTPS.
The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used
for streaming multimedia content within HTTPS requests to traverse
firewalls.
@section rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through
librtmp.
Requires the presence of the librtmp headers and library during
configuration. You need to explicitly configure the build with
"--enable-librtmp". If enabled this will replace the native RTMP
protocol.
This protocol provides most client functions and a few server
functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT),
encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled
variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
@example
@var{rtmp_proto}://@var{server}[:@var{port}][/@var{app}][/@var{playpath}] @var{options}
@end example
where @var{rtmp_proto} is one of the strings "rtmp", "rtmpt", "rtmpe",
"rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
@var{server}, @var{port}, @var{app} and @var{playpath} have the same
meaning as specified for the RTMP native protocol.
@var{options} contains a list of space-separated options of the form
@var{key}=@var{val}.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
@command{ffmpeg}:
@example
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
@end example
To play the same stream using @command{ffplay}:
@example
ffplay "rtmp://myserver/live/mystream live=1"
@end example
@section rtp
Real-Time Protocol.
@section rtsp
RTSP is not technically a protocol handler in libavformat, it is a demuxer
and muxer. The demuxer supports both normal RTSP (with data transferred
over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with
data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server
supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's
@uref{http://github.com/revmischa/rtsp-server, RTSP server}).
The required syntax for a RTSP url is:
@example
rtsp://@var{hostname}[:@var{port}]/@var{path}
@end example
The following options (set on the @command{ffmpeg}/@command{ffplay} command
line, or set in code via @code{AVOption}s or in @code{avformat_open_input}),
are supported:
Flags for @code{rtsp_transport}:
@table @option
@item udp
Use UDP as lower transport protocol.
@item tcp
Use TCP (interleaving within the RTSP control channel) as lower
transport protocol.
@item udp_multicast
Use UDP multicast as lower transport protocol.
@item http
Use HTTP tunneling as lower transport protocol, which is useful for
passing proxies.
@end table
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the @code{tcp} and @code{udp} options are supported.
Flags for @code{rtsp_flags}:
@table @option
@item filter_src
Accept packets only from negotiated peer address and port.
@item listen
Act as a server, listening for an incoming connection.
@end table
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). This
can be disabled by setting the maximum demuxing delay to zero (via
the @code{max_delay} field of AVFormatContext).
When watching multi-bitrate Real-RTSP streams with @command{ffplay}, the
streams to display can be chosen with @code{-vst} @var{n} and
@code{-ast} @var{n} for video and audio respectively, and can be switched
on the fly by pressing @code{v} and @code{a}.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
@example
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
@end example
To watch a stream tunneled over HTTP:
@example
ffplay -rtsp_transport http rtsp://server/video.mp4
@end example
To send a stream in realtime to a RTSP server, for others to watch:
@example
ffmpeg -re -i @var{input} -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
@end example
To receive a stream in realtime:
@example
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp @var{output}
@end example
@section sap
Session Announcement Protocol (RFC 2974). This is not technically a
protocol handler in libavformat, it is a muxer and demuxer.
It is used for signalling of RTP streams, by announcing the SDP for the
streams regularly on a separate port.
@subsection Muxer
The syntax for a SAP url given to the muxer is:
@example
sap://@var{destination}[:@var{port}][?@var{options}]
@end example
The RTP packets are sent to @var{destination} on port @var{port},
or to port 5004 if no port is specified.
@var{options} is a @code{&}-separated list. The following options
are supported:
@table @option
@item announce_addr=@var{address}
Specify the destination IP address for sending the announcements to.
If omitted, the announcements are sent to the commonly used SAP
announcement multicast address 224.2.127.254 (sap.mcast.net), or
ff0e::2:7ffe if @var{destination} is an IPv6 address.
@item announce_port=@var{port}
Specify the port to send the announcements on, defaults to
9875 if not specified.
@item ttl=@var{ttl}
Specify the time to live value for the announcements and RTP packets,
defaults to 255.
@item same_port=@var{0|1}
If set to 1, send all RTP streams on the same port pair. If zero (the
default), all streams are sent on unique ports, with each stream on a
port 2 numbers higher than the previous.
VLC/Live555 requires this to be set to 1, to be able to receive the stream.
The RTP stack in libavformat for receiving requires all streams to be sent
on unique ports.
@end table
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255?same_port=1
@end example
Similarly, for watching in @command{ffplay}:
@example
ffmpeg -re -i @var{input} -f sap sap://224.0.0.255
@end example
And for watching in @command{ffplay}, over IPv6:
@example
ffmpeg -re -i @var{input} -f sap sap://[ff0e::1:2:3:4]
@end example
@subsection Demuxer
The syntax for a SAP url given to the demuxer is:
@example
sap://[@var{address}][:@var{port}]
@end example
@var{address} is the multicast address to listen for announcements on,
if omitted, the default 224.2.127.254 (sap.mcast.net) is used. @var{port}
is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port.
Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
@example
ffplay sap://
@end example
To play back the first stream announced on one the default IPv6 SAP multicast address:
@example
ffplay sap://[ff0e::2:7ffe]
@end example
@section tcp
Trasmission Control Protocol.
The required syntax for a TCP url is:
@example
tcp://@var{hostname}:@var{port}[?@var{options}]
@end example
@table @option
@item listen
Listen for an incoming connection
@item timeout=@var{microseconds}
In read mode: if no data arrived in more than this time interval, raise error.
In write mode: if socket cannot be written in more than this time interval, raise error.
This also sets timeout on TCP connection establishing.
@example
ffmpeg -i @var{input} -f @var{format} tcp://@var{hostname}:@var{port}?listen
ffplay tcp://@var{hostname}:@var{port}
@end example
@end table
@section tls
Transport Layer Security/Secure Sockets Layer
The required syntax for a TLS/SSL url is:
@example
tls://@var{hostname}:@var{port}[?@var{options}]
@end example
@table @option
@item listen
Act as a server, listening for an incoming connection.
@item cafile=@var{filename}
Certificate authority file. The file must be in OpenSSL PEM format.
@item cert=@var{filename}
Certificate file. The file must be in OpenSSL PEM format.
@item key=@var{filename}
Private key file.
@item verify=@var{0|1}
Verify the peer's certificate.
@end table
Example command lines:
To create a TLS/SSL server that serves an input stream.
@example
ffmpeg -i @var{input} -f @var{format} tls://@var{hostname}:@var{port}?listen&cert=@var{server.crt}&key=@var{server.key}
@end example
To play back a stream from the TLS/SSL server using @command{ffplay}:
@example
ffplay tls://@var{hostname}:@var{port}
@end example
@section udp
User Datagram Protocol.
The required syntax for a UDP url is:
@example
udp://@var{hostname}:@var{port}[?@var{options}]
@end example
@var{options} contains a list of &-separated options of the form @var{key}=@var{val}.
In case threading is enabled on the system, a circular buffer is used
to store the incoming data, which allows to reduce loss of data due to
UDP socket buffer overruns. The @var{fifo_size} and
@var{overrun_nonfatal} options are related to this buffer.
The list of supported options follows.
@table @option
@item buffer_size=@var{size}
Set the UDP socket buffer size in bytes. This is used both for the
receiving and the sending buffer size.
@item localport=@var{port}
Override the local UDP port to bind with.
@item localaddr=@var{addr}
Choose the local IP address. This is useful e.g. if sending multicast
and the host has multiple interfaces, where the user can choose
which interface to send on by specifying the IP address of that interface.
@item pkt_size=@var{size}
Set the size in bytes of UDP packets.
