Bundling ffmpeg, initial attempt. This builds on MinGW; Linux is untested as of yet.
This commit is contained in:
Ben "root" Anderson
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Release Notes
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=============
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* 2.1 "Fourier" October, 2013
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General notes
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-------------
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See the Changelog file for a list of significant changes. Note, there
|
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are many more new features and bugfixes than whats listed there.
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|
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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,
|
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make your report against the development code following the usual bug reporting
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guidelines.
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@chapter Authors
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The FFmpeg developers.
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|
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For details about the authorship, see the Git history of the project
|
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(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
|
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@command{git log} in the FFmpeg source directory, or browsing the
|
||||
online repository at @url{http://source.ffmpeg.org}.
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|
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Maintainers for the specific components are listed in the file
|
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@file{MAINTAINERS} in the source code tree.
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AVUtil
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======
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libavutil is a small lightweight library of generally useful functions.
|
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It is not a library for code needed by both libavcodec and libavformat.
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||||
|
||||
|
||||
Overview:
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||||
=========
|
||||
adler32.c adler32 checksum
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||||
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:
|
||||
======
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||||
* 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
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@c man begin BITSTREAM FILTERS
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|
||||
When you configure your FFmpeg build, all the supported bitstream
|
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filters are enabled by default. You can list all available ones using
|
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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
|
||||
|
||||
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
|
||||
bitstream filter.
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|
||||
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4
|
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ADTS header and removes the ADTS header.
|
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|
||||
This is required for example when copying an AAC stream from a raw
|
||||
ADTS AAC container to a FLV or a MOV/MP4 file.
|
||||
|
||||
@section chomp
|
||||
|
||||
Remove zero padding at the end of a packet.
|
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|
||||
@section dump_extra
|
||||
|
||||
Add extradata to the beginning of the filtered packets.
|
||||
|
||||
The additional argument specifies which packets should be filtered.
|
||||
It accepts the values:
|
||||
@table @samp
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||||
@item a
|
||||
add extradata to all key packets, but only if @var{local_header} is
|
||||
set in the @option{flags2} codec context field
|
||||
|
||||
@item k
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||||
add extradata to all key packets
|
||||
|
||||
@item e
|
||||
add extradata to all packets
|
||||
@end table
|
||||
|
||||
If not specified it is assumed @samp{k}.
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||||
|
||||
For example the following @command{ffmpeg} command forces a global
|
||||
header (thus disabling individual packet headers) in the H.264 packets
|
||||
generated by the @code{libx264} encoder, but corrects them by adding
|
||||
the header stored in extradata to the key packets:
|
||||
@example
|
||||
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
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||||
@end example
|
||||
|
||||
@section h264_mp4toannexb
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||||
|
||||
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").
|
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|
||||
For example to remux an MP4 file containing an H.264 stream to mpegts
|
||||
format with @command{ffmpeg}, you can use the command:
|
||||
|
||||
@example
|
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ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
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@end example
|
||||
|
||||
@section imx_dump_header
|
||||
|
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@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
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|
||||
@example
|
||||
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
|
||||
@end example
|
||||
|
||||
Unfortunately, these chunks are incomplete JPEG images, because
|
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they lack the DHT segment required for decoding. Quoting from
|
||||
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
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||||
|
||||
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_extra
|
||||
|
||||
@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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||||
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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.
|
||||
|
||||
@section libcelt
|
||||
|
||||
libcelt decoder wrapper.
|
||||
|
||||
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
|
||||
Requires the presence of the libcelt headers and library during configuration.
|
||||
You need to explicitly configure the build with @code{--enable-libcelt}.
|
||||
|
||||
@section libgsm
|
||||
|
||||
libgsm decoder wrapper.
|
||||
|
||||
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
|
||||
the presence of the libgsm headers and library during configuration. You need
|
||||
to explicitly configure the build with @code{--enable-libgsm}.
|
||||
|
||||
This decoder supports both the ordinary GSM and the Microsoft variant.
|
||||
|
||||
@section libilbc
|
||||
|
||||
libilbc decoder wrapper.
|
||||
|
||||
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
|
||||
audio codec. Requires the presence of the libilbc headers and library during
|
||||
configuration. You need to explicitly configure the build with
|
||||
@code{--enable-libilbc}.
|
||||
|
||||
@subsection Options
|
||||
|
||||
The following option is supported by the libilbc wrapper.
|
||||
|
||||
@table @option
|
||||
@item enhance
|
||||
|
||||
Enable the enhancement of the decoded audio when set to 1. The default
|
||||
value is 0 (disabled).
|
||||
|
||||
@end table
|
||||
|
||||
@section libopencore-amrnb
|
||||
|
||||
libopencore-amrnb decoder wrapper.
|
||||
|
||||
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
|
||||
Narrowband audio codec. Using it requires the presence of the
|
||||
libopencore-amrnb headers and library during configuration. You need to
|
||||
explicitly configure the build with @code{--enable-libopencore-amrnb}.
|
||||
|
||||
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
|
||||
without this library.
|
||||
|
||||
@section libopencore-amrwb
|
||||
|
||||
libopencore-amrwb decoder wrapper.
|
||||
|
||||
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
|
||||
Wideband audio codec. Using it requires the presence of the
|
||||
libopencore-amrwb headers and library during configuration. You need to
|
||||
explicitly configure the build with @code{--enable-libopencore-amrwb}.
|
||||
|
||||
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
|
||||
without this library.
|
||||
|
||||
@section libopus
|
||||
|
||||
libopus decoder wrapper.
|
||||
|
||||
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
|
||||
Requires the presence of the libopus headers and library during
|
||||
configuration. You need to explicitly configure the build with
|
||||
@code{--enable-libopus}.
|
||||
|
||||
@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
|
||||
|
||||
@section libzvbi-teletext
|
||||
|
||||
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
|
||||
subtitles. Requires the presence of the libzvbi headers and library during
|
||||
configuration. You need to explicitly configure the build with
|
||||
@code{--enable-libzvbi}.
|
||||
|
||||
@subsection Options
|
||||
|
||||
@table @option
|
||||
@item txt_page
|
||||
List of teletext page numbers to decode. You may use the special * string to
|
||||
match all pages. Pages that do not match the specified list are dropped.
|
||||
Default value is *.
|
||||
@item txt_chop_top
|
||||
Discards the top teletext line. Default value is 1.
|
||||
@item txt_format
|
||||
Specifies the format of the decoded subtitles. The teletext decoder is capable
|
||||
of decoding the teletext pages to bitmaps or to simple text, you should use
|
||||
"bitmap" for teletext pages, because certain graphics and colors cannot be
|
||||
expressed in simple text. You might use "text" for teletext based subtitles if
|
||||
your application can handle simple text based subtitles. Default value is
|
||||
bitmap.
|
||||
@item txt_left
|
||||
X offset of generated bitmaps, default is 0.
|
||||
@item txt_top
|
||||
Y offset of generated bitmaps, default is 0.
|
||||
@item txt_chop_spaces
|
||||
Chops leading and trailing spaces and removes empty lines from the generated
|
||||
text. This option is useful for teletext based subtitles where empty spaces may
|
||||
be present at the start or at the end of the lines or empty lines may be
|
||||
present between the subtitle lines because of double-sized teletext charactes.
|
||||
Default value is 1.
|
||||
@item txt_duration
|
||||
Sets the display duration of the decoded teletext pages or subtitles in
|
||||
miliseconds. Default value is 30000 which is 30 seconds.
|
||||
@item txt_transparent
|
||||
Force transparent background of the generated teletext bitmaps. Default value
|
||||
is 0 which means an opaque (black) background.
|
||||
@end table
|
||||
|
||||
@c man end SUBTILES DECODERS
|
||||
@@ -0,0 +1,165 @@
|
||||
a.summary-letter {
|
||||
text-decoration: none;
|
||||
}
|
||||
|
||||
a {
|
||||
color: #2D6198;
|
||||
}
|
||||
|
||||
a:visited {
|
||||
color: #884488;
|
||||
}
|
||||
|
||||
#banner {
|
||||
background-color: white;
|
||||
position: relative;
|
||||
text-align: center;
|
||||
}
|
||||
|
||||
#banner img {
|
||||
padding-bottom: 1px;
|
||||
padding-top: 5px;
|
||||
}
|
||||
|
||||
#body {
|
||||
margin-left: 1em;
|
||||
margin-right: 1em;
|
||||
}
|
||||
|
||||
body {
|
||||
background-color: #313131;
|
||||
margin: 0;
|
||||
text-align: justify;
|
||||
}
|
||||
|
||||
.center {
|
||||
margin-left: auto;
|
||||
margin-right: auto;
|
||||
text-align: center;
|
||||
}
|
||||
|
||||
#container {
|
||||
background-color: white;
|
||||
color: #202020;
|
||||
margin-left: 1em;
|
||||
margin-right: 1em;
|
||||
}
|
||||
|
||||
#footer {
|
||||
text-align: center;
|
||||
}
|
||||
|
||||
h1 a, h2 a, h3 a, h4 a {
|
||||
text-decoration: inherit;
|
||||
color: inherit;
|
||||
}
|
||||
|
||||
h1, h2, h3, h4 {
|
||||
padding-left: 0.4em;
|
||||
border-radius: 4px;
|
||||
padding-bottom: 0.25em;
|
||||
padding-top: 0.25em;
|
||||
border: 1px solid #6A996A;
|
||||
}
|
||||
|
||||
h1 {
|
||||
background-color: #7BB37B;
|
||||
color: #151515;
|
||||
font-size: 1.2em;
|
||||
padding-bottom: 0.3em;
|
||||
padding-top: 0.3em;
|
||||
}
|
||||
|
||||
h2 {
|
||||
color: #313131;
|
||||
font-size: 1.0em;
|
||||
background-color: #ABE3AB;
|
||||
}
|
||||
|
||||
h3 {
|
||||
color: #313131;
|
||||
font-size: 0.9em;
|
||||
margin-bottom: -6px;
|
||||
background-color: #BBF3BB;
|
||||
}
|
||||
|
||||
h4 {
|
||||
color: #313131;
|
||||
font-size: 0.8em;
|
||||
margin-bottom: -8px;
|
||||
background-color: #D1FDD1;
|
||||
}
|
||||
|
||||
img {
|
||||
border: 0;
|
||||
}
|
||||
|
||||
#navbar {
|
||||
background-color: #738073;
|
||||
border-bottom: 1px solid #5C665C;
|
||||
border-top: 1px solid #5C665C;
|
||||
margin-top: 12px;
|
||||
padding: 0.3em;
|
||||
position: relative;
|
||||
text-align: center;
|
||||
}
|
||||
|
||||
#navbar a, #navbar_secondary a {
|
||||
color: white;
|
||||
padding: 0.3em;
|
||||
text-decoration: none;
|
||||
}
|
||||
|
||||
#navbar a:hover, #navbar_secondary a:hover {
|
||||
background-color: #313131;
|
||||
color: white;
|
||||
text-decoration: none;
|
||||
}
|
||||
|
||||
#navbar_secondary {
|
||||
background-color: #738073;
|
||||
border-bottom: 1px solid #5C665C;
|
||||
border-left: 1px solid #5C665C;
|
||||
border-right: 1px solid #5C665C;
|
||||
padding: 0.3em;
|
||||
position: relative;
|
||||
text-align: center;
|
||||
}
|
||||
|
||||
p {
|
||||
margin-left: 1em;
|
||||
margin-right: 1em;
|
||||
}
|
||||
|
||||
pre {
|
||||
margin-left: 3em;
|
||||
margin-right: 3em;
|
||||
padding: 0.3em;
|
||||
border: 1px solid #bbb;
|
||||
background-color: #f7f7f7;
|
||||
}
|
||||
|
||||
dl dt {
|
||||
font-weight: bold;
|
||||
}
|
||||
|
||||
#proj_desc {
|
||||
font-size: 1.2em;
|
||||
}
|
||||
|
||||
#repos {
|
||||
margin-left: 1em;
|
||||
margin-right: 1em;
|
||||
border-collapse: collapse;
|
||||
border: solid 1px #6A996A;
|
||||
}
|
||||
|
||||
#repos th {
|
||||
background-color: #7BB37B;
|
||||
border: solid 1px #6A996A;
|
||||
}
|
||||
|
||||
#repos td {
|
||||
padding: 0.2em;
|
||||
border: solid 1px #6A996A;
|
||||
}
|
||||
@@ -0,0 +1,375 @@
|
||||
@chapter Demuxers
|
||||
@c man begin DEMUXERS
|
||||
|
||||
Demuxers are configured elements in FFmpeg that can 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 @code{--list-demuxers}.
|
||||
|
||||
You can disable all the demuxers using the configure option
|
||||
@code{--disable-demuxers}, and selectively enable a single demuxer with
|
||||
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
|
||||
with the option @code{--disable-demuxer=@var{DEMUXER}}.
|
||||
|
||||
The option @code{-formats} of the ff* tools will display the list of
|
||||
enabled demuxers.
|
||||
|
||||
The description of some of the currently available demuxers follows.
|
||||
|
||||
@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 asf
|
||||
|
||||
Advanced Systems Format demuxer.
