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@@ -33,3 +33,5 @@ saved_model/**/* filter=lfs diff=lfs merge=lfs -text
 *.zip filter=lfs diff=lfs merge=lfs -text
 *.zst filter=lfs diff=lfs merge=lfs -text
 *tfevents* filter=lfs diff=lfs merge=lfs -text
+ffmpeg-6.1-amd64-static/ffmpeg filter=lfs diff=lfs merge=lfs -text
+ffmpeg-6.1-amd64-static/ffprobe filter=lfs diff=lfs merge=lfs -text

ffmpeg-6.1-amd64-static/GPLv3.txt

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+                    GNU GENERAL PUBLIC LICENSE
+                       Version 3, 29 June 2007
+
+ Copyright (C) 2007 Free Software Foundation, Inc. <http://fsf.org/>
+ Everyone is permitted to copy and distribute verbatim copies
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+
+                            Preamble
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+OF ANY KIND, EITHER EXPRESSED OR IMPLIED, INCLUDING, BUT NOT LIMITED TO,
+THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+PURPOSE.  THE ENTIRE RISK AS TO THE QUALITY AND PERFORMANCE OF THE PROGRAM
+IS WITH YOU.  SHOULD THE PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF
+ALL NECESSARY SERVICING, REPAIR OR CORRECTION.
+
+  16. Limitation of Liability.
+
+  IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING
+WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MODIFIES AND/OR CONVEYS
+THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES, INCLUDING ANY
+GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING OUT OF THE
+USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED TO LOSS OF
+DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY YOU OR THIRD
+PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER PROGRAMS),
+EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE POSSIBILITY OF
+SUCH DAMAGES.
+
+  17. Interpretation of Sections 15 and 16.
+
+  If the disclaimer of warranty and limitation of liability provided
+above cannot be given local legal effect according to their terms,
+reviewing courts shall apply local law that most closely approximates
+an absolute waiver of all civil liability in connection with the
+Program, unless a warranty or assumption of liability accompanies a
+copy of the Program in return for a fee.
+
+                     END OF TERMS AND CONDITIONS
+
+            How to Apply These Terms to Your New Programs
+
+  If you develop a new program, and you want it to be of the greatest
+possible use to the public, the best way to achieve this is to make it
+free software which everyone can redistribute and change under these terms.
+
+  To do so, attach the following notices to the program.  It is safest
+to attach them to the start of each source file to most effectively
+state the exclusion of warranty; and each file should have at least
+the "copyright" line and a pointer to where the full notice is found.
+
+    <one line to give the program's name and a brief idea of what it does.>
+    Copyright (C) <year>  <name of author>
+
+    This program 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 3 of the License, or
+    (at your option) any later version.
+
+    This program 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 this program.  If not, see <http://www.gnu.org/licenses/>.
+
+Also add information on how to contact you by electronic and paper mail.
+
+  If the program does terminal interaction, make it output a short
+notice like this when it starts in an interactive mode:
+
+    <program>  Copyright (C) <year>  <name of author>
+    This program comes with ABSOLUTELY NO WARRANTY; for details type `show w'.
+    This is free software, and you are welcome to redistribute it
+    under certain conditions; type `show c' for details.
+
+The hypothetical commands `show w' and `show c' should show the appropriate
+parts of the General Public License.  Of course, your program's commands
+might be different; for a GUI interface, you would use an "about box".
+
+  You should also get your employer (if you work as a programmer) or school,
+if any, to sign a "copyright disclaimer" for the program, if necessary.
+For more information on this, and how to apply and follow the GNU GPL, see
+<http://www.gnu.org/licenses/>.
+
+  The GNU General Public License does not permit incorporating your program
+into proprietary programs.  If your program is a subroutine library, you
+may consider it more useful to permit linking proprietary applications with
+the library.  If this is what you want to do, use the GNU Lesser General
+Public License instead of this License.  But first, please read
+<http://www.gnu.org/philosophy/why-not-lgpl.html>.