@item reuse=@var{1|0}
Explicitly allow or disallow reusing UDP sockets.
@item ttl=@var{ttl}
Set the time to live value (for multicast only).
@item connect=@var{1|0}
Initialize the UDP socket with @code{connect()}. In this case, the
destination address can't be changed with ff_udp_set_remote_url later.
If the destination address isn't known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
@item sources=@var{address}[,@var{address}]
Only receive packets sent to the multicast group from one of the
specified sender IP addresses.
@item block=@var{address}[,@var{address}]
Ignore packets sent to the multicast group from the specified
sender IP addresses.
@item fifo_size=@var{units}
Set the UDP receiving circular buffer size, expressed as a number of
packets with size of 188 bytes. If not specified defaults to 7*4096.
@item overrun_nonfatal=@var{1|0}
Survive in case of UDP receiving circular buffer overrun. Default
value is 0.
@item timeout=@var{microseconds}
In read mode: if no data arrived in more than this time interval, raise error.
@end table
Some usage examples of the UDP protocol with @command{ffmpeg} follow.
To stream over UDP to a remote endpoint:
@example
ffmpeg -i @var{input} -f @var{format} udp://@var{hostname}:@var{port}
@end example
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
@example
ffmpeg -i @var{input} -f mpegts udp://@var{hostname}:@var{port}?pkt_size=188&buffer_size=65535
@end example
To receive over UDP from a remote endpoint:
@example
ffmpeg -i udp://[@var{multicast-address}]:@var{port}
@end example
@c man end PROTOCOLS

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A Quick Description Of Rate Distortion Theory.
We want to encode a video, picture or piece of music optimally. What does
"optimally" really mean? It means that we want to get the best quality at a
given filesize OR we want to get the smallest filesize at a given quality
(in practice, these 2 goals are usually the same).
Solving this directly is not practical; trying all byte sequences 1
megabyte in length and selecting the "best looking" sequence will yield
256^1000000 cases to try.
But first, a word about quality, which is also called distortion.
Distortion can be quantified by almost any quality measurement one chooses.
Commonly, the sum of squared differences is used but more complex methods
that consider psychovisual effects can be used as well. It makes no
difference in this discussion.
First step: that rate distortion factor called lambda...
Let's consider the problem of minimizing:
distortion + lambda*rate
rate is the filesize
distortion is the quality
lambda is a fixed value chosen as a tradeoff between quality and filesize
Is this equivalent to finding the best quality for a given max
filesize? The answer is yes. For each filesize limit there is some lambda
factor for which minimizing above will get you the best quality (using your
chosen quality measurement) at the desired (or lower) filesize.
Second step: splitting the problem.
Directly splitting the problem of finding the best quality at a given
filesize is hard because we do not know how many bits from the total
filesize should be allocated to each of the subproblems. But the formula
from above:
distortion + lambda*rate
can be trivially split. Consider:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
This creates a problem made of 2 independent subproblems. The subproblems
might be 2 16x16 macroblocks in a frame of 32x16 size. To minimize:
(distortion0 + distortion1) + lambda*(rate0 + rate1)
we just have to minimize:
distortion0 + lambda*rate0
and
distortion1 + lambda*rate1
I.e, the 2 problems can be solved independently.
Author: Michael Niedermayer
Copyright: LGPL

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=============================================
Snow Video Codec Specification Draft 20080110
=============================================
Introduction:
=============
This specification describes the Snow bitstream syntax and semantics as
well as the formal Snow decoding process.
The decoding process is described precisely and any compliant decoder
MUST produce the exact same output for a spec-conformant Snow stream.
For encoding, though, any process which generates a stream compliant to
the syntactical and semantic requirements and which is decodable by
the process described in this spec shall be considered a conformant
Snow encoder.
Definitions:
============
MUST the specific part must be done to conform to this standard
SHOULD it is recommended to be done that way, but not strictly required
ilog2(x) is the rounded down logarithm of x with basis 2
ilog2(0) = 0
Type definitions:
=================
b 1-bit range coded
u unsigned scalar value range coded
s signed scalar value range coded
Bitstream syntax:
=================
frame:
header
prediction
residual
header:
keyframe b MID_STATE
if(keyframe || always_reset)
reset_contexts
if(keyframe){
version u header_state
always_reset b header_state
temporal_decomposition_type u header_state
temporal_decomposition_count u header_state
spatial_decomposition_count u header_state
colorspace_type u header_state
chroma_h_shift u header_state
chroma_v_shift u header_state
spatial_scalability b header_state
max_ref_frames-1 u header_state
qlogs
}
if(!keyframe){
update_mc b header_state
if(update_mc){
for(plane=0; plane<2; plane++){
diag_mc b header_state
htaps/2-1 u header_state
for(i= p->htaps/2; i; i--)
|hcoeff[i]| u header_state
}
}
update_qlogs b header_state
if(update_qlogs){
spatial_decomposition_count u header_state
qlogs
}
}
spatial_decomposition_type s header_state
qlog s header_state
mv_scale s header_state
qbias s header_state
block_max_depth s header_state
qlogs:
for(plane=0; plane<2; plane++){
quant_table[plane][0][0] s header_state
for(level=0; level < spatial_decomposition_count; level++){
quant_table[plane][level][1]s header_state
quant_table[plane][level][3]s header_state
}
}
reset_contexts
*_state[*]= MID_STATE
prediction:
for(y=0; y<block_count_vertical; y++)
for(x=0; x<block_count_horizontal; x++)
block(0)
block(level):
mvx_diff=mvy_diff=y_diff=cb_diff=cr_diff=0
if(keyframe){
intra=1
}else{
if(level!=max_block_depth){
s_context= 2*left->level + 2*top->level + topleft->level + topright->level
leaf b block_state[4 + s_context]
}
if(level==max_block_depth || leaf){
intra b block_state[1 + left->intra + top->intra]
if(intra){
y_diff s block_state[32]
cb_diff s block_state[64]
cr_diff s block_state[96]
}else{
ref_context= ilog2(2*left->ref) + ilog2(2*top->ref)
if(ref_frames > 1)
ref u block_state[128 + 1024 + 32*ref_context]
mx_context= ilog2(2*abs(left->mx - top->mx))
my_context= ilog2(2*abs(left->my - top->my))
mvx_diff s block_state[128 + 32*(mx_context + 16*!!ref)]
mvy_diff s block_state[128 + 32*(my_context + 16*!!ref)]
}
}else{
block(level+1)
block(level+1)
block(level+1)
block(level+1)
}
}
residual:
residual2(luma)
residual2(chroma_cr)
residual2(chroma_cb)
residual2:
for(level=0; level<spatial_decomposition_count; level++){
if(level==0)
subband(LL, 0)
subband(HL, level)
subband(LH, level)
subband(HH, level)
}
subband:
FIXME
Tag description:
----------------
version
0
this MUST NOT change within a bitstream
always_reset
if 1 then the range coder contexts will be reset after each frame
temporal_decomposition_type
0
temporal_decomposition_count
0
spatial_decomposition_count
FIXME
colorspace_type
0
this MUST NOT change within a bitstream
chroma_h_shift
log2(luma.width / chroma.width)
this MUST NOT change within a bitstream
chroma_v_shift
log2(luma.height / chroma.height)
this MUST NOT change within a bitstream
spatial_scalability
0
max_ref_frames
maximum number of reference frames
this MUST NOT change within a bitstream
update_mc
indicates that motion compensation filter parameters are stored in the
header
diag_mc
flag to enable faster diagonal interpolation
this SHOULD be 1 unless it turns out to be covered by a valid patent
htaps
number of half pel interpolation filter taps, MUST be even, >0 and <10
hcoeff
half pel interpolation filter coefficients, hcoeff[0] are the 2 middle
coefficients [1] are the next outer ones and so on, resulting in a filter
like: ...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ...