|
||||
|
||||
This demuxer is used to demux ASF files and MMS network streams.
|
||||
|
||||
@table @option
|
||||
@item -no_resync_search @var{bool}
|
||||
Do not try to resynchronize by looking for a certain optional start code.
|
||||
@end table
|
||||
|
||||
@anchor{concat}
|
||||
@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.).
|
||||
|
||||
The duration of each file is used to adjust the timestamps of the next file:
|
||||
if the duration is incorrect (because it was computed using the bit-rate or
|
||||
because the file is truncated, for example), it can cause artifacts. The
|
||||
@code{duration} directive can be used to override the duration stored in
|
||||
each file.
|
||||
|
||||
@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.
|
||||
|
||||
All subsequent directives apply to that file.
|
||||
|
||||
@item @code{ffconcat version 1.0}
|
||||
Identify the script type and version. It also sets the @option{safe} option
|
||||
to 1 if it was to its default -1.
|
||||
|
||||
To make FFmpeg recognize the format automatically, this directive must
|
||||
appears exactly as is (no extra space or byte-order-mark) on the very first
|
||||
line of the script.
|
||||
|
||||
@item @code{duration @var{dur}}
|
||||
Duration of the file. This information can be specified from the file;
|
||||
specifying it here may be more efficient or help if the information from the
|
||||
file is not available or accurate.
|
||||
|
||||
If the duration is set for all files, then it is possible to seek in the
|
||||
whole concatenated video.
|
||||
|
||||
@end table
|
||||
|
||||
@subsection Options
|
||||
|
||||
This demuxer accepts the following option:
|
||||
|
||||
@table @option
|
||||
|
||||
@item safe
|
||||
If set to 1, reject unsafe file paths. A file path is considered safe if it
|
||||
does not contain a protocol specification and is relative and all components
|
||||
only contain characters from the portable character set (letters, digits,
|
||||
period, underscore and hyphen) and have no period at the beginning of a
|
||||
component.
|
||||
|
||||
If set to 0, any file name is accepted.
|
||||
|
||||
The default is -1, it is equivalent to 1 if the format was automatically
|
||||
probed and 0 otherwise.
|
||||
|
||||
@end table
|
||||
|
||||
@section flv
|
||||
|
||||
Adobe Flash Video Format demuxer.
|
||||
|
||||
This demuxer is used to demux FLV files and RTMP network streams.
|
||||
|
||||
@table @option
|
||||
@item -flv_metadata @var{bool}
|
||||
Allocate the streams according to the onMetaData array content.
|
||||
@end table
|
||||
|
||||
@section libgme
|
||||
|
||||
The Game Music Emu library is a collection of video game music file emulators.
|
||||
|
||||
See @url{http://code.google.com/p/game-music-emu/} for more information.
|
||||
|
||||
Some files have multiple tracks. The demuxer will pick the first track by
|
||||
default. The @option{track_index} option can be used to select a different
|
||||
track. Track indexes start at 0. The demuxer exports the number of tracks as
|
||||
@var{tracks} meta data entry.
|
||||
|
||||
For very large files, the @option{max_size} option may have to be adjusted.
|
||||
|
||||
@section libquvi
|
||||
|
||||
Play media from Internet services using the quvi project.
|
||||
|
||||
The demuxer accepts a @option{format} option to request a specific quality. It
|
||||
is by default set to @var{best}.
|
||||
|
||||
See @url{http://quvi.sourceforge.net/} for more information.
|
||||
|
||||
FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
|
||||
enabled.
|
||||
|
||||
@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 frame rate 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 ts_from_file
|
||||
If set to 1, will set frame timestamp to modification time of image file. Note
|
||||
that monotonity of timestamps is not provided: images go in the same order as
|
||||
without this option. Default value is 0.
|
||||
@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 -framerate 10 -i 'img-%03d.jpeg' out.mkv
|
||||
@end example
|
||||
|
||||
@item
|
||||
As above, but start by reading from a file with index 100 in the sequence:
|
||||
@example
|
||||
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
|
||||
@end example
|
||||
|
||||
@item
|
||||
Read images matching the "*.png" glob pattern , that is all the files
|
||||
terminating with the ".png" suffix:
|
||||
@example
|
||||
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
@section mpegts
|
||||
|
||||
MPEG-2 transport stream demuxer.
|
||||
|
||||
@table @option
|
||||
|
||||
@item fix_teletext_pts
|
||||
Overrides teletext packet PTS and DTS values with the timestamps calculated
|
||||
from the PCR of the first program which the teletext stream is part of and is
|
||||
not discarded. Default value is 1, set this option to 0 if you want your
|
||||
teletext packet PTS and DTS values untouched.
|
||||
@end table
|
||||
|
||||
@section rawvideo
|
||||
|
||||
Raw video demuxer.
|
||||
|
||||
This demuxer allows to read raw video data. Since there is no header
|
||||
specifying the assumed video parameters, the user must specify them
|
||||
in order to be able to decode the data correctly.
|
||||
|
||||
This demuxer accepts the following options:
|
||||
@table @option
|
||||
|
||||
@item framerate
|
||||
Set input video frame rate. Default value is 25.
|
||||
|
||||
@item pixel_format
|
||||
Set the input video pixel format. Default value is @code{yuv420p}.
|
||||
|
||||
@item video_size
|
||||
Set the input video size. This value must be specified explicitly.
|
||||
@end table
|
||||
|
||||
For example to read a rawvideo file @file{input.raw} with
|
||||
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
|
||||
size of @code{320x240}, and a frame rate of 10 images per second, use
|
||||
the command:
|
||||
@example
|
||||
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
|
||||
@end example
|
||||
|
||||
@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 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 DEMUXERS
|
||||
@@ -0,0 +1,797 @@
|
||||
\input texinfo @c -*- texinfo -*-
|
||||
|
||||
@settitle Developer Documentation
|
||||
@titlepage
|
||||
@center @titlefont{Developer Documentation}
|
||||
@end titlepage
|
||||
|
||||
@top
|
||||
|
||||
@contents
|
||||
|
||||
@chapter Developers Guide
|
||||
|
||||
@section Notes for external developers
|
||||
|
||||
This document is mostly useful for internal FFmpeg developers.
|
||||
External developers who need to use the API in their application should
|
||||
refer to the API doxygen documentation in the public headers, and
|
||||
check the examples in @file{doc/examples} and in the source code to
|
||||
see how the public API is employed.
|
||||
|
||||
You can use the FFmpeg libraries in your commercial program, but you
|
||||
are encouraged to @emph{publish any patch you make}. In this case the
|
||||
best way to proceed is to send your patches to the ffmpeg-devel
|
||||
mailing list following the guidelines illustrated in the remainder of
|
||||
this document.
|
||||
|
||||
For more detailed legal information about the use of FFmpeg in
|
||||
external programs read the @file{LICENSE} file in the source tree and
|
||||
consult @url{http://ffmpeg.org/legal.html}.
|
||||
|
||||
@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 should be composed with underscores (_), not CamelCase. For example,
|
||||
@samp{avfilter_get_video_buffer} is an acceptable 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 the following conventions for naming variables and functions:
|
||||
|
||||
@itemize @bullet
|
||||
@item
|
||||
For local variables no prefix is required.
|
||||
|
||||
@item
|
||||
For file-scope variables and functions declared as @code{static}, no prefix
|
||||
is required.
|
||||
|
||||
@item
|
||||
For variables and functions visible outside of file scope, but only used
|
||||
internally by a library, an @code{ff_} prefix should be used,
|
||||
e.g. @samp{ff_w64_demuxer}.
|
||||
|
||||
@item
|
||||
For variables and functions visible outside of file scope, used internally
|
||||
across multiple libraries, use @code{avpriv_} as prefix, for example,
|
||||
@samp{avpriv_aac_parse_header}.
|
||||
|
||||
@item
|
||||
Each library has its own prefix for public symbols, in addition to the
|
||||
commonly used @code{av_} (@code{avformat_} for libavformat,
|
||||
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
|
||||
Check the existing code and choose names accordingly.
|
||||
Note that some symbols without these prefixes are also exported for
|
||||
retro-compatibility reasons. These exceptions are declared in the
|
||||
@code{lib<name>/lib<name>.v} files.
|
||||
@end itemize
|
||||
|
||||
Furthermore, name space reserved for the system should not be invaded.
|
||||
Identifiers ending in @code{_t} are reserved by
|
||||
@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
|
||||
Also avoid names starting with @code{__} or @code{_} followed by an uppercase
|
||||
letter as they are reserved by the C standard. Names starting with @code{_}
|
||||
are reserved at the file level and may not be used for externally visible
|
||||
symbols. If in doubt, just avoid names starting with @code{_} altogether.
|
||||
|
||||
@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,automake 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
|
||||
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
|
||||
including an "or any later version" clause, or, if you prefer
|
||||
a gift-style license, the
|
||||
@uref{http://www.isc.org/software/license/, ISC} or
|
||||
@uref{http://mit-license.org/, MIT} license.
|
||||
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
|
||||
an "or any later version" clause is also acceptable, but LGPL is
|
||||
preferred.
|
||||
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.
|
||||
|
||||
@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
|
||||
The commit message should have a short first line in the form of
|
||||
a @samp{topic: short description} as a header, separated by a newline
|
||||
from the body consisting of an explanation of why the change is necessary.
|
||||
If the commit fixes a known bug on the bug tracker, the commit message
|
||||
should include its bug ID. Referring to the issue on the bug tracker does
|
||||
not exempt you from writing an excerpt of the bug in the commit message.
|
||||
|
||||
@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
|
||||
Make sure that no parts of the codebase that you maintain are missing from the
|
||||
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
|
||||
your name after it.
|
||||
If at some point you no longer want to maintain some code, then please help
|
||||
finding a new maintainer and also don't forget updating the @file{MAINTAINERS} file.
|
||||
@end enumerate
|
||||
|
||||
We think our rules are not too hard. If you have comments, contact us.
|
||||
|
||||
@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.
|
||||
|
||||
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.
|
||||
|
||||
Patches should be posted to the
|
||||
@uref{http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
|
||||
mailing list. Use @code{git send-email} when possible since it will properly
|
||||
send patches without requiring extra care. If you cannot, then send patches
|
||||
as base64-encoded attachments, so your patch is not trashed during
|
||||
transmission.
|
||||
|
||||
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, the noise bitstream filter, and
|
||||
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
|
||||
should not crash, end in a (near) infinite loop, or allocate ridiculous
|
||||
amounts of memory 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.
|
||||
|
||||
@item
|
||||
Test your code with valgrind and or Address Sanitizer to ensure it's free
|
||||
of leaks, out of array accesses, etc.
|
||||
@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.
|
||||
|
||||
|
||||
@subsection Visualizing Test Coverage
|
||||
|
||||
The FFmpeg build system allows visualizing the test coverage in an easy
|
||||
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
|
||||
the following steps:
|
||||
|
||||
@enumerate
|
||||
@item
|
||||
Configure to compile with instrumentation enabled:
|
||||
@code{configure --toolchain=gcov}.
|
||||
|
||||
@item
|
||||
Run your test case, either manually or via FATE. This can be either
|
||||
the full FATE regression suite, or any arbitrary invocation of any
|
||||
front-end tool provided by FFmpeg, in any combination.
|
||||
|
||||
@item
|
||||
Run @code{make lcov} to generate coverage data in HTML format.
|
||||
|
||||
@item
|
||||
View @code{lcov/index.html} in your preferred HTML viewer.
|
||||
@end enumerate
|
||||
|
||||
You can use the command @code{make lcov-reset} to reset the coverage
|
||||
measurements. You will need to rerun @code{make lcov} after running a
|
||||
new test.
|
||||
|
||||
@subsection Using Valgrind
|
||||
|
||||
The configure script provides a shortcut for using valgrind to spot bugs
|
||||
related to memory handling. Just add the option
|
||||
@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
|
||||
to your configure line, and reasonable defaults will be set for running
|
||||
FATE under the supervision of either the @strong{memcheck} or the
|
||||
@strong{massif} tool of the valgrind suite.
|
||||
|
||||
In case you need finer control over how valgrind is invoked, use the
|
||||
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
|
||||
your configure line instead.
|
||||
|
||||
@anchor{Release process}
|
||||
@section Release process
|
||||
|
||||
FFmpeg maintains a set of @strong{release branches}, which are the
|
||||
recommended deliverable for system integrators and distributors (such as
|
||||
Linux distributions, etc.). At regular times, a @strong{release
|
||||
manager} prepares, tests and publishes tarballs on the
|
||||
@url{http://ffmpeg.org} website.
|
||||
|
||||
There are two kinds of releases:
|
||||
|
||||
@enumerate
|
||||
@item
|
||||
@strong{Major releases} always include the latest and greatest
|
||||
features and functionality.
|
||||
|
||||
@item
|
||||
@strong{Point releases} are cut from @strong{release} branches,
|
||||
which are named @code{release/X}, with @code{X} being the release
|
||||
version number.