ffmpeg-6.1-amd64-static/ffmpeg

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+oid sha256:a0082b064cc83f5606554fa2cc5b07194ade90f6669b1fcfd6499b29861ca403
+size 78937568

ffmpeg-6.1-amd64-static/ffprobe

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+version https://git-lfs.github.com/spec/v1
+oid sha256:c2b0313686684e48f5dedbe29e510d56e70dead57a5e4219d32c6db32455c32a
+size 78811552

ffmpeg-6.1-amd64-static/manpages/ffmpeg-bitstream-filters.txt

@@ -0,0 +1,863 @@
+FFMPEG-BITSTREAM-FILTERS(1)			   FFMPEG-BITSTREAM-FILTERS(1)
+
+NAME
+       ffmpeg-bitstream-filters - FFmpeg bitstream filters
+
+DESCRIPTION
+       This document describes the bitstream filters provided by the
+       libavcodec library.
+
+       A bitstream filter operates on the encoded stream data, and performs
+       bitstream level modifications without performing decoding.
+
+BITSTREAM FILTERS
+       When you configure your FFmpeg build, all the supported bitstream
+       filters are enabled by default. You can list all available ones using
+       the configure option "--list-bsfs".
+
+       You can disable all the bitstream filters using the configure option
+       "--disable-bsfs", and selectively enable any bitstream filter using the
+       option "--enable-bsf=BSF", or you can disable a particular bitstream
+       filter using the option "--disable-bsf=BSF".
+
+       The option "-bsfs" of the ff* tools will display the list of all the
+       supported bitstream filters included in your build.
+
+       The ff* tools have a -bsf option applied per stream, taking a comma-
+       separated list of filters, whose parameters follow the filter name
+       after a '='.
+
+	       ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
+
+       Below is a description of the currently available bitstream filters,
+       with their parameters, if any.
+
+   aac_adtstoasc
+       Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
+       bitstream.
+
+       This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS
+       header and removes the ADTS header.
+
+       This filter is required for example when copying an AAC stream from a
+       raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
+       to MOV/MP4 files and related formats such as 3GP or M4A. Please note
+       that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
+
+   av1_metadata
+       Modify metadata embedded in an AV1 stream.
+
+       td  Insert or remove temporal delimiter OBUs in all temporal units of
+	   the stream.
+
+	   insert
+	       Insert a TD at the beginning of every TU which does not already
+	       have one.
+
+	   remove
+	       Remove the TD from the beginning of every TU which has one.
+
+       color_primaries
+       transfer_characteristics
+       matrix_coefficients
+	   Set the color description fields in the stream (see AV1 section
+	   6.4.2).
+
+       color_range
+	   Set the color range in the stream (see AV1 section 6.4.2; note that
+	   this cannot be set for streams using BT.709 primaries, sRGB
+	   transfer characteristic and identity (RGB) matrix coefficients).
+
+	   tv  Limited range.
+
+	   pc  Full range.
+
+       chroma_sample_position
+	   Set the chroma sample location in the stream (see AV1 section
+	   6.4.2).  This can only be set for 4:2:0 streams.
+
+	   vertical
+	       Left position (matching the default in MPEG-2 and H.264).
+
+	   colocated
+	       Top-left position.
+
+       tick_rate
+	   Set the tick rate (time_scale / num_units_in_display_tick) in the
+	   timing info in the sequence header.
+
+       num_ticks_per_picture
+	   Set the number of ticks in each picture, to indicate that the
+	   stream has a fixed framerate.  Ignored if tick_rate is not also
+	   set.
+
+       delete_padding
+	   Deletes Padding OBUs.
+
+   chomp
+       Remove zero padding at the end of a packet.
+
+   dca_core
+       Extract the core from a DCA/DTS stream, dropping extensions such as
+       DTS-HD.
+
+   dump_extra
+       Add extradata to the beginning of the filtered packets except when said
+       packets already exactly begin with the extradata that is intended to be
+       added.
+
+       freq
+	   The additional argument specifies which packets should be filtered.
+	   It accepts the values:
+
+	   k
+	   keyframe
+	       add extradata to all key packets
+
+	   e
+	   all add extradata to all packets
+
+       If not specified it is assumed k.