the sign of the coefficients is not explicitly stored but alternates
after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,...
hcoeff[0] is not explicitly stored but found by subtracting the sum
of all stored coefficients with signs from 32
hcoeff[0]= 32 - hcoeff[1] - hcoeff[2] - ...
a good choice for hcoeff and htaps is
htaps= 6
hcoeff={40,-10,2}
an alternative which requires more computations at both encoder and
decoder side and may or may not be better is
htaps= 8
hcoeff={42,-14,6,-2}
ref_frames
minimum of the number of available reference frames and max_ref_frames
for example the first frame after a key frame always has ref_frames=1
spatial_decomposition_type
wavelet type
0 is a 9/7 symmetric compact integer wavelet
1 is a 5/3 symmetric compact integer wavelet
others are reserved
stored as delta from last, last is reset to 0 if always_reset || keyframe
qlog
quality (logarthmic quantizer scale)
stored as delta from last, last is reset to 0 if always_reset || keyframe
mv_scale
stored as delta from last, last is reset to 0 if always_reset || keyframe
FIXME check that everything works fine if this changes between frames
qbias
dequantization bias
stored as delta from last, last is reset to 0 if always_reset || keyframe
block_max_depth
maximum depth of the block tree
stored as delta from last, last is reset to 0 if always_reset || keyframe
quant_table
quantiztation table
Highlevel bitstream structure:
=============================
--------------------------------------------
| Header |
--------------------------------------------
| ------------------------------------ |
| | Block0 | |
| | split? | |
| | yes no | |
| | ......... intra? | |
| | : Block01 : yes no | |
| | : Block02 : ....... .......... | |
| | : Block03 : : y DC : : ref index: | |
| | : Block04 : : cb DC : : motion x : | |
| | ......... : cr DC : : motion y : | |
| | ....... .......... | |
| ------------------------------------ |
| ------------------------------------ |
| | Block1 | |
| ... |
--------------------------------------------
| ------------ ------------ ------------ |
|| Y subbands | | Cb subbands| | Cr subbands||
|| --- --- | | --- --- | | --- --- ||
|| |LL0||HL0| | | |LL0||HL0| | | |LL0||HL0| ||
|| --- --- | | --- --- | | --- --- ||
|| --- --- | | --- --- | | --- --- ||
|| |LH0||HH0| | | |LH0||HH0| | | |LH0||HH0| ||
|| --- --- | | --- --- | | --- --- ||
|| --- --- | | --- --- | | --- --- ||
|| |HL1||LH1| | | |HL1||LH1| | | |HL1||LH1| ||
|| --- --- | | --- --- | | --- --- ||
|| --- --- | | --- --- | | --- --- ||
|| |HH1||HL2| | | |HH1||HL2| | | |HH1||HL2| ||
|| ... | | ... | | ... ||
| ------------ ------------ ------------ |
--------------------------------------------
Decoding process:
=================
------------
| |
| Subbands |
------------ | |
| | ------------
| Intra DC | |
| | LL0 subband prediction
------------ |
\ Dequantizaton
------------------- \ |
| Reference frames | \ IDWT
| ------- ------- | Motion \ |
||Frame 0| |Frame 1|| Compensation . OBMC v -------
| ------- ------- | --------------. \------> + --->|Frame n|-->output
| ------- ------- | -------
||Frame 2| |Frame 3||<----------------------------------/
| ... |
-------------------
Range Coder:
============
Binary Range Coder:
-------------------
The implemented range coder is an adapted version based upon "Range encoding:
an algorithm for removing redundancy from a digitised message." by G. N. N.
Martin.
The symbols encoded by the Snow range coder are bits (0|1). The
associated probabilities are not fix but change depending on the symbol mix
seen so far.
bit seen | new state
---------+-----------------------------------------------
0 | 256 - state_transition_table[256 - old_state];
1 | state_transition_table[ old_state];
state_transition_table = {
0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27,
28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42,
43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57,
58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73,
74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88,
89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103,
104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118,
119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133,
134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179,
180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194,
195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209,
210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225,
226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240,
241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};
FIXME
Range Coding of integers:
-------------------------
FIXME
Neighboring Blocks:
===================
left and top are set to the respective blocks unless they are outside of
the image in which case they are set to the Null block
top-left is set to the top left block unless it is outside of the image in
which case it is set to the left block
if this block has no larger parent block or it is at the left side of its
parent block and the top right block is not outside of the image then the
top right block is used for top-right else the top-left block is used
Null block
y,cb,cr are 128
level, ref, mx and my are 0
Motion Vector Prediction:
=========================
1. the motion vectors of all the neighboring blocks are scaled to
compensate for the difference of reference frames
scaled_mv= (mv * (256 * (current_reference+1) / (mv.reference+1)) + 128)>>8
2. the median of the scaled left, top and top-right vectors is used as
motion vector prediction
3. the used motion vector is the sum of the predictor and
(mvx_diff, mvy_diff)*mv_scale
Intra DC Predicton:
======================
the luma and chroma values of the left block are used as predictors
the used luma and chroma is the sum of the predictor and y_diff, cb_diff, cr_diff
to reverse this in the decoder apply the following:
block[y][x].dc[0] = block[y][x-1].dc[0] + y_diff;
block[y][x].dc[1] = block[y][x-1].dc[1] + cb_diff;
block[y][x].dc[2] = block[y][x-1].dc[2] + cr_diff;
block[*][-1].dc[*]= 128;
Motion Compensation:
====================
Halfpel interpolation:
----------------------
halfpel interpolation is done by convolution with the halfpel filter stored
in the header:
horizontal halfpel samples are found by
H1[y][x] = hcoeff[0]*(F[y][x ] + F[y][x+1])
+ hcoeff[1]*(F[y][x-1] + F[y][x+2])
+ hcoeff[2]*(F[y][x-2] + F[y][x+3])
+ ...
h1[y][x] = (H1[y][x] + 32)>>6;
vertical halfpel samples are found by
H2[y][x] = hcoeff[0]*(F[y ][x] + F[y+1][x])
+ hcoeff[1]*(F[y-1][x] + F[y+2][x])
+ ...
h2[y][x] = (H2[y][x] + 32)>>6;
vertical+horizontal halfpel samples are found by
H3[y][x] = hcoeff[0]*(H2[y][x ] + H2[y][x+1])
+ hcoeff[1]*(H2[y][x-1] + H2[y][x+2])
+ ...