|
||||
@end enumerate
|
||||
|
||||
Note that we promise to our users that shared libraries from any FFmpeg
|
||||
release never break programs that have been @strong{compiled} against
|
||||
previous versions of @strong{the same release series} in any case!
|
||||
|
||||
However, from time to time, we do make API changes that require adaptations
|
||||
in applications. Such changes are only allowed in (new) major releases and
|
||||
require further steps such as bumping library version numbers and/or
|
||||
adjustments to the symbol versioning file. Please discuss such changes
|
||||
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
|
||||
|
||||
@anchor{Criteria for Point Releases}
|
||||
@subsection Criteria for Point Releases
|
||||
|
||||
Changes that match the following criteria are valid candidates for
|
||||
inclusion into a point release:
|
||||
|
||||
@enumerate
|
||||
@item
|
||||
Fixes a security issue, preferably identified by a @strong{CVE
|
||||
number} issued by @url{http://cve.mitre.org/}.
|
||||
|
||||
@item
|
||||
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
|
||||
|
||||
@item
|
||||
Improves the included documentation.
|
||||
|
||||
@item
|
||||
Retains both source code and binary compatibility with previous
|
||||
point releases of the same release branch.
|
||||
@end enumerate
|
||||
|
||||
The order for checking the rules is (1 OR 2 OR 3) AND 4.
|
||||
|
||||
|
||||
@subsection Release Checklist
|
||||
|
||||
The release process involves the following steps:
|
||||
|
||||
@enumerate
|
||||
@item
|
||||
Ensure that the @file{RELEASE} file contains the version number for
|
||||
the upcoming release.
|
||||
|
||||
@item
|
||||
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
|
||||
|
||||
@item
|
||||
Announce the intent to do a release to the mailing list.
|
||||
|
||||
@item
|
||||
Make sure all relevant security fixes have been backported. See
|
||||
@url{https://ffmpeg.org/security.html}.
|
||||
|
||||
@item
|
||||
Ensure that the FATE regression suite still passes in the release
|
||||
branch on at least @strong{i386} and @strong{amd64}
|
||||
(cf. @ref{Regression tests}).
|
||||
|
||||
@item
|
||||
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
|
||||
supplementing files that contain @code{gpg} signatures
|
||||
|
||||
@item
|
||||
Publish the tarballs at @url{http://ffmpeg.org/releases}. Create and
|
||||
push an annotated tag in the form @code{nX}, with @code{X}
|
||||
containing the version number.
|
||||
|
||||
@item
|
||||
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
|
||||
with a news entry for the website.
|
||||
|
||||
@item
|
||||
Publish the news entry.
|
||||
|
||||
@item
|
||||
Send announcement to the mailing list.
|
||||
@end enumerate
|
||||
|
||||
@bye
|
||||
@@ -0,0 +1,21 @@
|
||||
@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
|
||||
@@ -0,0 +1,14 @@
|
||||
#!/bin/sh
|
||||
|
||||
SRC_PATH="${1}"
|
||||
DOXYFILE="${2}"
|
||||
|
||||
shift 2
|
||||
|
||||
doxygen - <<EOF
|
||||
@INCLUDE = ${DOXYFILE}
|
||||
INPUT = $@
|
||||
HTML_HEADER = ${SRC_PATH}/doc/doxy/header.html
|
||||
HTML_FOOTER = ${SRC_PATH}/doc/doxy/footer.html
|
||||
HTML_STYLESHEET = ${SRC_PATH}/doc/doxy/doxy_stylesheet.css
|
||||
EOF
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,9 @@
|
||||
|
||||
<footer class="footer pagination-right">
|
||||
<span class="label label-info">
|
||||
Generated on $datetime for $projectname by <a href="http://www.doxygen.org/index.html">doxygen</a> $doxygenversion
|
||||
</span>
|
||||
</footer>
|
||||
</div>
|
||||
</body>
|
||||
</html>
|
||||
@@ -0,0 +1,16 @@
|
||||
<!DOCTYPE html>
|
||||
<html>
|
||||
<head>
|
||||
<meta http-equiv="Content-Type" content="text/html; charset=UTF-8"/>
|
||||
<meta http-equiv="X-UA-Compatible" content="IE=9"/>
|
||||
<!--BEGIN PROJECT_NAME--><title>$projectname: $title</title><!--END PROJECT_NAME-->
|
||||
<!--BEGIN !PROJECT_NAME--><title>$title</title><!--END !PROJECT_NAME-->
|
||||
<link href="$relpath$doxy_stylesheet.css" rel="stylesheet" type="text/css" />
|
||||
<!--Header replace -->
|
||||
|
||||
</head>
|
||||
|
||||
<div class="container">
|
||||
|
||||
<!--Header replace -->
|
||||
<div class="menu">
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,174 @@
|
||||
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
|
||||
@@ -0,0 +1,18 @@
|
||||
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
|
||||
doc/examples and run a command such as PKG_CONFIG_PATH=pc-uninstalled make.
|
||||
@@ -0,0 +1,650 @@
|
||||
/*
|
||||
* 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_channels = 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_channels) {
|
||||
best_ch_layout = *p;
|
||||
best_nb_channels = 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;
|
||||
}
|
||||
@@ -0,0 +1,341 @@
|
||||
/*
|
||||
* 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 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;
|
||||
int decoded = pkt.size;
|
||||
|
||||
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;
|
||||
}
|
||||
/* Some audio decoders decode only part of the packet, and have to be
|
||||
* called again with the remainder of the packet data.
|
||||
* Sample: fate-suite/lossless-audio/luckynight-partial.shn
|
||||
* Also, some decoders might over-read the packet. */
|
||||
decoded = FFMIN(ret, pkt.size);
|
||||
|
||||
if (*got_frame) {
|
||||
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
|
||||
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));
|
||||
|
||||
/* Write the raw audio data samples of the first plane. This works
|
||||
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
|
||||
* most audio decoders output planar audio, which uses a separate
|
||||
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
|
||||
* In other words, this code will write only the first audio channel
|
||||
* in these cases.
|
||||
* You should use libswresample or libavfilter to convert the frame
|
||||
* to packed data. */
|
||||
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
|
||||
}
|
||||
}
|
||||
|
||||
return decoded;
|
||||
}
|
||||
|
||||
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) {
|
||||
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;
|
||||
}
|
||||
}
|
||||
|
||||
/* 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) {
|
||||
AVPacket orig_pkt = pkt;
|
||||
do {
|
||||
ret = decode_packet(&got_frame, 0);
|
||||
if (ret < 0)
|
||||
break;
|
||||
pkt.data += ret;
|
||||
pkt.size -= ret;
|
||||
} while (pkt.size > 0);
|
||||
av_free_packet(&orig_pkt);
|
||||
}
|
||||
|
||||
/* 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) {
|
||||
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
|
||||
int n_channels = audio_dec_ctx->channels;
|
||||
const char *fmt;
|
||||
|
||||
if (av_sample_fmt_is_planar(sfmt)) {
|
||||
const char *packed = av_get_sample_fmt_name(sfmt);
|
||||
printf("Warning: the sample format the decoder produced is planar "
|
||||
"(%s). This example will output the first channel only.\n",
|
||||
packed ? packed : "?");
|
||||
sfmt = av_get_packed_sample_fmt(sfmt);
|
||||
n_channels = 1;
|
||||
}
|
||||
|
||||
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
|
||||
goto end;
|
||||
|
||||
printf("Play the output audio file with the command:\n"
|
||||
"ffplay -f %s -ac %d -ar %d %s\n",
|
||||
fmt, n_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]);
|
||||
|
||||
return ret < 0;
|
||||
}
|
||||
@@ -0,0 +1,265 @@
|
||||
/*
|
||||
* 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>
|
||||
#include <libavutil/opt.h>
|
||||
|
||||
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
|
||||
static 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;
|
||||
av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
|
||||
|
||||
/* 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("abuffersink");
|
||||
AVFilterInOut *outputs = avfilter_inout_alloc();
|
||||
AVFilterInOut *inputs = avfilter_inout_alloc();
|
||||
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
|
||||
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
|
||||
static const int out_sample_rates[] = { 8000, -1 };
|
||||
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. */
|
||||
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
|
||||
NULL, NULL, filter_graph);
|
||||
if (ret < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
|
||||
AV_OPT_SEARCH_CHILDREN);
|
||||
if (ret < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
|
||||
AV_OPT_SEARCH_CHILDREN);
|
||||
if (ret < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
|
||||
return ret;
|
||||
}
|
||||
|
||||
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
|
||||
AV_OPT_SEARCH_CHILDREN);
|
||||
if (ret < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\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_ptr(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_frame(const AVFrame *frame)
|
||||
{
|
||||
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(av_frame_get_channel_layout(frame));
|
||||
const uint16_t *p = (uint16_t*)frame->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 = av_frame_alloc();
|
||||
AVFrame *filt_frame = av_frame_alloc();
|
||||
int got_frame;
|
||||
|
||||
if (!frame || !filt_frame) {
|
||||
perror("Could not allocate frame");
|
||||
exit(1);
|
||||
}
|
||||
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) {
|
||||
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_flags(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_frame(buffersink_ctx, filt_frame);
|
||||
if(ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
|
||||
break;
|
||||
if(ret < 0)
|
||||
goto end;
|
||||
print_frame(filt_frame);
|
||||
av_frame_unref(filt_frame);
|
||||
}
|
||||
}
|
||||
}
|
||||
av_free_packet(&packet);
|
||||
}
|
||||
end:
|
||||
avfilter_graph_free(&filter_graph);
|
||||
if (dec_ctx)
|
||||
avcodec_close(dec_ctx);
|
||||
avformat_close_input(&fmt_ctx);
|
||||
av_frame_free(&frame);
|
||||
av_frame_free(&filt_frame);
|
||||
|
||||
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);
|
||||
}
|
||||
@@ -0,0 +1,251 @@
|
||||
/*
|
||||
* 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("buffersink");
|
||||
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_ptr(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_frame(const AVFrame *frame, AVRational time_base)
|
||||
{
|
||||
int x, y;
|
||||
uint8_t *p0, *p;
|
||||
int64_t delay;
|
||||
|
||||
if (frame->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(frame->pts - last_pts,
|
||||
time_base, AV_TIME_BASE_Q);
|
||||
if (delay > 0 && delay < 1000000)
|
||||
usleep(delay);
|
||||
}
|
||||
last_pts = frame->pts;
|
||||
}
|
||||
|
||||
/* Trivial ASCII grayscale display. */
|
||||
p0 = frame->data[0];
|
||||
puts("\033c");
|
||||
for (y = 0; y < frame->height; y++) {
|
||||
p = p0;
|
||||
for (x = 0; x < frame->width; x++)
|
||||
putchar(" .-+#"[*(p++) / 52]);
|
||||
putchar('\n');
|
||||
p0 += frame->linesize[0];
|
||||
}
|
||||
fflush(stdout);
|
||||
}
|
||||
|
||||
int main(int argc, char **argv)
|
||||
{
|
||||
int ret;
|
||||
AVPacket packet;
|
||||
AVFrame *frame = av_frame_alloc();
|
||||
AVFrame *filt_frame = av_frame_alloc();
|
||||
int got_frame;
|
||||
|
||||
if (!frame || !filt_frame) {
|
||||
perror("Could not allocate frame");
|
||||
exit(1);
|
||||
}
|
||||
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) {
|
||||
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_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
|
||||
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
|
||||
break;
|
||||
}
|
||||
|
||||
/* pull filtered frames from the filtergraph */
|
||||
while (1) {
|
||||
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
|
||||
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
|
||||
break;
|
||||
if (ret < 0)
|
||||
goto end;
|
||||
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
|
||||
av_frame_unref(filt_frame);
|
||||
}
|
||||
}
|
||||
}
|
||||
av_free_packet(&packet);
|
||||
}
|
||||
end:
|
||||
avfilter_graph_free(&filter_graph);
|
||||
if (dec_ctx)
|
||||
avcodec_close(dec_ctx);
|
||||
avformat_close_input(&fmt_ctx);
|
||||
av_frame_free(&frame);
|
||||
av_frame_free(&filt_frame);
|
||||
|
||||
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);
|
||||
}
|
||||
@@ -0,0 +1,56 @@
|
||||
/*
|
||||
* 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;
|
||||
}
|
||||
@@ -0,0 +1,564 @@
|
||||
/*
|
||||
* 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/opt.h>
|
||||
#include <libavutil/mathematics.h>
|
||||
#include <libavformat/avformat.h>
|
||||
#include <libswscale/swscale.h>
|
||||
#include <libswresample/swresample.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;
|
||||
|
||||
/* 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:
|
||||
c->sample_fmt = AV_SAMPLE_FMT_FLTP;
|
||||
c->bit_rate = 64000;
|
||||
c->sample_rate = 44100;
|
||||
c->channels = 2;
|
||||
break;
|
||||
|
||||
case AVMEDIA_TYPE_VIDEO:
|
||||
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 uint8_t **src_samples_data;
|
||||
static int src_samples_linesize;
|
||||
static int src_nb_samples;
|
||||
|
||||
static int max_dst_nb_samples;
|
||||
uint8_t **dst_samples_data;
|
||||
int dst_samples_linesize;
|
||||
int dst_samples_size;
|
||||
|
||||
struct SwrContext *swr_ctx = NULL;
|
||||
|
||||
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;
|
||||
|
||||
src_nb_samples = c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE ?