+
+       For example the following ffmpeg command forces a global header (thus
+       disabling individual packet headers) in the H.264 packets generated by
+       the "libx264" encoder, but corrects them by adding the header stored in
+       extradata to the key packets:
+
+	       ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
+
+   dv_error_marker
+       Blocks in DV which are marked as damaged are replaced by blocks of the
+       specified color.
+
+       color
+	   The color to replace damaged blocks by
+
+       sta A 16 bit mask which specifies which of the 16 possible error status
+	   values are to be replaced by colored blocks. 0xFFFE is the default
+	   which replaces all non 0 error status values.
+
+	   ok  No error, no concealment
+
+	   err Error, No concealment
+
+	   res Reserved
+
+	   notok
+	       Error or concealment
+
+	   notres
+	       Not reserved
+
+	   Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
+	       The specific error status code
+
+	   see page 44-46 or section 5.5 of
+	   <http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf>
+
+   eac3_core
+       Extract the core from a E-AC-3 stream, dropping extra channels.
+
+   extract_extradata
+       Extract the in-band extradata.
+
+       Certain codecs allow the long-term headers (e.g. MPEG-2 sequence
+       headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-
+       band" (i.e. as a part of the bitstream containing the coded frames) or
+       "out of band" (e.g. on the container level). This latter form is called
+       "extradata" in FFmpeg terminology.
+
+       This bitstream filter detects the in-band headers and makes them
+       available as extradata.
+
+       remove
+	   When this option is enabled, the long-term headers are removed from
+	   the bitstream after extraction.
+
+   filter_units
+       Remove units with types in or not in a given set from the stream.
+
+       pass_types
+	   List of unit types or ranges of unit types to pass through while
+	   removing all others.  This is specified as a '|'-separated list of
+	   unit type values or ranges of values with '-'.
+
+       remove_types
+	   Identical to pass_types, except the units in the given set removed
+	   and all others passed through.
+
+       Extradata is unchanged by this transformation, but note that if the
+       stream contains inline parameter sets then the output may be unusable
+       if they are removed.
+
+       For example, to remove all non-VCL NAL units from an H.264 stream:
+
+	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
+
+       To remove all AUDs, SEI and filler from an H.265 stream:
+
+	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
+
+   hapqa_extract
+       Extract Rgb or Alpha part of an HAPQA file, without recompression, in
+       order to create an HAPQ or an HAPAlphaOnly file.
+
+       texture
+	   Specifies the texture to keep.
+
+	   color
+	   alpha
+
+       Convert HAPQA to HAPQ
+
+	       ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov
+
+       Convert HAPQA to HAPAlphaOnly
+
+	       ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov
+
+   h264_metadata
+       Modify metadata embedded in an H.264 stream.
+
+       aud Insert or remove AUD NAL units in all access units of the stream.
+
+	   pass
+	   insert
+	   remove
+
+	   Default is pass.
+
+       sample_aspect_ratio
+	   Set the sample aspect ratio of the stream in the VUI parameters.
+	   See H.264 table E-1.
+
+       overscan_appropriate_flag
+	   Set whether the stream is suitable for display using overscan or
+	   not (see H.264 section E.2.1).
+
+       video_format
+       video_full_range_flag
+	   Set the video format in the stream (see H.264 section E.2.1 and
+	   table E-2).
+
+       colour_primaries
+       transfer_characteristics
+       matrix_coefficients
+	   Set the colour description in the stream (see H.264 section E.2.1
+	   and tables E-3, E-4 and E-5).
+
+       chroma_sample_loc_type
+	   Set the chroma sample location in the stream (see H.264 section
+	   E.2.1 and figure E-1).
+
+       tick_rate
+	   Set the tick rate (time_scale / num_units_in_tick) in the VUI
+	   parameters.	This is the smallest time unit representable in the
+	   stream, and in many cases represents the field rate of the stream
+	   (double the frame rate).
+
+       fixed_frame_rate_flag
+	   Set whether the stream has fixed framerate - typically this
+	   indicates that the framerate is exactly half the tick rate, but the
+	   exact meaning is dependent on interlacing and the picture structure
+	   (see H.264 section E.2.1 and table E-6).
+
+       zero_new_constraint_set_flags
+	   Zero constraint_set4_flag and constraint_set5_flag in the SPS.