H3[y][x] = hcoeff[0]*(H1[y ][x] + H1[y+1][x])
+ hcoeff[1]*(H1[y+1][x] + H1[y+2][x])
+ ...
h3[y][x] = (H3[y][x] + 2048)>>12;
F H1 F
| | |
| | |
| | |
F H1 F
| | |
| | |
| | |
F-------F-------F-> H1<-F-------F-------F
v v v
H2 H3 H2
^ ^ ^
F-------F-------F-> H1<-F-------F-------F
| | |
| | |
| | |
F H1 F
| | |
| | |
| | |
F H1 F
unavailable fullpel samples (outside the picture for example) shall be equal
to the closest available fullpel sample
Smaller pel interpolation:
--------------------------
if diag_mc is set then points which lie on a line between 2 vertically,
horiziontally or diagonally adjacent halfpel points shall be interpolated
linearls with rounding to nearest and halfway values rounded up.
points which lie on 2 diagonals at the same time should only use the one
diagonal not containing the fullpel point
F-->O---q---O<--h1->O---q---O<--F
v \ / v \ / v
O O O O O O O
| / | \ |
q q q q q
| / | \ |
O O O O O O O
^ / \ ^ / \ ^
h2-->O---q---O<--h3->O---q---O<--h2
v \ / v \ / v
O O O O O O O
| \ | / |
q q q q q
| \ | / |
O O O O O O O
^ / \ ^ / \ ^
F-->O---q---O<--h1->O---q---O<--F
the remaining points shall be bilinearly interpolated from the
up to 4 surrounding halfpel and fullpel points, again rounding should be to
nearest and halfway values rounded up
compliant Snow decoders MUST support 1-1/8 pel luma and 1/2-1/16 pel chroma
interpolation at least
Overlapped block motion compensation:
-------------------------------------
FIXME
LL band prediction:
===================
Each sample in the LL0 subband is predicted by the median of the left, top and
left+top-topleft samples, samples outside the subband shall be considered to
be 0. To reverse this prediction in the decoder apply the following.
for(y=0; y<height; y++){
for(x=0; x<width; x++){
sample[y][x] += median(sample[y-1][x],
sample[y][x-1],
sample[y-1][x]+sample[y][x-1]-sample[y-1][x-1]);
}
}
sample[-1][*]=sample[*][-1]= 0;
width,height here are the width and height of the LL0 subband not of the final
video
Dequantizaton:
==============
FIXME
Wavelet Transform:
==================
Snow supports 2 wavelet transforms, the symmetric biorthogonal 5/3 integer
transform and a integer approximation of the symmetric biorthogonal 9/7
daubechies wavelet.
2D IDWT (inverse discrete wavelet transform)
--------------------------------------------
The 2D IDWT applies a 2D filter recursively, each time combining the
4 lowest frequency subbands into a single subband until only 1 subband
remains.
The 2D filter is done by first applying a 1D filter in the vertical direction
and then applying it in the horizontal one.
--------------- --------------- --------------- ---------------
|LL0|HL0| | | | | | | | | | | |
|---+---| HL1 | | L0|H0 | HL1 | | LL1 | HL1 | | | |
|LH0|HH0| | | | | | | | | | | |
|-------+-------|->|-------+-------|->|-------+-------|->| L1 | H1 |->...
| | | | | | | | | | | |
| LH1 | HH1 | | LH1 | HH1 | | LH1 | HH1 | | | |
| | | | | | | | | | | |
--------------- --------------- --------------- ---------------
1D Filter:
----------
1. interleave the samples of the low and high frequency subbands like
s={L0, H0, L1, H1, L2, H2, L3, H3, ... }
note, this can end with a L or a H, the number of elements shall be w
s[-1] shall be considered equivalent to s[1 ]
s[w ] shall be considered equivalent to s[w-2]
2. perform the lifting steps in order as described below
5/3 Integer filter:
1. s[i] -= (s[i-1] + s[i+1] + 2)>>2; for all even i < w
2. s[i] += (s[i-1] + s[i+1] )>>1; for all odd i < w
\ | /|\ | /|\ | /|\ | /|\
\|/ | \|/ | \|/ | \|/ |
+ | + | + | + | -1/4
/|\ | /|\ | /|\ | /|\ |
/ | \|/ | \|/ | \|/ | \|/
| + | + | + | + +1/2
Snow's 9/7 Integer filter:
1. s[i] -= (3*(s[i-1] + s[i+1]) + 4)>>3; for all even i < w
2. s[i] -= s[i-1] + s[i+1] ; for all odd i < w
3. s[i] += ( s[i-1] + s[i+1] + 4*s[i] + 8)>>4; for all even i < w
4. s[i] += (3*(s[i-1] + s[i+1]) )>>1; for all odd i < w
\ | /|\ | /|\ | /|\ | /|\
\|/ | \|/ | \|/ | \|/ |
+ | + | + | + | -3/8
/|\ | /|\ | /|\ | /|\ |
/ | \|/ | \|/ | \|/ | \|/
(| + (| + (| + (| + -1
\ + /|\ + /|\ + /|\ + /|\ +1/4
\|/ | \|/ | \|/ | \|/ |
+ | + | + | + | +1/16
/|\ | /|\ | /|\ | /|\ |
/ | \|/ | \|/ | \|/ | \|/
| + | + | + | + +3/2
optimization tips:
following are exactly identical
(3a)>>1 == a + (a>>1)
(a + 4b + 8)>>4 == ((a>>2) + b + 2)>>2
16bit implementation note:
The IDWT can be implemented with 16bits, but this requires some care to
prevent overflows, the following list, lists the minimum number of bits needed
for some terms
1. lifting step
A= s[i-1] + s[i+1] 16bit
3*A + 4 18bit
A + (A>>1) + 2 17bit
3. lifting step
s[i-1] + s[i+1] 17bit
4. lifiting step
3*(s[i-1] + s[i+1]) 17bit
TODO:
=====
Important:
finetune initial contexts
flip wavelet?
try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients
try the MV length as context for coding the residual coefficients
use extradata for stuff which is in the keyframes now?
the MV median predictor is patented IIRC
implement per picture halfpel interpolation
try different range coder state transition tables for different contexts
Not Important:
compare the 6 tap and 8 tap hpel filters (psnr/bitrate and subjective quality)
spatial_scalability b vs u (!= 0 breaks syntax anyway so we can add a u later)
Credits:
========
Michael Niedermayer
Loren Merritt
Copyright:
==========
GPL + GFDL + whatever is needed to make this a RFC

24
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Google Summer of Code and similar project guidelines
Summer of Code is a project by Google in which students are paid to implement
some nice new features for various participating open source projects ...
This text is a collection of things to take care of for the next soc as
it's a little late for this year's soc (2006).
The Goal:
Our goal in respect to soc is and must be of course exactly one thing and
that is to improve FFmpeg, to reach this goal, code must
* conform to the development policy and patch submission guidelines
* must improve FFmpeg somehow (faster, smaller, "better",
more codecs supported, fewer bugs, cleaner, ...)
for mentors and other developers to help students to reach that goal it is
essential that changes to their codebase are publicly visible, clean and
easy reviewable that again leads us to:
* use of a revision control system like git
* separation of cosmetic from non-cosmetic changes (this is almost entirely
ignored by mentors and students in soc 2006 which might lead to a surprise
when the code will be reviewed at the end before a possible inclusion in
FFmpeg, individual changes were generally not reviewable due to cosmetics).
* frequent commits, so that comments can be provided early

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The official guide to swresample for confused developers.
=========================================================
Current (simplified) Architecture:
---------------------------------
Input
v
__________________/|\___________
/ | \
/ input sample format convert v
/ | ___________/
| |/
| v
| ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
v \|/
Special Converter v
v ___________/|\___________ _____________
| / | \ | |
| Rematrix | resample <---->| Buffers |
| \___________ | ___________/ |_____________|
| \|/
| v
| |\___________
\ | \
\ output sample format convert v
\_________________ | ___________/
\|/
v
Output
Planar/Packed conversion is done when needed during sample format conversion.
Every step can be skipped without memcpy when its not needed.
Either Resampling and Rematrixing can be performed first depending on which
way its faster.
The Buffers are needed for resampling due to resamplng being a process that
requires future and past data, it thus also introduces inevitably a delay when
used.