|
||||
10000 : c->frame_size;
|
||||
|
||||
ret = av_samples_alloc_array_and_samples(&src_samples_data, &src_samples_linesize, c->channels,
|
||||
src_nb_samples, c->sample_fmt, 0);
|
||||
if (ret < 0) {
|
||||
fprintf(stderr, "Could not allocate source samples\n");
|
||||
exit(1);
|
||||
}
|
||||
|
||||
/* create resampler context */
|
||||
if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
|
||||
swr_ctx = swr_alloc();
|
||||
if (!swr_ctx) {
|
||||
fprintf(stderr, "Could not allocate resampler context\n");
|
||||
exit(1);
|
||||
}
|
||||
|
||||
/* set options */
|
||||
av_opt_set_int (swr_ctx, "in_channel_count", c->channels, 0);
|
||||
av_opt_set_int (swr_ctx, "in_sample_rate", c->sample_rate, 0);
|
||||
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
|
||||
av_opt_set_int (swr_ctx, "out_channel_count", c->channels, 0);
|
||||
av_opt_set_int (swr_ctx, "out_sample_rate", c->sample_rate, 0);
|
||||
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
|
||||
|
||||
/* initialize the resampling context */
|
||||
if ((ret = swr_init(swr_ctx)) < 0) {
|
||||
fprintf(stderr, "Failed to initialize the resampling context\n");
|
||||
exit(1);
|
||||
}
|
||||
}
|
||||
|
||||
/* 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 = src_nb_samples;
|
||||
ret = av_samples_alloc_array_and_samples(&dst_samples_data, &dst_samples_linesize, c->channels,
|
||||
max_dst_nb_samples, c->sample_fmt, 0);
|
||||
if (ret < 0) {
|
||||
fprintf(stderr, "Could not allocate destination samples\n");
|
||||
exit(1);
|
||||
}
|
||||
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, max_dst_nb_samples,
|
||||
c->sample_fmt, 0);
|
||||
}
|
||||
|
||||
/* 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, dst_nb_samples;
|
||||
|
||||
av_init_packet(&pkt);
|
||||
c = st->codec;
|
||||
|
||||
get_audio_frame((int16_t *)src_samples_data[0], src_nb_samples, c->channels);
|
||||
|
||||
/* convert samples from native format to destination codec format, using the resampler */
|
||||
if (swr_ctx) {
|
||||
/* compute destination number of samples */
|
||||
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + src_nb_samples,
|
||||
c->sample_rate, c->sample_rate, AV_ROUND_UP);
|
||||
if (dst_nb_samples > max_dst_nb_samples) {
|
||||
av_free(dst_samples_data[0]);
|
||||
ret = av_samples_alloc(dst_samples_data, &dst_samples_linesize, c->channels,
|
||||
dst_nb_samples, c->sample_fmt, 0);
|
||||
if (ret < 0)
|
||||
exit(1);
|
||||
max_dst_nb_samples = dst_nb_samples;
|
||||
dst_samples_size = av_samples_get_buffer_size(NULL, c->channels, dst_nb_samples,
|
||||
c->sample_fmt, 0);
|
||||
}
|
||||
|
||||
/* convert to destination format */
|
||||
ret = swr_convert(swr_ctx,
|
||||
dst_samples_data, dst_nb_samples,
|
||||
(const uint8_t **)src_samples_data, src_nb_samples);
|
||||
if (ret < 0) {
|
||||
fprintf(stderr, "Error while converting\n");
|
||||
exit(1);
|
||||
}
|
||||
} else {
|
||||
dst_samples_data[0] = src_samples_data[0];
|
||||
dst_nb_samples = src_nb_samples;
|
||||
}
|
||||
|
||||
frame->nb_samples = dst_nb_samples;
|
||||
avcodec_fill_audio_frame(frame, c->channels, c->sample_fmt,
|
||||
dst_samples_data[0], dst_samples_size, 0);
|
||||
|
||||
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(src_samples_data[0]);
|
||||
av_free(dst_samples_data[0]);
|
||||
}
|
||||
|
||||
/**************************************************************/
|
||||
/* 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 {
|
||||
AVPacket pkt = { 0 };
|
||||
int got_packet;
|
||||
av_init_packet(&pkt);
|
||||
|
||||
/* encode the image */
|
||||
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
|
||||
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 (!ret && got_packet && pkt.size) {
|
||||
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_time, video_time;
|
||||
int ret;
|
||||
|
||||
/* 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. */
|
||||
audio_time = audio_st ? audio_st->pts.val * av_q2d(audio_st->time_base) : 0.0;
|
||||
video_time = video_st ? video_st->pts.val * av_q2d(video_st->time_base) : 0.0;
|
||||
|
||||
if ((!audio_st || audio_time >= STREAM_DURATION) &&
|
||||
(!video_st || video_time >= STREAM_DURATION))
|
||||
break;
|
||||
|
||||
/* write interleaved audio and video frames */
|
||||
if (!video_st || (video_st && audio_st && audio_time < video_time)) {
|
||||
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);
|
||||
|
||||
if (!(fmt->flags & AVFMT_NOFILE))
|
||||
/* Close the output file. */
|
||||
avio_close(oc->pb);
|
||||
|
||||
/* free the stream */
|
||||
avformat_free_context(oc);
|
||||
|
||||
return 0;
|
||||
}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libavcodec
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libavcodec
|
||||
Description: FFmpeg codec library
|
||||
Version: 55.39.101
|
||||
Requires: libavutil = 52.48.101
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lavcodec
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libavdevice
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libavdevice
|
||||
Description: FFmpeg device handling library
|
||||
Version: 55.5.100
|
||||
Requires: libavfilter = 3.90.100, libavformat = 55.19.104
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lavdevice -lavicap32 -lpsapi -lole32 -lstrmiids -luuid -lws2_32 -liconv -lm -lz -lpsapi -ladvapi32 -lshell32
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libavfilter
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libavfilter
|
||||
Description: FFmpeg audio/video filtering library
|
||||
Version: 3.90.100
|
||||
Requires: libswresample = 0.17.104, libswscale = 2.5.101, libavformat = 55.19.104, libavcodec = 55.39.101, libavutil = 52.48.101
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lavfilter -lavicap32 -lpsapi -lole32 -lstrmiids -luuid -lws2_32 -liconv -lm -lz -lpsapi -ladvapi32 -lshell32
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libavformat
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libavformat
|
||||
Description: FFmpeg container format library
|
||||
Version: 55.19.104
|
||||
Requires: libavcodec = 55.39.101
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lavformat
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libavutil
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libavutil
|
||||
Description: FFmpeg utility library
|
||||
Version: 52.48.101
|
||||
Requires:
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lavutil
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libswresample
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libswresample
|
||||
Description: FFmpeg audio resampling library
|
||||
Version: 0.17.104
|
||||
Requires: libavutil = 52.48.101
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lswresample -lm
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,12 @@
|
||||
prefix=
|
||||
exec_prefix=
|
||||
libdir=${pcfiledir}/../../../libswscale
|
||||
includedir=${pcfiledir}/../../..
|
||||
|
||||
Name: libswscale
|
||||
Description: FFmpeg image rescaling library
|
||||
Version: 2.5.101
|
||||
Requires: libavutil = 52.48.101
|
||||
Conflicts:
|
||||
Libs: -L${libdir} -Wl,-rpath,${libdir} -lswscale
|
||||
Cflags: -I${includedir}
|
||||
@@ -0,0 +1,211 @@
|
||||
/*
|
||||
* 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 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 = av_samples_alloc_array_and_samples(&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 = av_samples_alloc_array_and_samples(&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;
|
||||
}
|
||||
@@ -0,0 +1,141 @@
|
||||
/*
|
||||
* 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 alignment */
|
||||
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;
|
||||
}
|
||||
@@ -0,0 +1,556 @@
|
||||
\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)
|
||||
|
||||
FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
|
||||
@code{concat}} protocol designed specifically for that, with examples in the
|
||||
documentation.
|
||||
|
||||
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
|
||||
video by merely concatenating the files containing 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 aresample}. 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{aformat} explicitly in the filtergraph,
|
||||
specifying the exact format.
|
||||
|
||||
@example
|
||||
aformat=sample_fmts=s16:channel_layouts=stereo
|
||||
@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 @command{ffmpeg_g}, @command{ffplay_g}, etc. which
|
||||
contain full debug information. Those binaries are stripped to create
|
||||
@command{ffmpeg}, @command{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 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.
|
||||
|
||||
@code{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 @code{r_frame_rate}
|
||||
will be 150 (it is the least common multiple).
|
||||
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
|
||||
|
||||
@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
|
||||
@@ -0,0 +1,205 @@
|
||||
\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 their 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:
|
||||
|
||||
@table @option
|
||||
@item RSA
|
||||
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
|
||||
@item ECDSA
|
||||
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
|
||||
@end table
|
||||
|
||||
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}.
|
||||
|
||||
@item GEN
|
||||
Set to @var{1} to generate the missing or mismatched references.
|
||||
@end table
|
||||
|
||||
@section Examples
|
||||
|
||||
@example
|
||||
make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
|
||||
@end example
|
||||
@@ -0,0 +1,29 @@
|
||||
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 [email protected]" # command to submit report
|
||||
comment= # optional description
|
||||
build_only= # set to "yes" for a compile-only instance that skips tests
|
||||
|
||||
# the following are optional and map to configure options
|
||||
arch=
|
||||
cpu=
|
||||
cross_prefix=
|
||||
as=
|
||||
cc=
|
||||
ld=
|
||||
target_os=
|
||||
sysroot=
|
||||
target_exec=
|
||||
target_path=
|
||||
target_samples=
|
||||
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'
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,47 @@
|
||||
:
|
||||
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 :
|
||||
:
|
||||
@@ -0,0 +1,277 @@
|
||||
\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 fftools-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{filtergraph}
|
||||
Create the filtergraph specified by @var{filtergraph} and use it to
|
||||
filter the video stream.
|
||||
|
||||
@var{filtergraph} is a description of the filtergraph to apply to
|
||||
the stream, and must have a single video input and a single video
|
||||
output. In the filtergraph, the input is associated to the label
|
||||
@code{in}, and the output to the label @code{out}. See the
|
||||
ffmpeg-filters manual for more information about the filtergraph
|
||||
syntax.
|
||||
|
||||
@item -af @var{filtergraph}
|
||||
@var{filtergraph} is a description of the filtergraph to apply to
|
||||
the input audio.
|
||||
Use the option "-filters" to show all the available filters (including
|
||||
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
|
||||
Print several playback statistics, in particular show the stream
|
||||
duration, the codec parameters, the current position in the stream and
|
||||
the audio/video synchronisation drift. It is on by default, to
|
||||
explicitly disable it you need to specify @code{-nostats}.
|
||||
|
||||
@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 in the curret program.
|
||||
|
||||
@item v
|
||||
Cycle video channel.
|
||||
|
||||
@item t
|
||||
Cycle subtitle channel in the current program.
|
||||
|
||||
@item c
|
||||
Cycle program.
|
||||
|
||||
@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
|
||||
|
||||
@include config.texi
|
||||
@ifset config-all
|
||||
@ifset config-avutil
|
||||
@include utils.texi
|
||||
@end ifset
|
||||
@ifset config-avcodec
|
||||
@include codecs.texi
|
||||
@include bitstream_filters.texi
|
||||
@end ifset
|
||||
@ifset config-avformat
|
||||
@include formats.texi
|
||||
@include protocols.texi
|
||||
@end ifset
|
||||
@ifset config-avdevice
|
||||
@include devices.texi
|
||||
@end ifset
|
||||
@ifset config-swresample
|
||||
@include resampler.texi
|
||||
@end ifset
|
||||
@ifset config-swscale
|
||||
@include scaler.texi
|
||||
@end ifset
|
||||
@ifset config-avfilter
|
||||
@include filters.texi
|
||||
@end ifset
|
||||
@end ifset
|
||||
|
||||
@chapter See Also
|
||||
|
||||
@ifhtml
|
||||
@ifset config-all
|
||||
@url{ffplay.html,ffplay},
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
@url{ffplay-all.html,ffmpeg-all},
|
||||
@end ifset
|
||||
@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.html,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
|
||||
@ifset config-all
|
||||
ffplay(1),
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
ffplay-all(1),
|
||||
@end ifset
|
||||
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
|
||||
@@ -0,0 +1,637 @@
|
||||
\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", "STREAM" or "PROGRAM_STREAM"
|
||||
section.
|
||||
|
||||
@c man end
|
||||
|
||||
@chapter Options
|
||||
@c man begin OPTIONS
|
||||
|
||||
@include fftools-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 a 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 -show_programs
|
||||
Show information about programs and their streams contained in the input
|
||||
multimedia stream.
|
||||
|
||||
Each media stream information is printed within a dedicated section
|
||||
with name "PROGRAM_STREAM".