+	   These bits were reserved in a previous version of the H.264 spec,
+	   and thus some hardware decoders require these to be zero. The
+	   result of zeroing this is still a valid bitstream.
+
+       crop_left
+       crop_right
+       crop_top
+       crop_bottom
+	   Set the frame cropping offsets in the SPS.  These values will
+	   replace the current ones if the stream is already cropped.
+
+	   These fields are set in pixels.  Note that some sizes may not be
+	   representable if the chroma is subsampled or the stream is
+	   interlaced (see H.264 section 7.4.2.1.1).
+
+       sei_user_data
+	   Insert a string as SEI unregistered user data.  The argument must
+	   be of the form UUID+string, where the UUID is as hex digits
+	   possibly separated by hyphens, and the string can be anything.
+
+	   For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert
+	   the string ``hello'' associated with the given UUID.
+
+       delete_filler
+	   Deletes both filler NAL units and filler SEI messages.
+
+       display_orientation
+	   Insert, extract or remove Display orientation SEI messages.	See
+	   H.264 section D.1.27 and D.2.27 for syntax and semantics.
+
+	   pass
+	   insert
+	   remove
+	   extract
+
+	   Default is pass.
+
+	   Insert mode works in conjunction with "rotate" and "flip" options.
+	   Any pre-existing Display orientation messages will be removed in
+	   insert or remove mode.  Extract mode attaches the display matrix to
+	   the packet as side data.
+
+       rotate
+	   Set rotation in display orientation SEI (anticlockwise angle in
+	   degrees).  Range is -360 to +360. Default is NaN.
+
+       flip
+	   Set flip in display orientation SEI.
+
+	   horizontal
+	   vertical
+
+	   Default is unset.
+
+       level
+	   Set the level in the SPS.  Refer to H.264 section A.3 and tables
+	   A-1 to A-5.
+
+	   The argument must be the name of a level (for example, 4.2), a
+	   level_idc value (for example, 42), or the special name auto
+	   indicating that the filter should attempt to guess the level from
+	   the input stream properties.
+
+   h264_mp4toannexb
+       Convert an H.264 bitstream from length prefixed mode to start code
+       prefixed mode (as defined in the Annex B of the ITU-T H.264
+       specification).
+
+       This is required by some streaming formats, typically the MPEG-2
+       transport stream format (muxer "mpegts").
+
+       For example to remux an MP4 file containing an H.264 stream to mpegts
+       format with ffmpeg, you can use the command:
+
+	       ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
+
+       Please note that this filter is auto-inserted for MPEG-TS (muxer
+       "mpegts") and raw H.264 (muxer "h264") output formats.
+
+   h264_redundant_pps
+       This applies a specific fixup to some Blu-ray streams which contain
+       redundant PPSs modifying irrelevant parameters of the stream which
+       confuse other transformations which require correct extradata.
+
+   hevc_metadata
+       Modify metadata embedded in an HEVC stream.
+
+       aud Insert or remove AUD NAL units in all access units of the stream.
+
+	   insert
+	   remove
+       sample_aspect_ratio
+	   Set the sample aspect ratio in the stream in the VUI parameters.
+
+       video_format
+       video_full_range_flag
+	   Set the video format in the stream (see H.265 section E.3.1 and
+	   table E.2).
+
+       colour_primaries
+       transfer_characteristics
+       matrix_coefficients
+	   Set the colour description in the stream (see H.265 section E.3.1
+	   and tables E.3, E.4 and E.5).
+
+       chroma_sample_loc_type
+	   Set the chroma sample location in the stream (see H.265 section
+	   E.3.1 and figure E.1).
+
+       tick_rate
+	   Set the tick rate in the VPS and VUI parameters (time_scale /
+	   num_units_in_tick). Combined with num_ticks_poc_diff_one, this can
+	   set a constant framerate in the stream.  Note that it is likely to
+	   be overridden by container parameters when the stream is in a
+	   container.
+
+       num_ticks_poc_diff_one
+	   Set poc_proportional_to_timing_flag in VPS and VUI and use this
+	   value to set num_ticks_poc_diff_one_minus1 (see H.265 sections
+	   7.4.3.1 and E.3.1).	Ignored if tick_rate is not also set.