Internally 32bit float and 16bit int is supported currently, other formats can
easily be added.
Externally all sample formats in packed and planar configuration are supported
It's also trivial to add special converters for common cases.
If only sample format and/or packed/planar conversion is needed, it
is performed from input to output directly in a single pass with no intermediates.

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The official guide to swscale for confused developers.
========================================================
Current (simplified) Architecture:
---------------------------------
Input
v
_______OR_________
/ \
/ \
special converter [Input to YUV converter]
| |
| (8bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 )
| |
| v
| Horizontal scaler
| |
| (15bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 )
| |
| v
| Vertical scaler and output converter
| |
v v
output
Swscale has 2 scaler paths. Each side must be capable of handling
slices, that is, consecutive non-overlapping rectangles of dimension
(0,slice_top) - (picture_width, slice_bottom).
special converter
These generally are unscaled converters of common
formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also
in principle contain scalers optimized for specific common cases.
Main path
The main path is used when no special converter can be used. The code
is designed as a destination line pull architecture. That is, for each
output line the vertical scaler pulls lines from a ring buffer. When
the ring buffer does not contain the wanted line, then it is pulled from
the input slice through the input converter and horizontal scaler.
The result is also stored in the ring buffer to serve future vertical
scaler requests.
When no more output can be generated because lines from a future slice
would be needed, then all remaining lines in the current slice are
converted, horizontally scaled and put in the ring buffer.
[This is done for luma and chroma, each with possibly different numbers
of lines per picture.]
Input to YUV Converter
When the input to the main path is not planar 8 bits per component YUV or
8-bit gray, it is converted to planar 8-bit YUV. Two sets of converters
exist for this currently: One performs horizontal downscaling by 2
before the conversion, the other leaves the full chroma resolution,
but is slightly slower. The scaler will try to preserve full chroma
when the output uses it. It is possible to force full chroma with
SWS_FULL_CHR_H_INP even for cases where the scaler thinks it is useless.
Horizontal scaler
There are several horizontal scalers. A special case worth mentioning is
the fast bilinear scaler that is made of runtime-generated MMXEXT code
using specially tuned pshufw instructions.
The remaining scalers are specially-tuned for various filter lengths.
They scale 8-bit unsigned planar data to 16-bit signed planar data.
Future >8 bits per component inputs will need to add a new horizontal
scaler that preserves the input precision.
Vertical scaler and output converter
There is a large number of combined vertical scalers + output converters.
Some are:
* unscaled output converters
* unscaled output converters that average 2 chroma lines
* bilinear converters (C, MMX and accurate MMX)
* arbitrary filter length converters (C, MMX and accurate MMX)
And
* Plain C 8-bit 4:2:2 YUV -> RGB converters using LUTs
* Plain C 17-bit 4:4:4 YUV -> RGB converters using multiplies
* MMX 11-bit 4:2:2 YUV -> RGB converters
* Plain C 16-bit Y -> 16-bit gray
...
RGB with less than 8 bits per component uses dither to improve the
subjective quality and low-frequency accuracy.
Filter coefficients:
--------------------
There are several different scalers (bilinear, bicubic, lanczos, area,
sinc, ...). Their coefficients are calculated in initFilter().
Horizontal filter coefficients have a 1.0 point at 1 << 14, vertical ones at
1 << 12. The 1.0 points have been chosen to maximize precision while leaving
a little headroom for convolutional filters like sharpening filters and
minimizing SIMD instructions needed to apply them.
It would be trivial to use a different 1.0 point if some specific scaler
would benefit from it.
Also, as already hinted at, initFilter() accepts an optional convolutional
filter as input that can be used for contrast, saturation, blur, sharpening
shift, chroma vs. luma shift, ...

230
project/jni/ffmpeg/doc/syntax.texi Normal file
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@chapter Syntax
@c man begin SYNTAX
This section documents the syntax and formats employed by the FFmpeg
libraries and tools.
@anchor{quoting_and_escaping}
@section Quoting and escaping
FFmpeg adopts the following quoting and escaping mechanism, unless
explicitly specified. The following rules are applied:
@itemize
@item
@code{'} and @code{\} are special characters (respectively used for
quoting and escaping). In addition to them, there might be other
special characters depending on the specific syntax where the escaping
and quoting are employed.
@item
A special character is escaped by prefixing it with a '\'.
@item
All characters enclosed between '' are included literally in the
parsed string. The quote character @code{'} itself cannot be quoted,
so you may need to close the quote and escape it.
@item
Leading and trailing whitespaces, unless escaped or quoted, are
removed from the parsed string.
@end itemize
Note that you may need to add a second level of escaping when using
the command line or a script, which depends on the syntax of the
adopted shell language.
The function @code{av_get_token} defined in
@file{libavutil/avstring.h} can be used to parse a token quoted or
escaped according to the rules defined above.
The tool @file{tools/ffescape} in the FFmpeg source tree can be used
to automatically quote or escape a string in a script.
@subsection Examples
@itemize
@item
Escape the string @code{Crime d'Amour} containing the @code{'} special
character:
@example
Crime d\'Amour
@end example
@item
The string above contains a quote, so the @code{'} needs to be escaped
when quoting it:
@example
'Crime d'\''Amour'
@end example
@item
Include leading or trailing whitespaces using quoting:
@example
' this string starts and ends with whitespaces '
@end example
@item
Escaping and quoting can be mixed together:
@example
' The string '\'string\'' is a string '
@end example
@item
To include a literal @code{\} you can use either escaping or quoting:
@example
'c:\foo' can be written as c:\\foo
@end example
@end itemize
@anchor{date syntax}
@section Date
The accepted syntax is:
@example
[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
now
@end example
If the value is "now" it takes the current time.
Time is local time unless Z is appended, in which case it is
interpreted as UTC.
If the year-month-day part is not specified it takes the current
year-month-day.
@anchor{time duration syntax}
@section Time duration
The accepted syntax is:
@example
[-]HH:MM:SS[.m...]
[-]S+[.m...]
@end example
@var{HH} expresses the number of hours, @var{MM} the number a of minutes
and @var{SS} the number of seconds.
@anchor{video size syntax}
@section Video size
Specify the size of the sourced video, it may be a string of the form
@var{width}x@var{height}, or the name of a size abbreviation.
The following abbreviations are recognized:
@table @samp
@item sqcif
128x96
@item qcif
176x144
@item cif
352x288
@item 4cif
704x576
@item 16cif
1408x1152
@item qqvga
160x120
@item qvga
320x240
@item vga
640x480
@item svga
800x600
@item xga
1024x768
@item uxga
1600x1200
@item qxga
2048x1536
@item sxga
1280x1024
@item qsxga
2560x2048
@item hsxga
5120x4096
@item wvga
852x480
@item wxga
1366x768
@item wsxga
1600x1024
@item wuxga
1920x1200
@item woxga
2560x1600
@item wqsxga
3200x2048
@item wquxga
3840x2400
@item whsxga
6400x4096
@item whuxga
7680x4800
@item cga
320x200
@item ega
640x350
@item hd480
852x480
@item hd720
1280x720
@item hd1080
1920x1080
@end table
@anchor{video rate syntax}
@section Video rate
Specify the frame rate of a video, expressed as the number of frames
generated per second. It has to be a string in the format
@var{frame_rate_num}/@var{frame_rate_den}, an integer number, a float
number or a valid video frame rate abbreviation.