|
||||
|
||||
@item -show_chapters
|
||||
Show information about chapters stored in the format.
|
||||
|
||||
Each chapter is printed within a dedicated section with name "CHAPTER".
|
||||
|
||||
@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 -read_intervals @var{read_intervals}
|
||||
|
||||
Read only the specified intervals. @var{read_intervals} must be a
|
||||
sequence of interval specifications separated by ",".
|
||||
@command{ffprobe} will seek to the interval starting point, and will
|
||||
continue reading from that.
|
||||
|
||||
Each interval is specified by two optional parts, separated by "%".
|
||||
|
||||
The first part specifies the interval start position. It is
|
||||
interpreted as an abolute position, or as a relative offset from the
|
||||
current position if it is preceded by the "+" character. If this first
|
||||
part is not specified, no seeking will be performed when reading this
|
||||
interval.
|
||||
|
||||
The second part specifies the interval end position. It is interpreted
|
||||
as an absolute position, or as a relative offset from the current
|
||||
position if it is preceded by the "+" character. If the offset
|
||||
specification starts with "#", it is interpreted as the number of
|
||||
packets to read (not including the flushing packets) from the interval
|
||||
start. If no second part is specified, the program will read until the
|
||||
end of the input.
|
||||
|
||||
Note that seeking is not accurate, thus the actual interval start
|
||||
point may be different from the specified position. Also, when an
|
||||
interval duration is specified, the absolute end time will be computed
|
||||
by adding the duration to the interval start point found by seeking
|
||||
the file, rather than to the specified start value.
|
||||
|
||||
The formal syntax is given by:
|
||||
@example
|
||||
@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
|
||||
@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
|
||||
@end example
|
||||
|
||||
A few examples follow.
|
||||
@itemize
|
||||
@item
|
||||
Seek to time 10, read packets until 20 seconds after the found seek
|
||||
point, then seek to position @code{01:30} (1 minute and thirty
|
||||
seconds) and read packets until position @code{01:45}.
|
||||
@example
|
||||
10%+20,01:30%01:45
|
||||
@end example
|
||||
|
||||
@item
|
||||
Read only 42 packets after seeking to position @code{01:23}:
|
||||
@example
|
||||
01:23%+#42
|
||||
@end example
|
||||
|
||||
@item
|
||||
Read only the first 20 seconds from the start:
|
||||
@example
|
||||
%+20
|
||||
@end example
|
||||
|
||||
@item
|
||||
Read from the start until position @code{02:30}:
|
||||
@example
|
||||
%02:30
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
@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, STREAM or
|
||||
PROGRAM_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
|
||||
|
||||
@include config.texi
|
||||
@ifset config-all
|
||||
@ifset config-avutil
|
||||
@include utils.texi
|
||||
@end ifset
|
||||
@ifset config-avcodec
|
||||
@include codecs.texi
|
||||
@include bitstream_filters.texi
|
||||
@end ifset
|
||||
@ifset config-avformat
|
||||
@include formats.texi
|
||||
@include protocols.texi
|
||||
@end ifset
|
||||
@ifset config-avdevice
|
||||
@include devices.texi
|
||||
@end ifset
|
||||
@ifset config-swresample
|
||||
@include resampler.texi
|
||||
@end ifset
|
||||
@ifset config-swscale
|
||||
@include scaler.texi
|
||||
@end ifset
|
||||
@ifset config-avfilter
|
||||
@include filters.texi
|
||||
@end ifset
|
||||
@end ifset
|
||||
|
||||
@chapter See Also
|
||||
|
||||
@ifhtml
|
||||
@ifset config-all
|
||||
@url{ffprobe.html,ffprobe},
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
@url{ffprobe-all.html,ffprobe-all},
|
||||
@end ifset
|
||||
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @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.html,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
|
||||
@ifset config-all
|
||||
ffprobe(1),
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
ffprobe-all(1),
|
||||
@end ifset
|
||||
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
|
||||
@@ -0,0 +1,245 @@
|
||||
<?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="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
|
||||
<xsd:element name="chapters" type="ffprobe:chaptersType" 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: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="programsType">
|
||||
<xsd:sequence>
|
||||
<xsd:element name="program" type="ffprobe:programType" 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="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
|
||||
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
|
||||
</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="channel_layout" type="xsd:string"/>
|
||||
<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="programType">
|
||||
<xsd:sequence>
|
||||
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
|
||||
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
|
||||
</xsd:sequence>
|
||||
|
||||
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="start_time" type="xsd:float"/>
|
||||
<xsd:attribute name="start_pts" type="xsd:long"/>
|
||||
<xsd:attribute name="end_time" type="xsd:float"/>
|
||||
<xsd:attribute name="end_pts" type="xsd:long"/>
|
||||
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
|
||||
</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="nb_programs" 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:attribute name="probe_score" type="xsd:int"/>
|
||||
</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="chaptersType">
|
||||
<xsd:sequence>
|
||||
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
|
||||
</xsd:sequence>
|
||||
</xsd:complexType>
|
||||
|
||||
<xsd:complexType name="chapterType">
|
||||
<xsd:sequence>
|
||||
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
|
||||
</xsd:sequence>
|
||||
|
||||
<xsd:attribute name="id" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
|
||||
<xsd:attribute name="start" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="start_time" type="xsd:float"/>
|
||||
<xsd:attribute name="end" type="xsd:int" use="required"/>
|
||||
<xsd:attribute name="end_time" type="xsd:float" 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>
|
||||
@@ -0,0 +1,371 @@
|
||||
# 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>
|
||||
@@ -0,0 +1,320 @@
|
||||
\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
|
||||
|
||||
@command{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
|
||||
@file{ffserver.conf}).
|
||||
|
||||
@command{ffserver} receives prerecorded files or FFM streams from some
|
||||
@command{ffmpeg} instance as input, then streams them over
|
||||
RTP/RTSP/HTTP.
|
||||
|
||||
An @command{ffserver} instance will listen on some port as specified
|
||||
in the configuration file. You can launch one or more instances of
|
||||
@command{ffmpeg} and send one or more FFM streams to the port where
|
||||
ffserver is expecting to receive them. Alternately, you can make
|
||||
@command{ffserver} launch such @command{ffmpeg} instances at startup.
|
||||
|
||||
Input streams are called feeds, and each one is specified by a
|
||||
@code{<Feed>} section in the configuration file.
|
||||
|
||||
For each feed you can have different output streams in various
|
||||
formats, each one specified by a @code{<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 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 it is 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 fftools-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
|
||||
|
||||
@include config.texi
|
||||
@ifset config-all
|
||||
@ifset config-avutil
|
||||
@include utils.texi
|
||||
@end ifset
|
||||
@ifset config-avcodec
|
||||
@include codecs.texi
|
||||
@include bitstream_filters.texi
|
||||
@end ifset
|
||||
@ifset config-avformat
|
||||
@include formats.texi
|
||||
@include protocols.texi
|
||||
@end ifset
|
||||
@ifset config-avdevice
|
||||
@include devices.texi
|
||||
@end ifset
|
||||
@ifset config-swresample
|
||||
@include resampler.texi
|
||||
@end ifset
|
||||
@ifset config-swscale
|
||||
@include scaler.texi
|
||||
@end ifset
|
||||
@ifset config-avfilter
|
||||
@include filters.texi
|
||||
@end ifset
|
||||
@end ifset
|
||||
|
||||
@chapter See Also
|
||||
|
||||
@ifhtml
|
||||
@ifset config-all
|
||||
@url{ffserver.html,ffserver},
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
@url{ffserver-all.html,ffserver-all},
|
||||
@end ifset
|
||||
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.html,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
|
||||
@ifset config-all
|
||||
ffserver(1),
|
||||
@end ifset
|
||||
@ifset config-not-all
|
||||
ffserver-all(1),
|
||||
@end ifset
|
||||
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
|
||||
@@ -0,0 +1,292 @@
|
||||
All the numerical options, if not specified otherwise, accept a string
|
||||
representing a number as input, which may be followed by one of the SI
|
||||
unit prefixes, for example: 'K', 'M', or 'G'.
|
||||
|
||||
If 'i' is appended to the SI unit prefix, the complete prefix will be
|
||||
interpreted as a unit prefix for binary multiplies, which are based on
|
||||
powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
|
||||
prefix multiplies the value by 8. This allows using, for example:
|
||||
'KB', 'MiB', 'G' and 'B' as number suffixes.
|
||||
|
||||
Options which do not take arguments are boolean options, and set the
|
||||
corresponding value to true. They can be set to false by prefixing
|
||||
the option name with "no". For example using "-nofoo"
|
||||
will set the boolean option with name "foo" to false.
|
||||
|
||||
@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) a given option belongs 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} contains the
|
||||
@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, so that the option is 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 following: 'v' for video, 'a' for audio, 's' for subtitle,
|
||||
'd' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
|
||||
stream number @var{stream_index} of this type. Otherwise, it matches all
|
||||
streams of this type.
|
||||
@item p:@var{program_id}[:@var{stream_index}]
|
||||
If @var{stream_index} is given, then it matches the stream with number @var{stream_index}
|
||||
in the program with the id @var{program_id}. Otherwise, it matches all streams in the
|
||||
program.
|
||||
@item #@var{stream_id}
|
||||
Matches the stream by a format-specific ID.
|
||||
@end table
|
||||
|
||||
@section Generic options
|
||||
|
||||
These options are shared amongst the ff* 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. If no argument is specified, only basic (non advanced) tool
|
||||
options are shown.
|
||||
|
||||
Possible values of @var{arg} are:
|
||||
@table @option
|
||||
@item long
|
||||
Print advanced tool options in addition to the basic tool options.
|
||||
|
||||
@item full
|
||||
Print complete list of options, including shared and private options
|
||||
for encoders, decoders, demuxers, muxers, filters, etc.
|
||||
|
||||
@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.
|
||||
|
||||
@item filter=@var{filter_name}
|
||||
Print detailed information about the filter name @var{filter_name}. Use the
|
||||
@option{-filters} option to get a list of all filters.
|
||||
@end table
|
||||
|
||||
@item -version
|
||||
Show version.
|
||||
|
||||
@item -formats
|
||||
Show available formats.
|
||||
|
||||
@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 -colors
|
||||
Show recognized color names.
|
||||
|
||||
@item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
|
||||
Set the logging level used by the library.
|
||||
Adding "repeat+" indicates that repeated log output should not be compressed
|
||||
to the first line and the "Last message repeated n times" line will be
|
||||
omitted. "repeat" can also be used alone.
|
||||
If "repeat" is used alone, and with no prior loglevel set, the default
|
||||
loglevel will be used. If multiple loglevel parameters are given, using
|
||||
'repeat' will not change the loglevel.
|
||||
@var{loglevel} is a number or a string containing one of the following values:
|
||||
@table @samp
|
||||
@item quiet
|
||||
Show nothing at all; be silent.
|
||||
@item panic
|
||||
Only show fatal errors which could lead the process to crash, such as
|
||||
and assert failure. This is not currently used for anything.
|
||||
@item fatal
|
||||
Only show fatal errors. These are errors after which the process absolutely
|
||||
cannot continue after.
|
||||
@item error
|
||||
Show all errors, including ones which can be recovered from.
|
||||
@item warning
|
||||
Show all warnings and errors. Any message related to possibly
|
||||
incorrect or unexpected events will be shown.
|
||||
@item info
|
||||
Show informative messages during processing. This is in addition to
|
||||
warnings and errors. This is the default value.
|
||||
@item verbose
|
||||
Same as @code{info}, except more verbose.
|
||||
@item debug
|
||||
Show everything, including debugging information.
|
||||
@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
|
||||
Possible flags for this option are:
|
||||
@table @samp
|
||||
@item x86
|
||||
@table @samp
|
||||
@item mmx
|
||||
@item mmxext
|
||||
@item sse
|
||||
@item sse2
|
||||
@item sse2slow
|
||||
@item sse3
|
||||
@item sse3slow
|
||||
@item ssse3
|
||||
@item atom
|
||||
@item sse4.1
|
||||
@item sse4.2
|
||||
@item avx
|
||||
@item xop
|
||||
@item fma4
|
||||
@item 3dnow
|
||||
@item 3dnowext
|
||||
@item cmov
|
||||
@end table
|
||||
@item ARM
|
||||
@table @samp
|
||||
@item armv5te
|
||||
@item armv6
|
||||
@item armv6t2
|
||||
@item vfp
|
||||
@item vfpv3
|
||||
@item neon
|
||||
@end table
|
||||
@item PowerPC
|
||||
@table @samp
|
||||
@item altivec
|
||||
@end table
|
||||
@item Specific Processors
|
||||
@table @samp
|
||||
@item pentium2
|
||||
@item pentium3
|
||||
@item pentium4
|
||||
@item k6
|
||||
@item k62
|
||||
@item athlon
|
||||
@item athlonxp
|
||||
@item k8
|
||||
@end table
|
||||
@end table
|
||||
|
||||
@item -opencl_options options (@emph{global})
|
||||
Set OpenCL environment options. This option is only available when
|
||||
FFmpeg has been compiled with @code{--enable-opencl}.