+
+       crop_left
+       crop_right
+       crop_top
+       crop_bottom
+	   Set the conformance window cropping offsets in the SPS.  These
+	   values will replace the current ones if the stream is already
+	   cropped.
+
+	   These fields are set in pixels.  Note that some sizes may not be
+	   representable if the chroma is subsampled (H.265 section
+	   7.4.3.2.1).
+
+       level
+	   Set the level in the VPS and SPS.  See H.265 section A.4 and tables
+	   A.6 and A.7.
+
+	   The argument must be the name of a level (for example, 5.1), a
+	   general_level_idc value (for example, 153 for level 5.1), or the
+	   special name auto indicating that the filter should attempt to
+	   guess the level from the input stream properties.
+
+   hevc_mp4toannexb
+       Convert an HEVC/H.265 bitstream from length prefixed mode to start code
+       prefixed mode (as defined in the Annex B of the ITU-T H.265
+       specification).
+
+       This is required by some streaming formats, typically the MPEG-2
+       transport stream format (muxer "mpegts").
+
+       For example to remux an MP4 file containing an HEVC stream to mpegts
+       format with ffmpeg, you can use the command:
+
+	       ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
+
+       Please note that this filter is auto-inserted for MPEG-TS (muxer
+       "mpegts") and raw HEVC/H.265 (muxer "h265" or "hevc") output formats.
+
+   imxdump
+       Modifies the bitstream to fit in MOV and to be usable by the Final Cut
+       Pro decoder. This filter only applies to the mpeg2video codec, and is
+       likely not needed for Final Cut Pro 7 and newer with the appropriate
+       -tag:v.
+
+       For example, to remux 30 MB/sec NTSC IMX to MOV:
+
+	       ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
+
+   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
+
+	       ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
+
+       Unfortunately, these chunks are incomplete JPEG images, because they
+       lack the DHT segment required for decoding. Quoting from
+       <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:
+
+       Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
+       commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG
+       fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman
+       table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must
+       use basic Huffman encoding, not arithmetic or progressive. . . . You
+       can indeed extract the MJPEG frames and decode them with a regular JPEG
+       decoder, but you have to prepend the DHT segment to them, or else the
+       decoder won't have any idea how to decompress the data. The exact table
+       necessary is given in the OpenDML spec."
+
+       This bitstream filter patches the header of frames extracted from an
+       MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to
+       produce fully qualified JPEG images.
+
+	       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
+
+   mjpegadump
+       Add an MJPEG A header to the bitstream, to enable decoding by
+       Quicktime.
+
+   mov2textsub
+       Extract a representable text file from MOV subtitles, stripping the
+       metadata header from each subtitle packet.
+
+       See also the text2movsub filter.
+
+   mp3decomp
+       Decompress non-standard compressed MP3 audio headers.
+
+   mpeg2_metadata
+       Modify metadata embedded in an MPEG-2 stream.
+
+       display_aspect_ratio
+	   Set the display aspect ratio in the stream.
+
+	   The following fixed values are supported:
+
+	   4/3
+	   16/9
+	   221/100
+
+	   Any other value will result in square pixels being signalled
+	   instead (see H.262 section 6.3.3 and table 6-3).
+
+       frame_rate
+	   Set the frame rate in the stream.  This is constructed from a table
+	   of known values combined with a small multiplier and divisor - if
+	   the supplied value is not exactly representable, the nearest
+	   representable value will be used instead (see H.262 section 6.3.3
+	   and table 6-4).
+
+       video_format
+	   Set the video format in the stream (see H.262 section 6.3.6 and
+	   table 6-6).
+
+       colour_primaries
+       transfer_characteristics
+       matrix_coefficients
+	   Set the colour description in the stream (see H.262 section 6.3.6
+	   and tables 6-7, 6-8 and 6-9).
+
+   mpeg4_unpack_bframes
+       Unpack DivX-style packed B-frames.
+
+       DivX-style packed B-frames are not valid MPEG-4 and were only a
+       workaround for the broken Video for Windows subsystem.  They use more
+       space, can cause minor AV sync issues, require more CPU power to decode
+       (unless the player has some decoded picture queue to compensate the
+       2,0,2,0 frame per packet style) and cause trouble if copied into a
+       standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may
+       not be able to decode them, since they are not valid MPEG-4.