The following abbreviations are recognized:
@table @samp
@item ntsc
30000/1001
@item pal
25/1
@item qntsc
30000/1
@item qpal
25/1
@item sntsc
30000/1
@item spal
25/1
@item film
24/1
@item ntsc-film
24000/1
@end table
@anchor{ratio syntax}
@section Ratio
A ratio can be expressed as an expression, or in the form
@var{numerator}:@var{denominator}.
Note that a ratio with infinite (1/0) or negative value is
considered valid, so you should check on the returned value if you
want to exclude those values.
The undefined value can be expressed using the "0:0" string.
@anchor{color syntax}
@section Color
It can be the name of a color (case insensitive match) or a
[0x|#]RRGGBB[AA] sequence, possibly followed by "@@" and a string
representing the alpha component.
The alpha component may be a string composed by "0x" followed by an
hexadecimal number or a decimal number between 0.0 and 1.0, which
represents the opacity value (0x00/0.0 means completely transparent,
0xff/1.0 completely opaque).
If the alpha component is not specified then 0xff is assumed.
The string "random" will result in a random color.
@c man end SYNTAX

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# no horiz rules between sections
$end_section = \&FFmpeg_end_section;
sub FFmpeg_end_section($$)
{
}
$EXTRA_HEAD =
'<link rel="icon" href="favicon.png" type="image/png" />
';
$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT;
<link rel="stylesheet" type="text/css" href="default.css" />
EOT
my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
<link rel="icon" href="favicon.png" type="image/png" />
</head>
<body>
<div id="container">
EOT
$PRE_BODY_CLOSE = '</div></div>';
$SMALL_RULE = '';
$BODYTEXT = '';
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
my $program_string = defined &T2H_DEFAULT_program_string ?
T2H_DEFAULT_program_string() : program_string();
print $fh '<footer class="footer pagination-right">' . "\n";
print $fh '<span class="label label-info">' . $program_string;
print $fh "</span></footer></div>\n";
}
$float = \&FFmpeg_float;
sub FFmpeg_float($$$$)
{
my $text = shift;
my $float = shift;
my $caption = shift;
my $shortcaption = shift;
my $label = '';
if (exists($float->{'id'}))
{
$label = &$anchor($float->{'id'});
}
my $class = '';
my $subject = '';
if ($caption =~ /NOTE/)
{
$class = "alert alert-info";
}
elsif ($caption =~ /IMPORTANT/)
{
$class = "alert alert-warning";
}
return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>';
}
$print_page_head = \&FFmpeg_print_page_head;
sub FFmpeg_print_page_head($$)
{
my $fh = shift;
my $longtitle = "$Texi2HTML::THISDOC{'title_no_texi'}";
$longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
my $description = $DOCUMENT_DESCRIPTION;
$description = $longtitle if (!defined($description));
$description = "<meta name=\"description\" content=\"$description\">" if
($description ne '');
$description = $Texi2HTML::THISDOC{'documentdescription'} if (defined($Texi2HTML::THISDOC{'documentdescription'}));
my $encoding = '';
$encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne ''));
$longtitle =~ s/Documentation.*//g;
$longtitle = "FFmpeg documentation : " . $longtitle;
print $fh <<EOT;
<!DOCTYPE html>
<html>
$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
<!--
$Texi2HTML::THISDOC{program_authors}
-->
<head>
<title>$longtitle</title>
$description
<meta name="keywords" content="$longtitle">
<meta name="resource-type" content="document">
<meta name="distribution" content="global">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
$encoding
$CSS_LINES
$TEMPLATE_HEADER
EOT
}
# declare encoding in header
$IN_ENCODING = $ENCODING = "utf-8";
# no navigation elements
$SECTION_NAVIGATION = 0;
# the same for texi2html 5.0
$HEADERS = 0;
# TOC and Chapter headings link
$TOC_LINKS = 1;
# print the TOC where @contents is used
$INLINE_CONTENTS = 1;

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Writing a table generator
This documentation is preliminary.
Parts of the API are not good and should be changed.
Basic concepts
A table generator consists of two files, *_tablegen.c and *_tablegen.h.
The .h file will provide the variable declarations and initialization
code for the tables, the .c calls the initialization code and then prints
the tables as a header file using the tableprint.h helpers.
Both of these files will be compiled for the host system, so to avoid
breakage with cross-compilation neither of them may include, directly
or indirectly, config.h or avconfig.h.
This means that e.g. libavutil/mathematics.h is ok but libavutil/libm.h is not.
Due to this, the .c file or Makefile may have to provide additional defines
or stubs, though if possible this should be avoided.
In particular, CONFIG_HARDCODED_TABLES should always be defined to 0.
The .c file
This file should include the *_tablegen.h and tableprint.h files and
anything else it needs as long as it does not depend on config.h or
avconfig.h.
In addition to that it must contain a main() function which initializes
all tables by calling the init functions from the .h file and then prints
them.
The printing code typically looks like this:
write_fileheader();
printf("static const uint8_t my_array[100] = {\n");
write_uint8_t_array(my_array, 100);
printf("};\n");
This is the more generic form, in case you need to do something special.
Usually you should instead use the short form:
write_fileheader();
WRITE_ARRAY("static const", uint8_t, my_array);
write_fileheader() adds some minor things like a "this is a generated file"
comment and some standard includes.
tablegen.h defines some write functions for one- and two-dimensional arrays
for standard types - they print only the "core" parts so they are easier
to reuse for multi-dimensional arrays so the outermost {} must be printed
separately.
If there's no standard function for printing the type you need, the
WRITE_1D_FUNC_ARGV macro is a very quick way to create one.
See libavcodec/dv_tablegen.c for an example.
The .h file
This file should contain:
- one or more initialization functions
- the table variable declarations
If CONFIG_HARDCODED_TABLES is set, the initialization functions should
not do anything, and instead of the variable declarations the
generated *_tables.h file should be included.
Since that will be generated in the build directory, the path must be
included, i.e.
#include "libavcodec/example_tables.h"
not
#include "example_tables.h"
Makefile changes
To make the automatic table creation work, you must manually declare the
new dependency.
For this add a line similar to this:
$(SUBDIR)example.o: $(SUBDIR)example_tables.h
under the "ifdef CONFIG_HARDCODED_TABLES" section in the Makefile.

436
project/jni/ffmpeg/doc/texi2pod.pl Executable file
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#! /usr/bin/perl
# Copyright (C) 1999, 2000, 2001 Free Software Foundation, Inc.
# This file is part of GNU CC.
# GNU CC is free software; you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation; either version 2, or (at your option)
# any later version.
# GNU CC is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
# You should have received a copy of the GNU General Public License
# along with GNU CC; see the file COPYING. If not, write to
# the Free Software Foundation, 51 Franklin Street, Fifth Floor,
# Boston, MA 02110-1301 USA
# This does trivial (and I mean _trivial_) conversion of Texinfo
# markup to Perl POD format. It's intended to be used to extract
# something suitable for a manpage from a Texinfo document.
use warnings;
$output = 0;
$skipping = 0;
%chapters = ();
@chapters_sequence = ();
$chapter = "";
@icstack = ();
@endwstack = ();
@skstack = ();
@instack = ();
$shift = "";
%defs = ();
$fnno = 1;
$inf = "";
@ibase = ();
while ($_ = shift) {
if (/^-D(.*)$/) {
if ($1 ne "") {
$flag = $1;
} else {
$flag = shift;
}
$value = "";
($flag, $value) = ($flag =~ /^([^=]+)(?:=(.+))?/);
die "no flag specified for -D\n"
unless $flag ne "";
die "flags may only contain letters, digits, hyphens, dashes and underscores\n"
unless $flag =~ /^[a-zA-Z0-9_-]+$/;
$defs{$flag} = $value;
} elsif (/^-I(.*)$/) {
push @ibase, $1 ne "" ? $1 : shift;
} elsif (/^-/) {
usage();
} else {
$in = $_, next unless defined $in;
$out = $_, next unless defined $out;
usage();
}
}
push @ibase, ".";
if (defined $in) {
$inf = gensym();
open($inf, "<$in") or die "opening \"$in\": $!\n";
push @ibase, $1 if $in =~ m|^(.+)/[^/]+$|;
} else {
$inf = \*STDIN;
}
if (defined $out) {
open(STDOUT, ">$out") or die "opening \"$out\": $!\n";