|
||||
|
||||
@var{options} must be a list of @var{key}=@var{value} option pairs
|
||||
separated by ':'. See the ``OpenCL Options'' section in the
|
||||
ffmpeg-utils manual for the list of supported options.
|
||||
@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 per-stream, and thus a stream specifier
|
||||
should be attached to them.
|
||||
|
||||
Note: the @option{-nooption} syntax cannot be used for boolean
|
||||
AVOptions, use @option{-option 0}/@option{-option 1}.
|
||||
|
||||
Note: the 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.
|
||||
@@ -0,0 +1,270 @@
|
||||
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, sample format (the sample packing is implied by the sample
|
||||
format) and sample rate.
|
||||
|
||||
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 end 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.
|
||||
|
||||
query_formats can leave some formats unset and return AVERROR(EAGAIN) to
|
||||
cause the negotiation mechanism to try again later. That can be used by
|
||||
filters with complex requirements to use the format negotiated on one link
|
||||
to set the formats supported on another.
|
||||
|
||||
|
||||
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 filter_frame method (or its start_frame
|
||||
deprecated version) belongs 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_filter_frame (or the deprecated
|
||||
ff_start_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. The
|
||||
cur_buf and out_buf were used with the deprecated
|
||||
start_frame/draw_slice/end_frame API and should no longer be used.
|
||||
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 that implement the deprecated
|
||||
start_frame/draw_slice/end_frame API, 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 must request a buffer
|
||||
with the WRITE permission.
|
||||
|
||||
* Filters that intend to keep a reference after the filtering process
|
||||
is finished (after 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.
|
||||
|
||||
filter_frame
|
||||
------------
|
||||
|
||||
This method is called when a frame is pushed to the filter's input. It
|
||||
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: frame rate-doubling filter: filter_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 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 = ff_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 filter_frame method will be called and do the work.
|
||||
|
||||
Legacy API
|
||||
==========
|
||||
|
||||
Until libavfilter 3.23, the filter_frame method was split:
|
||||
|
||||
- for video filters, it was made of start_frame, draw_slice (that could be
|
||||
called several times on distinct parts of the frame) and end_frame;
|
||||
|
||||
- for audio filters, it was called filter_samples.
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,188 @@
|
||||
@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 seek2any @var{integer} (@emph{input})
|
||||
Allow seeking to non-keyframes on demuxer level when supported if set to 1.
|
||||
Default is 0.
|
||||
|
||||
@item analyzeduration @var{integer} (@emph{input})
|
||||
Specify how many microseconds are analyzed to probe the input. A
|
||||
higher value will allow to detect more accurate information, but will
|
||||
increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
|
||||
|
||||
@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 non-negative. A value of 1 enables shifting,
|
||||
a value of 0 disables it, the default value of -1 enables shifting
|
||||
when required by the target format.
|
||||
|
||||
When shifting is enabled, all output timestamps are shifted by the
|
||||
same amount. Audio, video, and subtitles desynching and relative
|
||||
timestamp differences are preserved compared to how they would have
|
||||
been without shifting.
|
||||
|
||||
Also note that this affects only leading negative timestamps, and not
|
||||
non-monotonic negative timestamps.
|
||||
|
||||
@item skip_initial_bytes @var{integer} (@emph{input})
|
||||
Set number of bytes to skip before reading header and frames if set to 1.
|
||||
Default is 0.
|
||||
|
||||
@item correct_ts_overflow @var{integer} (@emph{input})
|
||||
Correct single timestamp overflows if set to 1. Default is 1.
|
||||
|
||||
@item flush_packets @var{integer} (@emph{output})
|
||||
Flush the underlying I/O stream after each packet. Default 1 enables it, and
|
||||
has the effect of reducing the latency; 0 disables it and may slightly
|
||||
increase performance in some cases.
|
||||
@end table
|
||||
|
||||
@c man end FORMAT OPTIONS
|
||||
|
||||
@anchor{Format stream specifiers}
|
||||
@section Format stream specifiers
|
||||
|
||||
Format stream specifiers allow selection of one or more streams that
|
||||
match specific properties.
|
||||
|
||||
Possible forms of stream specifiers are:
|
||||
@table @option
|
||||
@item @var{stream_index}
|
||||
Matches the stream with this index.
|
||||
|
||||
@item @var{stream_type}[:@var{stream_index}]
|
||||
@var{stream_type} is one of following: 'v' for video, 'a' for audio,
|
||||
's' for subtitle, 'd' for data, and 't' for attachments. If
|
||||
@var{stream_index} is given, then it matches the stream number
|
||||
@var{stream_index} of this type. Otherwise, it matches all streams of
|
||||
this type.
|
||||
|
||||
@item p:@var{program_id}[:@var{stream_index}]
|
||||
If @var{stream_index} is given, then it matches the stream with number
|
||||
@var{stream_index} in the program with the id
|
||||
@var{program_id}. Otherwise, it matches all streams in the program.
|
||||
|
||||
@item #@var{stream_id}
|
||||
Matches the stream by a format-specific ID.
|
||||
@end table
|
||||
|
||||
The exact semantics of stream specifiers is defined by the
|
||||
@code{avformat_match_stream_specifier()} function declared in the
|
||||
@file{libavformat/avformat.h} header.
|
||||
|
||||
@include demuxers.texi
|
||||
@include muxers.texi
|
||||
@include metadata.texi
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,415 @@
|
||||
\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.
|
||||
@@ -0,0 +1,273 @@
|
||||
|
||||
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 [email protected]: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 [email protected]
|
||||
(--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.
|
||||
@@ -0,0 +1,763 @@
|
||||
@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 frame rate 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 output device you need to configure FFmpeg with @code{--enable-libpulse}.
|
||||
|
||||
The filename to provide to the input device is a source device or the
|
||||
string "default"
|
||||
|
||||
To list the PulseAudio source devices and their properties you can invoke
|
||||
the command @command{pactl list sources}.
|
||||
|
||||
More information about PulseAudio can be found on @url{http://www.pulseaudio.org}.
|
||||
|
||||
@subsection Options
|
||||
@table @option
|
||||
@item server
|
||||
Connect to a specific PulseAudio server, specified by an IP address.
|
||||
Default server is used when not provided.
|
||||
|
||||
@item name
|
||||
Specify the application name PulseAudio will use when showing active clients,
|
||||
by default it is the @code{LIBAVFORMAT_IDENT} string.
|
||||
|
||||
@item stream_name
|
||||
Specify the stream name PulseAudio will use when showing active streams,
|
||||
by default it is "record".
|
||||
|
||||
@item sample_rate
|
||||
Specify the samplerate in Hz, by default 48kHz is used.
|
||||
|
||||
@item channels
|
||||
Specify the channels in use, by default 2 (stereo) is set.
|
||||
|
||||
@item frame_size
|
||||
Specify the number of bytes per frame, by default it is set to 1024.
|
||||
|
||||
@item fragment_size
|
||||
Specify the minimal buffering fragment in PulseAudio, it will affect the
|
||||
audio latency. By default it is unset.
|
||||
@end table
|
||||
|
||||
@subsection Examples
|
||||
Record a stream from default device:
|
||||
@example
|
||||
ffmpeg -f pulse -i default /tmp/pulse.wav
|
||||
@end example
|
||||
|
||||
@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, v4l2
|
||||
|
||||
Video4Linux2 input video device.
|
||||
|
||||
"v4l2" can be used as alias for "video4linux2".
|
||||
|
||||
If FFmpeg is built with v4l-utils support (by using the
|
||||
@code{--enable-libv4l2} configure option), it is possible to use it with the
|
||||
@code{-use_libv4l2} input device option.
|
||||
|
||||
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 frame rates. You can check which are
|
||||
supported using @command{-list_formats all} for Video4Linux2 devices.
|
||||
Some devices, like TV cards, support one or more standards. It is possible
|
||||
to list all the supported standards using @command{-list_standards all}.
|
||||
|
||||
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.
|
||||
|
||||
Some usage examples of the video4linux2 device with @command{ffmpeg}
|
||||
and @command{ffplay}:
|
||||
@itemize
|
||||
@item
|
||||
Grab and show the input of a video4linux2 device:
|
||||
@example
|
||||
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
|
||||
@end example
|
||||
|
||||
@item
|
||||
Grab and record the input of a video4linux2 device, leave the
|
||||
frame rate and size as previously set:
|
||||
@example
|
||||
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
For more information about Video4Linux, check @url{http://linuxtv.org/}.
|
||||
|
||||
@subsection Options
|
||||
|
||||
@table @option
|
||||
@item standard
|
||||
Set the standard. Must be the name of a supported standard. To get a
|
||||
list of the supported standards, use the @option{list_standards}
|
||||
option.
|
||||
|
||||
@item channel
|
||||
Set the input channel number. Default to -1, which means using the
|
||||
previously selected channel.
|
||||
|
||||
@item video_size
|
||||
Set the video frame size. The argument must be a string in the form
|
||||
@var{WIDTH}x@var{HEIGHT} or a valid size abbreviation.
|
||||
|
||||
@item pixel_format
|
||||
Select the pixel format (only valid for raw video input).
|
||||
|
||||
@item input_format
|
||||
Set the preferred pixel format (for raw video) or a codec name.
|
||||
This option allows to select the input format, when several are
|
||||
available.
|
||||
|
||||
@item framerate
|
||||
Set the preferred video frame rate.
|
||||
|
||||
@item list_formats
|
||||
List available formats (supported pixel formats, codecs, and frame
|
||||
sizes) and exit.
|
||||
|
||||
Available values are:
|
||||
@table @samp
|
||||
@item all
|
||||
Show all available (compressed and non-compressed) formats.
|
||||
|
||||
@item raw
|
||||
Show only raw video (non-compressed) formats.
|
||||
|
||||
@item compressed
|
||||
Show only compressed formats.
|
||||
@end table
|
||||
|
||||
@item list_standards
|
||||
List supported standards and exit.
|
||||
|
||||
Available values are:
|
||||
@table @samp
|
||||
@item all
|
||||
Show all supported standards.
|
||||
@end table
|
||||
|
||||
@item timestamps, ts
|
||||
Set type of timestamps for grabbed frames.
|
||||
|
||||
Available values are:
|
||||
@table @samp
|
||||
@item default
|
||||
Use timestamps from the kernel.
|
||||
|
||||
@item abs
|
||||
Use absolute timestamps (wall clock).
|
||||
|
||||
@item mono2abs
|
||||
Force conversion from monotonic to absolute timestamps.
|
||||
@end table
|
||||
|
||||
Default value is @code{default}.
|
||||
@end table
|
||||
|
||||
@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 -framerate 25 -video_size cif -i :0.0 out.mpg
|
||||
@end example
|
||||
|
||||
Grab at position @code{10,20}:
|
||||
@example
|
||||
ffmpeg -f x11grab -framerate 25 -video_size 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 -framerate 25 -video_size 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 -framerate 25 -video_size cif -i :0.0 out.mpg
|
||||
@end example
|
||||
|
||||
@item framerate
|
||||
Set the grabbing frame rate. Default value is @code{ntsc},
|
||||
corresponding to a frame rate 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 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
|
||||
@end example
|
||||
|
||||
With @var{follow_mouse}:
|
||||
@example
|
||||
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size 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
|
||||
@@ -0,0 +1,194 @@
|
||||
FFmpeg's bug/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, 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)://trac.ffmpeg.org
|
||||
|
||||
Type:
|
||||
-----
|
||||
art
|
||||
Artwork such as photos, music, banners, and logos.
|
||||
|
||||
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
|
||||
|
||||
sponsoring request
|
||||
Developer requests for hardware, software, specifications, money,
|
||||
refunds, etc.
|
||||
|
||||
Priority:
|
||||
---------
|
||||
critical
|
||||
Bugs about data loss and security issues.
|
||||
No feature request can be critical.
|
||||
|
||||
important
|
||||
Bugs which make FFmpeg unusable for a significant number of users.
|
||||
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 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:
|
||||
----------
|
||||
*/new
|
||||
Initial state of new bugs and feature requests submitted by
|
||||
users.
|
||||
|
||||
*/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 or feature requests which are 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/wontfix
|
||||
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.
|
||||
|
||||
feature_request/closed/fixed
|
||||
Feature requests which have been implemented.
|
||||
|
||||
feature_request/closed/wontfix
|
||||
Feature requests which will not be implemented. The reasons here could
|
||||
be legal, philosophical or others.
|
||||
|
||||
Note2, if you provide the requested info do not forget to remove the
|
||||
needs_more_info resolution.
|
||||
|
||||
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
|
||||
@@ -0,0 +1,48 @@
|
||||
\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
|
||||
@@ -0,0 +1,45 @@
|
||||
\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
|
||||
@@ -0,0 +1,44 @@
|
||||
\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
|
||||
@@ -0,0 +1,48 @@
|
||||
\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
|
||||
@@ -0,0 +1,44 @@
|
||||
\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
|
||||
@@ -0,0 +1,70 @@
|
||||
\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 a 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
|
||||
@@ -0,0 +1,63 @@
|
||||
\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
|
||||
@@ -0,0 +1,84 @@
|
||||
@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
|
||||
|
||||
By using the ffmetadata muxer and demuxer it is possible to extract
|
||||
metadata from an input file to an ffmetadata file, and then transcode
|
||||
the file into an output file with the edited ffmetadata file.