+
+       For example to fix an AVI file containing an MPEG-4 stream with DivX-
+       style packed B-frames using ffmpeg, you can use the command:
+
+	       ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
+
+   noise
+       Damages the contents of packets or simply drops them without damaging
+       the container. Can be used for fuzzing or testing error
+       resilience/concealment.
+
+       Parameters:
+
+       amount
+	   Accepts an expression whose evaluation per-packet determines how
+	   often bytes in that packet will be modified. A value below 0 will
+	   result in a variable frequency.  Default is 0 which results in no
+	   modification. However, if neither amount nor drop is specified,
+	   amount will be set to -1. See below for accepted variables.
+
+       drop
+	   Accepts an expression evaluated per-packet whose value determines
+	   whether that packet is dropped.  Evaluation to a positive value
+	   results in the packet being dropped. Evaluation to a negative value
+	   results in a variable chance of it being dropped, roughly inverse
+	   in proportion to the magnitude of the value. Default is 0 which
+	   results in no drops. See below for accepted variables.
+
+       dropamount
+	   Accepts a non-negative integer, which assigns a variable chance of
+	   it being dropped, roughly inverse in proportion to the value.
+	   Default is 0 which results in no drops. This option is kept for
+	   backwards compatibility and is equivalent to setting drop to a
+	   negative value with the same magnitude i.e. "dropamount=4" is the
+	   same as "drop=-4". Ignored if drop is also specified.
+
+       Both "amount" and "drop" accept expressions containing the following
+       variables:
+
+       n   The index of the packet, starting from zero.
+
+       tb  The timebase for packet timestamps.
+
+       pts Packet presentation timestamp.
+
+       dts Packet decoding timestamp.
+
+       nopts
+	   Constant representing AV_NOPTS_VALUE.
+
+       startpts
+	   First non-AV_NOPTS_VALUE PTS seen in the stream.
+
+       startdts
+	   First non-AV_NOPTS_VALUE DTS seen in the stream.
+
+       duration
+       d   Packet duration, in timebase units.
+
+       pos Packet position in input; may be -1 when unknown or not set.
+
+       size
+	   Packet size, in bytes.
+
+       key Whether packet is marked as a keyframe.
+
+       state
+	   A pseudo random integer, primarily derived from the content of
+	   packet payload.
+
+       Examples
+
+       Apply modification to every byte but don't drop any packets.
+
+	       ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv
+
+       Drop every video packet not marked as a keyframe after timestamp 30s
+       but do not modify any of the remaining packets.
+
+	       ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv
+
+       Drop one second of audio every 10 seconds and add some random noise to
+       the rest.
+
+	       ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv
+
+   null
+       This bitstream filter passes the packets through unchanged.
+
+   pcm_rechunk
+       Repacketize PCM audio to a fixed number of samples per packet or a
+       fixed packet rate per second. This is similar to the asetnsamples audio
+       filter but works on audio packets instead of audio frames.
+
+       nb_out_samples, n
+	   Set the number of samples per each output audio packet. The number
+	   is intended as the number of samples per each channel. Default
+	   value is 1024.
+
+       pad, p
+	   If set to 1, the filter will pad the last audio packet with
+	   silence, so that it will contain the same number of samples (or
+	   roughly the same number of samples, see frame_rate) as the previous
+	   ones. Default value is 1.
+
+       frame_rate, r
+	   This option makes the filter output a fixed number of packets per
+	   second instead of a fixed number of samples per packet. If the
+	   audio sample rate is not divisible by the frame rate then the
+	   number of samples will not be constant but will vary slightly so
+	   that each packet will start as close to the frame boundary as
+	   possible. Using this option has precedence over nb_out_samples.
+
+       You can generate the well known 1602-1601-1602-1601-1602 pattern of
+       48kHz audio for NTSC frame rate using the frame_rate option.
+
+	       ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
+
+   pgs_frame_merge
+       Merge a sequence of PGS Subtitle segments ending with an "end of
+       display set" segment into a single packet.
+
+       This is required by some containers that support PGS subtitles (muxer
+       "matroska").