}
while(defined $inf) {
INF: while(<$inf>) {
# Certain commands are discarded without further processing.
/^\@(?:
[a-z]+index # @*index: useful only in complete manual
|need # @need: useful only in printed manual
|(?:end\s+)?group # @group .. @end group: ditto
|page # @page: ditto
|node # @node: useful only in .info file
|(?:end\s+)?ifnottex # @ifnottex .. @end ifnottex: use contents
)\b/x and next;
chomp;
# Look for filename and title markers.
/^\@setfilename\s+([^.]+)/ and $fn = $1, next;
/^\@settitle\s+([^.]+)/ and $tl = postprocess($1), next;
# Identify a man title but keep only the one we are interested in.
/^\@c\s+man\s+title\s+([A-Za-z0-9-]+)\s+(.+)/ and do {
if (exists $defs{$1}) {
$fn = $1;
$tl = postprocess($2);
}
next;
};
/^\@include\s+(.+)$/ and do {
push @instack, $inf;
$inf = gensym();
for (@ibase) {
open($inf, "<" . $_ . "/" . $1) and next INF;
}
die "cannot open $1: $!\n";
};
/^\@chapter\s+([A-Za-z ]+)/ and do {
# close old chapter
$chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
# start new chapter
$chapter_name = $1, push (@chapters_sequence, $chapter_name);
$chapters{$chapter_name} = "" unless exists $chapters{$chapter_name};
$chapter = "";
$output = 1;
next;
};
/^\@bye/ and do {
# close old chapter
$chapters{$chapter_name} .= postprocess($chapter) if ($chapter_name);
last INF;
};
# handle variables
/^\@set\s+([a-zA-Z0-9_-]+)\s*(.*)$/ and do {
$defs{$1} = $2;
next;
};
/^\@clear\s+([a-zA-Z0-9_-]+)/ and do {
delete $defs{$1};
next;
};
next unless $output;
# Discard comments. (Can't do it above, because then we'd never see
# @c man lines.)
/^\@c\b/ and next;
# End-block handler goes up here because it needs to operate even
# if we are skipping.
/^\@end\s+([a-z]+)/ and do {
# Ignore @end foo, where foo is not an operation which may
# cause us to skip, if we are presently skipping.
my $ended = $1;
next if $skipping && $ended !~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/;
die "\@end $ended without \@$ended at line $.\n" unless defined $endw;
die "\@$endw ended by \@end $ended at line $.\n" unless $ended eq $endw;
$endw = pop @endwstack;
if ($ended =~ /^(?:ifset|ifclear|ignore|menu|iftex|ifhtml|ifnothtml)$/) {
$skipping = pop @skstack;
next;
} elsif ($ended =~ /^(?:example|smallexample|display)$/) {
$shift = "";
$_ = ""; # need a paragraph break
} elsif ($ended =~ /^(?:itemize|enumerate|[fv]?table)$/) {
$_ = "\n=back\n";
$ic = pop @icstack;
} else {
die "unknown command \@end $ended at line $.\n";
}
};
# We must handle commands which can cause skipping even while we
# are skipping, otherwise we will not process nested conditionals
# correctly.
/^\@ifset\s+([a-zA-Z0-9_-]+)/ and do {
push @endwstack, $endw;
push @skstack, $skipping;
$endw = "ifset";
$skipping = 1 unless exists $defs{$1};
next;
};
/^\@ifclear\s+([a-zA-Z0-9_-]+)/ and do {
push @endwstack, $endw;
push @skstack, $skipping;
$endw = "ifclear";
$skipping = 1 if exists $defs{$1};
next;
};
/^\@(ignore|menu|iftex|ifhtml|ifnothtml)\b/ and do {
push @endwstack, $endw;
push @skstack, $skipping;
$endw = $1;
$skipping = $endw !~ /ifnothtml/;
next;
};
next if $skipping;
# Character entities. First the ones that can be replaced by raw text
# or discarded outright:
s/\@copyright\{\}/(c)/g;
s/\@dots\{\}/.../g;
s/\@enddots\{\}/..../g;
s/\@([.!? ])/$1/g;
s/\@[:-]//g;
s/\@bullet(?:\{\})?/*/g;
s/\@TeX\{\}/TeX/g;
s/\@pounds\{\}/\#/g;
s/\@minus(?:\{\})?/-/g;
# Now the ones that have to be replaced by special escapes
# (which will be turned back into text by unmunge())
s/&/&amp;/g;
s/\@\{/&lbrace;/g;
s/\@\}/&rbrace;/g;
s/\@\@/&at;/g;
# Inside a verbatim block, handle @var specially.
if ($shift ne "") {
s/\@var\{([^\}]*)\}/<$1>/g;
}
# POD doesn't interpret E<> inside a verbatim block.
if ($shift eq "") {
s/</&lt;/g;
s/>/&gt;/g;
} else {
s/</&LT;/g;
s/>/&GT;/g;
}
# Single line command handlers.
/^\@(?:section|unnumbered|unnumberedsec|center|heading)\s+(.+)$/
and $_ = "\n=head2 $1\n";
/^\@(?:subsection|subheading)\s+(.+)$/
and $_ = "\n=head3 $1\n";
/^\@(?:subsubsection|subsubheading)\s+(.+)$/
and $_ = "\n=head4 $1\n";
# Block command handlers:
/^\@itemize\s*(\@[a-z]+|\*|-)?/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$ic = $1 ? $1 : "*";
$_ = "\n=over 4\n";
$endw = "itemize";
};
/^\@enumerate(?:\s+([a-zA-Z0-9]+))?/ and do {
push @endwstack, $endw;
push @icstack, $ic;
if (defined $1) {
$ic = $1 . ".";
} else {
$ic = "1.";
}
$_ = "\n=over 4\n";
$endw = "enumerate";
};
/^\@([fv]?table)\s+(\@[a-z]+)/ and do {
push @endwstack, $endw;
push @icstack, $ic;
$endw = $1;
$ic = $2;
$ic =~ s/\@(?:samp|strong|key|gcctabopt|option|env|command)/B/;
$ic =~ s/\@(?:code|kbd)/C/;
$ic =~ s/\@(?:dfn|var|emph|cite|i)/I/;
$ic =~ s/\@(?:file)/F/;
$_ = "\n=over 4\n";
};
/^\@((?:small)?example|display)/ and do {
push @endwstack, $endw;
$endw = $1;
$shift = "\t";
$_ = ""; # need a paragraph break
};