|
||||
|
||||
Extracting an ffmetadata file with @file{ffmpeg} goes as follows:
|
||||
@example
|
||||
ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
|
||||
@end example
|
||||
|
||||
Reinserting edited metadata information from the FFMETADATAFILE file can
|
||||
be done as:
|
||||
@example
|
||||
ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
|
||||
@end example
|
||||
|
||||
@c man end METADATA
|
||||
@@ -0,0 +1,75 @@
|
||||
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/
|
||||
fft_fixed_32.c
|
||||
fft_init_table.c
|
||||
fft_table.h
|
||||
mdct_fixed_32.c
|
||||
* libavcodec/mips/
|
||||
aaccoder_mips.c
|
||||
aacpsy_mips.h
|
||||
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
|
||||
iirfilter_mips.c
|
||||
mpegaudiodsp_mips_fixed.c
|
||||
mpegaudiodsp_mips_float.c
|
||||
@@ -0,0 +1,70 @@
|
||||
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.
|
||||
|
||||
If there are inter-frame dependencies, so the codec calls
|
||||
ff_thread_report/await_progress(), set AVCodecInternal.allocate_progress. The
|
||||
frames must then be freed with ff_thread_release_buffer().
|
||||
Otherwise leave it at zero and decode directly into the user-supplied frames.
|
||||
|
||||
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().
|
||||
@@ -0,0 +1,908 @@
|
||||
@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{aiff}
|
||||
@section aiff
|
||||
|
||||
Audio Interchange File Format muxer.
|
||||
|
||||
It accepts the following options:
|
||||
|
||||
@table @option
|
||||
@item write_id3v2
|
||||
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
|
||||
|
||||
@item id3v2_version
|
||||
Select ID3v2 version to write. Currently only version 3 and 4 (aka.
|
||||
ID3v2.3 and ID3v2.4) are supported. The default is version 4.
|
||||
|
||||
@end table
|
||||
|
||||
@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}. Default value is 1. Must be a
|
||||
non-negative number.
|
||||
|
||||
@item -update @var{number}
|
||||
If @var{number} is nonzero, the filename will always be interpreted as just a
|
||||
filename, not a pattern, and this file will be continuously overwritten with new
|
||||
images.
|
||||
|
||||
@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.
|
||||
|
||||
@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
|
||||
|
||||
This muxer supports the following options:
|
||||
|
||||
@table @option
|
||||
|
||||
@item reserve_index_space
|
||||
By default, this muxer writes the index for seeking (called cues in Matroska
|
||||
terms) at the end of the file, because it cannot know in advance how much space
|
||||
to leave for the index at the beginning of the file. However for some use cases
|
||||
-- e.g. streaming where seeking is possible but slow -- it is useful to put the
|
||||
index at the beginning of the file.
|
||||
|
||||
If this option is set to a non-zero value, the muxer will reserve a given amount
|
||||
of space in the file header and then try to write the cues there when the muxing
|
||||
finishes. If the available space does not suffice, muxing will fail. A safe size
|
||||
for most use cases should be about 50kB per hour of video.
|
||||
|
||||
Note that cues are only written if the output is seekable and this option will
|
||||
have no effect if it is not.
|
||||
|
||||
@end table
|
||||
|
||||
@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 index (moov atom) to the beginning 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.
|
||||
@item -movflags rtphint
|
||||
Add RTP hinting tracks to the output file.
|
||||
@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 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
|
||||
|
||||
@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).
|
||||
@item -mpegts_m2ts_mode @var{number}
|
||||
Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
|
||||
@item -muxrate @var{number}
|
||||
Set muxrate.
|
||||
@item -pes_payload_size @var{number}
|
||||
Set minimum PES packet payload in bytes.
|
||||
@item -mpegts_flags @var{flags}
|
||||
Set flags (see below).
|
||||
@item -mpegts_copyts @var{number}
|
||||
Preserve original timestamps, if value is set to 1. Default value is -1, which
|
||||
results in shifting timestamps so that they start from 0.
|
||||
@item -tables_version @var{number}
|
||||
Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
|
||||
This option allows updating stream structure so that standard consumer may
|
||||
detect the change. To do so, reopen output AVFormatContext (in case of API
|
||||
usage) or restart ffmpeg instance, cyclically changing tables_version value:
|
||||
@example
|
||||
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
|
||||
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
|
||||
...
|
||||
ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
|
||||
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
|
||||
ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
|
||||
...
|
||||
@end example
|
||||
@end table
|
||||
|
||||
Option mpegts_flags may take a set of such flags:
|
||||
|
||||
@table @option
|
||||
@item resend_headers
|
||||
Reemit PAT/PMT before writing the next packet.
|
||||
@item latm
|
||||
Use LATM packetization for AAC.
|
||||
@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 ogg
|
||||
|
||||
Ogg container muxer.
|
||||
|
||||
@table @option
|
||||
@item -page_duration @var{duration}
|
||||
Preferred page duration, in microseconds. The muxer will attempt to create
|
||||
pages that are approximately @var{duration} microseconds long. This allows the
|
||||
user to compromise between seek granularity and container overhead. The default
|
||||
is 1 second. A value of 0 will fill all segments, making pages as large as
|
||||
possible. A value of 1 will effectively use 1 packet-per-page in most
|
||||
situations, giving a small seek granularity at the cost of additional container
|
||||
overhead.
|
||||
@end table
|
||||
|
||||
@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 @code{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 @samp
|
||||
@item cache
|
||||
Allow caching (only affects M3U8 list files).
|
||||
|
||||
@item live
|
||||
Allow live-friendly file generation.
|
||||
@end table
|
||||
|
||||
Default value is @code{samp}.
|
||||
|
||||
@item segment_list_size @var{size}
|
||||
Update the list file so that it contains at most the last @var{size}
|
||||
segments. If 0 the list file will contain all the segments. Default
|
||||
value is 0.
|
||||
|
||||
@item segment_list_type @var{type}
|
||||
Specify the format for the segment list file.
|
||||
|
||||
The following values are recognized:
|
||||
@table @samp
|
||||
@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.
|
||||
|
||||
@samp{ext} is deprecated in favor or @samp{csv}.
|
||||
|
||||
@item ffconcat
|
||||
Generate an ffconcat file for the created segments. The resulting file
|
||||
can be read using the FFmpeg @ref{concat} demuxer.
|
||||
|
||||
A list file with the suffix @code{".ffcat"} or @code{".ffconcat"} will
|
||||
auto-select this format.
|
||||
|
||||
@item m3u8
|
||||
Generate an extended M3U8 file, version 3, compliant with
|
||||
@url{http://tools.ietf.org/id/draft-pantos-http-live-streaming}.
|
||||
|
||||
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}, the value must be a duration
|
||||
specification. Default value is "2". See also the
|
||||
@option{segment_times} option.
|
||||
|
||||
Note that splitting may not be accurate, unless you force the
|
||||
reference stream key-frames at the given time. See the introductory
|
||||
notice and the examples below.
|
||||
|
||||
@item segment_time_delta @var{delta}
|
||||
Specify the accuracy time when selecting the start time for a
|
||||
segment, expressed as a duration specification. 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. See also
|
||||
the @option{segment_time} option.
|
||||
|
||||
@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.
|
||||
|
||||
@item initial_offset @var{offset}
|
||||
Specify timestamp offset to apply to the output packet timestamps. The
|
||||
argument must be a time duration specification, and defaults to 0.
|
||||
@end table
|
||||
|
||||
@subsection 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 @command{ffmpeg} @option{force_key_frames}
|
||||
option to force key frames in the input at the specified location, together
|
||||
with the segment option @option{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 @option{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 tee
|
||||
|
||||
The tee muxer can be used to write the same data to several files or any
|
||||
other kind of muxer. It can be used, for example, to both stream a video to
|
||||
the network and save it to disk at the same time.
|
||||
|
||||
It is different from specifying several outputs to the @command{ffmpeg}
|
||||
command-line tool because the audio and video data will be encoded only once
|
||||
with the tee muxer; encoding can be a very expensive process. It is not
|
||||
useful when using the libavformat API directly because it is then possible
|
||||
to feed the same packets to several muxers directly.
|
||||
|
||||
The slave outputs are specified in the file name given to the muxer,
|
||||
separated by '|'. If any of the slave name contains the '|' separator,
|
||||
leading or trailing spaces or any special character, it must be
|
||||
escaped (see the ``Quoting and escaping'' section in the ffmpeg-utils
|
||||
manual).
|
||||
|
||||
Muxer options can be specified for each slave by prepending them as a list of
|
||||
@var{key}=@var{value} pairs separated by ':', between square brackets. If
|
||||
the options values contain a special character or the ':' separator, they
|
||||
must be escaped; note that this is a second level escaping.
|
||||
|
||||
The following special options are also recognized:
|
||||
@table @option
|
||||
@item f
|
||||
Specify the format name. Useful if it cannot be guessed from the
|
||||
output name suffix.
|
||||
|
||||
@item bsfs[/@var{spec}]
|
||||
Specify a list of bitstream filters to apply to the specified
|
||||
output. It is possible to specify to which streams a given bitstream
|
||||
filter applies, by appending a stream specifier to the option
|
||||
separated by @code{/}. If the stream specifier is not specified, the
|
||||
bistream filters will be applied to all streams in the output.
|
||||
|
||||
Several bitstream filters can be specified, separated by ",".
|
||||
|
||||
@item select
|
||||
Select the streams that should be mapped to the slave output,
|
||||
specified by a stream specifier. If not specified, this defaults to
|
||||
all the input streams.
|
||||
@end table
|
||||
|
||||
Some examples follow.
|
||||
@itemize
|
||||
@item
|
||||
Encode something and both archive it in a WebM file and stream it
|
||||
as MPEG-TS over UDP (the streams need to be explicitly mapped):
|
||||
@example
|
||||
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
|
||||
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
|
||||
@end example
|
||||
|
||||
@item
|
||||
Use @command{ffmpeg} to encode the input, and send the output
|
||||
to three different destinations. The @code{dump_extra} bitstream
|
||||
filter is used to add extradata information to all the output video
|
||||
keyframes packets, as requested by the MPEG-TS format. The select
|
||||
option is applied to @file{out.aac} in order to make it contain only
|
||||
audio packets.
|
||||
@example
|
||||
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
|
||||
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
Note: some codecs may need different options depending on the output format;
|
||||
the auto-detection of this can not work with the tee muxer. The main example
|
||||
is the @option{global_header} flag.
|
||||
|
||||
@c man end MUXERS
|
||||
@@ -0,0 +1,138 @@
|
||||
\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
|
||||
|
||||
@@ -0,0 +1,288 @@
|
||||
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 int16_t *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
|
||||
@@ -0,0 +1,289 @@
|
||||
@chapter Output Devices
|
||||
@c man begin OUTPUT DEVICES
|
||||
|
||||
Output devices are configured elements in FFmpeg that can 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 device 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 fbdev
|
||||
|
||||
Linux framebuffer output 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
|
||||
@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
|
||||
|
||||
@subsection Options
|
||||
@table @option
|
||||
|
||||
@item xoffset
|
||||
@item yoffset
|
||||
Set x/y coordinate of top left corner. Default is 0.
|
||||
@end table
|
||||
|
||||
@subsection Examples
|
||||
Play a file on framebuffer device @file{/dev/fb0}.
|
||||
Required pixel format depends on current framebuffer settings.
|
||||
@example
|
||||
ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0
|
||||
@end example
|
||||
|
||||
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
|
||||
|
||||
@section oss
|
||||
|
||||
OSS (Open Sound System) output device.
|
||||
|
||||
@section pulse
|
||||
|
||||
PulseAudio output device.
|
||||
|
||||
To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
|
||||
|
||||
More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
|
||||
|
||||
@subsection Options
|
||||
@table @option
|
||||
|
||||
@item server
|
||||
Connect to a specific PulseAudio server, specified by an IP address.
|
||||
Default server is used when not provided.
|
||||
|
||||
@item name
|
||||
Specify the application name PulseAudio will use when showing active clients,
|
||||
by default it is the @code{LIBAVFORMAT_IDENT} string.
|
||||
|
||||
@item stream_name
|
||||
Specify the stream name PulseAudio will use when showing active streams,
|
||||
by default it is set to the specified output name.
|
||||
|
||||
@item device
|
||||
Specify the device to use. Default device is used when not provided.
|
||||
List of output devices can be obtained with command @command{pactl list sinks}.
|
||||
|
||||
@end table
|
||||
|
||||
@subsection Examples
|
||||
Play a file on default device on default server:
|
||||
@example
|
||||
ffmpeg -i INPUT -f pulse "stream name"
|
||||
@end example
|
||||
|
||||
@section sdl
|
||||
|
||||
SDL (Simple DirectMedia Layer) output device.
|
||||
|
||||
This output device 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.
|
||||
|
||||
@item window_fullscreen
|
||||
Set fullscreen mode when non-zero value is provided.