/^\@itemx?\s*(.+)?$/ and do {
if (defined $1) {
# Entity escapes prevent munging by the <> processing below.
$_ = "\n=item $ic\&LT;$1\&GT;\n";
} else {
$_ = "\n=item $ic\n";
$ic =~ y/A-Ya-y/B-Zb-z/;
$ic =~ s/(\d+)/$1 + 1/eg;
}
};
$chapter .= $shift.$_."\n";
}
# End of current file.
close($inf);
$inf = pop @instack;
}
die "No filename or title\n" unless defined $fn && defined $tl;
$chapters{NAME} = "$fn \- $tl\n";
$chapters{FOOTNOTES} .= "=back\n" if exists $chapters{FOOTNOTES};
unshift @chapters_sequence, "NAME";
for $chapter (@chapters_sequence) {
if (exists $chapters{$chapter}) {
$head = uc($chapter);
print "=head1 $head\n\n";
print scalar unmunge ($chapters{$chapter});
print "\n";
}
}
sub usage
{
die "usage: $0 [-D toggle...] [infile [outfile]]\n";
}
sub postprocess
{
local $_ = $_[0];
# @value{foo} is replaced by whatever 'foo' is defined as.
while (m/(\@value\{([a-zA-Z0-9_-]+)\})/g) {
if (! exists $defs{$2}) {
print STDERR "Option $2 not defined\n";
s/\Q$1\E//;
} else {
$value = $defs{$2};
s/\Q$1\E/$value/;
}
}
# Formatting commands.
# Temporary escape for @r.
s/\@r\{([^\}]*)\}/R<$1>/g;
s/\@(?:dfn|var|emph|cite|i)\{([^\}]*)\}/I<$1>/g;
s/\@(?:code|kbd)\{([^\}]*)\}/C<$1>/g;
s/\@(?:gccoptlist|samp|strong|key|option|env|command|b)\{([^\}]*)\}/B<$1>/g;
s/\@sc\{([^\}]*)\}/\U$1/g;
s/\@file\{([^\}]*)\}/F<$1>/g;
s/\@w\{([^\}]*)\}/S<$1>/g;
s/\@(?:dmn|math)\{([^\}]*)\}/$1/g;
# Cross references are thrown away, as are @noindent and @refill.
# (@noindent is impossible in .pod, and @refill is unnecessary.)
# @* is also impossible in .pod; we discard it and any newline that
# follows it. Similarly, our macro @gol must be discarded.
s/\@anchor{(?:[^\}]*)\}//g;
s/\(?\@xref\{(?:[^\}]*)\}(?:[^.<]|(?:<[^<>]*>))*\.\)?//g;
s/\s+\(\@pxref\{(?:[^\}]*)\}\)//g;
s/;\s+\@pxref\{(?:[^\}]*)\}//g;
s/\@ref\{([^\}]*)\}/$1/g;
s/\@noindent\s*//g;
s/\@refill//g;
s/\@gol//g;
s/\@\*\s*\n?//g;
# @uref can take one, two, or three arguments, with different
# semantics each time. @url and @email are just like @uref with
# one argument, for our purposes.
s/\@(?:uref|url|email)\{([^\},]*),?[^\}]*\}/&lt;B<$1>&gt;/g;
s/\@uref\{([^\},]*),([^\},]*)\}/$2 (C<$1>)/g;
s/\@uref\{([^\},]*),([^\},]*),([^\},]*)\}/$3/g;
# Turn B<blah I<blah> blah> into B<blah> I<blah> B<blah> to
# match Texinfo semantics of @emph inside @samp. Also handle @r
# inside bold.
s/&LT;/</g;
s/&GT;/>/g;
1 while s/B<((?:[^<>]|I<[^<>]*>)*)R<([^>]*)>/B<$1>${2}B</g;
1 while (s/B<([^<>]*)I<([^>]+)>/B<$1>I<$2>B</g);
1 while (s/I<([^<>]*)B<([^>]+)>/I<$1>B<$2>I</g);
s/[BI]<>//g;
s/([BI])<(\s+)([^>]+)>/$2$1<$3>/g;
s/([BI])<([^>]+?)(\s+)>/$1<$2>$3/g;
# Extract footnotes. This has to be done after all other
# processing because otherwise the regexp will choke on formatting
# inside @footnote.
while (/\@footnote/g) {
s/\@footnote\{([^\}]+)\}/[$fnno]/;
add_footnote($1, $fnno);
$fnno++;
}
return $_;
}
sub unmunge
{
# Replace escaped symbols with their equivalents.
local $_ = $_[0];
s/&lt;/E<lt>/g;
s/&gt;/E<gt>/g;
s/&lbrace;/\{/g;
s/&rbrace;/\}/g;
s/&at;/\@/g;
s/&amp;/&/g;
return $_;
}
sub add_footnote
{
unless (exists $chapters{FOOTNOTES}) {
$chapters{FOOTNOTES} = "\n=over 4\n\n";
}
$chapters{FOOTNOTES} .= "=item $fnno.\n\n"; $fnno++;
$chapters{FOOTNOTES} .= $_[0];
$chapters{FOOTNOTES} .= "\n\n";
}
# stolen from Symbol.pm
{
my $genseq = 0;
sub gensym
{
my $name = "GEN" . $genseq++;
my $ref = \*{$name};
delete $::{$name};
return $ref;
}
}

109
project/jni/ffmpeg/doc/viterbi.txt Normal file
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@@ -0,0 +1,109 @@
This is a quick description of the viterbi aka dynamic programing
algorthm.
Its reason for existence is that wikipedia has become very poor on
describing algorithms in a way that makes it useable for understanding
them or anything else actually. It tends now to describe the very same
algorithm under 50 different names and pages with few understandable
by even people who fully understand the algorithm and the theory behind.
Problem description: (that is what it can solve)
assume we have a 2d table, or you could call it a graph or matrix if you
prefer
O O O O O O O
O O O O O O O
O O O O O O O
O O O O O O O
That table has edges connecting points from each column to the next column
and each edge has a score like: (only some edge and scores shown to keep it
readable)
O--5--O-----O-----O-----O-----O
2 / 7 / \ / \ / \ /
\ / \ / \ / \ / \ /
O7-/--O--/--O--/--O--/--O--/--O
\/ \/ 1/ \/ \/ \/ \/ \/ \/ \/
/\ /\ 2\ /\ /\ /\ /\ /\ /\ /\
O3-/--O--/--O--/--O--/--O--/--O
/ \ / \ / \ / \ / \
1 \ 9 \ / \ / \ / \
O--2--O--1--O--5--O--3--O--8--O
Our goal is to find a path from left to right through it which
minimizes the sum of the score of all edges.
(and of course left/right is just a convention here it could be top down too)
Similarly the minimum could be the maximum by just fliping the sign,
Example of a path with scores:
O O O O O O O
>---O. O O .O-2-O O O
5. .7 .
O O-1-O O O 8 O O
.
O O O O O O-1-O---> (sum here is 24)
The viterbi algorthm now solves this simply column by column
For the previous column each point has a best path and a associated
score:
O-----5 O
\
\
O \ 1 O
\/
/\
O / 2 O
/
/
O-----2 O
To move one column forward we just need to find the best path and associated
scores for the next column
here are some edges we could choose from:
O-----5--3--O
\ \8
\ \
O \ 1--9--O
\/ \3
/\ \
O / 2--1--O
/ \2
/ \
O-----2--4--O
Finding the new best paths and scores for each point of our new column is
trivial given we know the previous column best paths and scores:
O-----0-----8
\
\
O \ 0----10
\/
/\
O / 0-----3
/ \
/ \
O 0 4
the viterbi algorthm continues exactly like this column for column until the
end and then just picks the path with the best score (above that would be the
one with score 3)
Author: Michael niedermayer
Copyright LGPL