|
||||
Zero is a default.
|
||||
@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.
|
||||
|
||||
@section xv
|
||||
|
||||
XV (XVideo) output device.
|
||||
|
||||
This output device allows to show a video stream in a X Window System
|
||||
window.
|
||||
|
||||
@subsection Options
|
||||
|
||||
@table @option
|
||||
@item display_name
|
||||
Specify the hardware display name, which determines the display and
|
||||
communications domain to be used.
|
||||
|
||||
The display name or DISPLAY environment variable can be a string in
|
||||
the format @var{hostname}[:@var{number}[.@var{screen_number}]].
|
||||
|
||||
@var{hostname} specifies the name of the host machine on which the
|
||||
display is physically attached. @var{number} specifies the number of
|
||||
the display server on that host machine. @var{screen_number} specifies
|
||||
the screen to be used on that server.
|
||||
|
||||
If unspecified, it defaults to the value of the DISPLAY environment
|
||||
variable.
|
||||
|
||||
For example, @code{dual-headed:0.1} would specify screen 1 of display
|
||||
0 on the machine named ``dual-headed''.
|
||||
|
||||
Check the X11 specification for more detailed information about the
|
||||
display name format.
|
||||
|
||||
@item window_size
|
||||
Set the created 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 window_x
|
||||
@item window_y
|
||||
Set the X and Y window offsets for the created window. They are both
|
||||
set to 0 by default. The values may be ignored by the window manager.
|
||||
|
||||
@item window_title
|
||||
Set the window title, if not specified default to the filename
|
||||
specified for the output device.
|
||||
@end table
|
||||
|
||||
For more information about XVideo see @url{http://www.x.org/}.
|
||||
|
||||
@subsection Examples
|
||||
|
||||
@itemize
|
||||
@item
|
||||
Decode, display and encode video input with @command{ffmpeg} at the
|
||||
same time:
|
||||
@example
|
||||
ffmpeg -i INPUT OUTPUT -f xv display
|
||||
@end example
|
||||
|
||||
@item
|
||||
Decode and display the input video to multiple X11 windows:
|
||||
@example
|
||||
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
|
||||
@end example
|
||||
@end itemize
|
||||
|
||||
@c man end OUTPUT DEVICES
|
||||
@@ -0,0 +1,370 @@
|
||||
\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++ or Intel C++ Compiler for Windows
|
||||
|
||||
FFmpeg can be built with MSVC or ICL using a C99-to-C89 conversion utility and
|
||||
wrapper. For ICL, only the wrapper is used, since ICL supports C99.
|
||||
|
||||
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 environment in MSYS, you need to run @code{msys.bat} from
|
||||
the Visual Studio or Intel Compiler 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 the compiler 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
|
||||
For MSVC:
|
||||
./configure --toolchain=msvc
|
||||
|
||||
For ICL:
|
||||
./configure --toolchain=icl
|
||||
|
||||
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 and ICL handle 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 the following on i686 and x86_64:
|
||||
@itemize
|
||||
@item Visual Studio 2010 Pro and Express
|
||||
@item Visual Studio 2012 Pro and Express
|
||||
@item Intel Composer XE 2013
|
||||
@end itemize
|
||||
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.
|
||||
|
||||
You will 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
|
||||
@@ -0,0 +1,126 @@
|
||||
/*
|
||||
* Copyright (c) 2012 Anton Khirnov
|
||||
*
|
||||
* This file is part of FFmpeg.
|
||||
*
|
||||
* FFmpeg 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.
|
||||
*
|
||||
* FFmpeg 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 FFmpeg; 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 "libavformat/options_table.h"
|
||||
#include "libavcodec/avcodec.h"
|
||||
#include "libavcodec/options_table.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)
|
||||
{
|
||||
if (!(o->flags & (AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_ENCODING_PARAM)))
|
||||
return;
|
||||
|
||||
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)
|
||||
{
|
||||
printf("@section Format AVOptions\n");
|
||||
show_opts(avformat_options, 0);
|
||||
}
|
||||
|
||||
static void show_codec_opts(void)
|
||||
{
|
||||
printf("@section Codec AVOptions\n");
|
||||
show_opts(avcodec_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;
|
||||
}
|
||||
File diff suppressed because it is too large
Load Diff
@@ -0,0 +1,61 @@
|
||||
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
|
||||
@@ -0,0 +1,232 @@
|
||||
@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 input 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}.
|
||||
This will automatically be chosen when it is not explicitly set.
|
||||
|
||||
@item icl, in_channel_layout
|
||||
@item ocl, out_channel_layout
|
||||
Set the input/output channel layout.
|
||||
|
||||
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
|
||||
for the required syntax.
|
||||
|
||||
@item clev, center_mix_level
|
||||
Set the center mix level. It is a value expressed in deciBel, and must be
|
||||
in the interval [-32,32].
|
||||
|
||||
@item slev, surround_mix_level
|
||||
Set the surround mix level. It is a value expressed in deciBel, and must
|
||||
be in the interval [-32,32].
|
||||
|
||||
@item lfe_mix_level
|
||||
Set LFE mix into non LFE level. It is used when there is a LFE input but no
|
||||
LFE output. It is a value expressed in deciBel, and must
|
||||
be in the interval [-32,32].
|
||||
|
||||
@item rmvol, rematrix_volume
|
||||
Set rematrix volume. Default value is 1.0.
|
||||
|
||||
@item rematrix_maxval
|
||||
Set maximum output value for rematrixing.
|
||||
This can be used to prevent clipping vs. preventing volumn reduction
|
||||
A value of 1.0 prevents cliping.
|
||||
|
||||
@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, this flag forces resampling to be used even when the
|
||||
input and output sample rates match.
|
||||
@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
|
||||
@item lipshitz
|
||||
select lipshitz noise shaping dither
|
||||
@item shibata
|
||||
select shibata noise shaping dither
|
||||
@item low_shibata
|
||||
select low shibata noise shaping dither
|
||||
@item high_shibata
|
||||
select high shibata noise shaping dither
|
||||
@item f_weighted
|
||||
select f-weighted noise shaping dither
|
||||
@item modified_e_weighted
|
||||
select modified-e-weighted noise shaping dither
|
||||
@item improved_e_weighted
|
||||
select improved-e-weighted noise shaping dither
|
||||
|
||||
@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, and must be in
|
||||
the interval [0,30].
|
||||
|
||||
@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 first_pts
|
||||
For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
|
||||
This allows for padding/trimming at the start of stream. By default, no
|
||||
assumption is made about the first frame's expected pts, so no padding or
|
||||
trimming is done. For example, this could be set to 0 to pad the beginning with
|
||||
silence if an audio stream starts after the video stream or to trim any samples
|
||||
with a negative pts due to encoder delay.
|
||||
|
||||
@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 in the
|
||||
interval [2,16], default value is 9.
|
||||
|
||||
@item output_sample_bits
|
||||
For swr only, set number of used output sample bits for dithering. Must be an integer in the
|
||||
interval [0,64], default value is 0, which means it's not used.
|
||||
|
||||
@end table
|
||||
|
||||
@c man end RESAMPLER OPTIONS
|
||||
@@ -0,0 +1,119 @@
|
||||
@anchor{scaler_options}
|
||||
@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
|
||||
|
||||
@anchor{sws_flags}
|
||||
@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.
|
||||
|
||||
@item sws_dither
|
||||
Set the dithering algorithm. Accepts one of the following
|
||||
values. Default value is @samp{auto}.
|
||||
|
||||
@table @samp
|
||||
@item auto
|
||||
automatic choice
|
||||
|
||||
@item none
|
||||
no dithering
|
||||
|
||||
@item bayer
|
||||
bayer dither
|
||||
|
||||
@item ed
|
||||
error diffusion dither
|
||||
@end table
|
||||
|
||||
@end table
|
||||
|
||||
@c man end SCALER OPTIONS
|
||||
@@ -0,0 +1,638 @@
|
||||
=============================================
|
||||
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
|
||||
if (nb_planes > 2) {
|
||||
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<nb_plane_types; 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<nb_plane_types; 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)
|
||||
if (nb_planes > 2) {
|
||||
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
|
||||
|
||||
nb_plane_types = gray ? 1 : 2;
|
||||
|
||||
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 unspecified YcbCr
|
||||
1 Gray
|
||||
2 Gray + Alpha
|
||||
3 GBR
|
||||
4 GBRA
|
||||
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
|
||||
@@ -0,0 +1,24 @@
|
||||
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
|
||||
@@ -0,0 +1,46 @@
|
||||
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 it is not needed.
|
||||
Either Resampling and Rematrixing can be performed first depending on which
|
||||
way it is 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.
|
||||
@@ -0,0 +1,98 @@
|
||||
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, ...
|
||||
@@ -0,0 +1,115 @@
|
||||
# no horiz rules between sections
|
||||
$end_section = \&FFmpeg_end_section;
|
||||
sub FFmpeg_end_section($$)
|
||||
{
|
||||
}
|
||||
|
||||
$EXTRA_HEAD =
|
||||
'<link rel="icon" href="favicon.png" type="image/png" />
|
||||
';
|
||||
|
||||
$CSS_LINES = $ENV{"FFMPEG_CSS"} || <<EOT;
|
||||
<link rel="stylesheet" type="text/css" href="default.css" />
|
||||
EOT
|
||||
|
||||
my $TEMPLATE_HEADER = $ENV{"FFMPEG_HEADER"} || <<EOT;
|
||||
<link rel="icon" href="favicon.png" type="image/png" />
|
||||
</head>
|
||||
<body>
|
||||
<div id="container">
|
||||
<div id="body">
|
||||
EOT
|
||||
|
||||
$PRE_BODY_CLOSE = '</div></div>';
|
||||
|
||||
$SMALL_RULE = '';
|
||||
$BODYTEXT = '';
|
||||
|
||||
$print_page_foot = \&FFmpeg_print_page_foot;
|
||||
sub FFmpeg_print_page_foot($$)
|
||||
{
|
||||
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></div></body>\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{'fulltitle_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="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;
|
||||
@@ -0,0 +1,70 @@
|
||||
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.
|
||||
@@ -0,0 +1,453 @@
|
||||
#! /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) unless $skipping;
|
||||
$chapters{$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|(?:multi|[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/\@([.!? ])/$1/g;
|
||||
s/\@[:-]//g;
|
||||
s/\@bullet(?:\{\})?/*/g;
|
||||
s/\@TeX\{\}/TeX/g;
|
||||
s/\@pounds\{\}/\#/g;
|
||||
s/\@minus(?:\{\})?/-/g;
|
||||
|
||||
# Now the ones that have to be replaced by special escapes
|
||||
# (which will be turned back into text by unmunge())
|
||||
s/&/&/g;
|
||||
s/\@\{/{/g;
|
||||
s/\@\}/}/g;
|
||||
s/\@\@/&at;/g;
|
||||
|
||||
# Inside a verbatim block, handle @var specially.
|
||||
if ($shift ne "") {
|
||||
s/\@var\{([^\}]*)\}/<$1>/g;
|
||||
}
|
||||
|
||||
# POD doesn't interpret E<> inside a verbatim block.
|
||||
if ($shift eq "") {
|
||||
s/</</g;
|
||||
s/>/>/g;
|
||||
} else {
|
||||
s/</</g;
|
||||
s/>/>/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";
|
||||
};
|
||||
|
||||
/^\@((?:multi|[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/;
|
||||
$ic =~ s/\@(?:columnfractions)//;
|
||||
$_ = "\n=over 4\n";
|
||||
};
|
||||
|
||||
/^\@((?:small)?example|display)/ and do {
|
||||
push @endwstack, $endw;
|
||||
$endw = $1;
|
||||
$shift = "\t";
|
||||
$_ = ""; # need a paragraph break
|
||||
};
|
||||
|
||||
/^\@item\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
|
||||
my $columns = $1;
|
||||
$columns =~ s/\@tab/ : /;
|
||||
|
||||
$_ = "\n=item B<". $columns .">\n";
|
||||
};
|
||||
|
||||
/^\@tab\s+(.*\S)\s*$/ and $endw eq "multitable" and do {
|
||||
my $columns = $1;
|
||||
$columns =~ s/\@tab/ : /;
|
||||
|
||||
$_ = " : ". $columns;
|
||||
$chapter =~ s/\n+\s+$//;
|
||||
};
|
||||
|
||||
/^\@itemx?\s*(.+)?$/ and do {
|
||||
if (defined $1) {
|
||||
# Entity escapes prevent munging by the <> processing below.
|
||||
$_ = "\n=item $ic\<$1\>\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/\@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)\{([^\},]*),?[^\}]*\}/<B<$1>>/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/</</g;
|
||||
s/>/>/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/</E<lt>/g;
|
||||
s/>/E<gt>/g;
|
||||
s/{/\{/g;
|
||||
s/}/\}/g;
|
||||
s/&at;/\@/g;
|
||||
s/&/&/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;
|
||||
}
|
||||
}
|
||||
File diff suppressed because it is too large
Load Diff
@@ -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
|
||||
Reference in New Issue
Block a user