+
+   prores_metadata
+       Modify color property metadata embedded in prores stream.
+
+       color_primaries
+	   Set the color primaries.  Available values are:
+
+	   auto
+	       Keep the same color primaries property (default).
+
+	   unknown
+	   bt709
+	   bt470bg
+	       BT601 625
+
+	   smpte170m
+	       BT601 525
+
+	   bt2020
+	   smpte431
+	       DCI P3
+
+	   smpte432
+	       P3 D65
+
+       transfer_characteristics
+	   Set the color transfer.  Available values are:
+
+	   auto
+	       Keep the same transfer characteristics property (default).
+
+	   unknown
+	   bt709
+	       BT 601, BT 709, BT 2020
+
+	   smpte2084
+	       SMPTE ST 2084
+
+	   arib-std-b67
+	       ARIB STD-B67
+
+       matrix_coefficients
+	   Set the matrix coefficient.	Available values are:
+
+	   auto
+	       Keep the same colorspace property (default).
+
+	   unknown
+	   bt709
+	   smpte170m
+	       BT 601
+
+	   bt2020nc
+
+       Set Rec709 colorspace for each frame of the file
+
+	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
+
+       Set Hybrid Log-Gamma parameters for each frame of the file
+
+	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
+
+   remove_extra
+       Remove extradata from packets.
+
+       It accepts the following parameter:
+
+       freq
+	   Set which frame types to remove extradata from.
+
+	   k   Remove extradata from non-keyframes only.
+
+	   keyframe
+	       Remove extradata from keyframes only.
+
+	   e, all
+	       Remove extradata from all frames.
+
+   setts
+       Set PTS and DTS in packets.
+
+       It accepts the following parameters:
+
+       ts
+       pts
+       dts Set expressions for PTS, DTS or both.
+
+       duration
+	   Set expression for duration.
+
+       time_base
+	   Set output time base.
+
+       The expressions are evaluated through the eval API and can contain the
+       following constants:
+
+       N   The count of the input packet. Starting from 0.
+
+       TS  The demux timestamp in input in case of "ts" or "dts" option or
+	   presentation timestamp in case of "pts" option.
+
+       POS The original position in the file of the packet, or undefined if
+	   undefined for the current packet
+
+       DTS The demux timestamp in input.
+
+       PTS The presentation timestamp in input.
+
+       DURATION
+	   The duration in input.
+
+       STARTDTS
+	   The DTS of the first packet.
+
+       STARTPTS
+	   The PTS of the first packet.
+
+       PREV_INDTS
+	   The previous input DTS.
+
+       PREV_INPTS
+	   The previous input PTS.
+
+       PREV_INDURATION
+	   The previous input duration.
+
+       PREV_OUTDTS
+	   The previous output DTS.
+
+       PREV_OUTPTS
+	   The previous output PTS.
+
+       PREV_OUTDURATION
+	   The previous output duration.
+
+       NEXT_DTS
+	   The next input DTS.
+
+       NEXT_PTS
+	   The next input PTS.
+
+       NEXT_DURATION
+	   The next input duration.
+
+       TB  The timebase of stream packet belongs.
+
+       TB_OUT
+	   The output timebase.
+
+       SR  The sample rate of stream packet belongs.
+
+       NOPTS
+	   The AV_NOPTS_VALUE constant.
+
+   text2movsub
+       Convert text subtitles to MOV subtitles (as used by the "mov_text"
+       codec) with metadata headers.
+
+       See also the mov2textsub filter.
+
+   trace_headers
+       Log trace output containing all syntax elements in the coded stream
+       headers (everything above the level of individual coded blocks).  This
+       can be useful for debugging low-level stream issues.
+
+       Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on
+       the build only a subset of these may be available.
+
+   truehd_core
+       Extract the core from a TrueHD stream, dropping ATMOS data.
+
+   vp9_metadata
+       Modify metadata embedded in a VP9 stream.
+
+       color_space
+	   Set the color space value in the frame header.  Note that any frame
+	   set to RGB will be implicitly set to PC range and that RGB is
+	   incompatible with profiles 0 and 2.
+
+	   unknown
+	   bt601
+	   bt709
+	   smpte170
+	   smpte240
+	   bt2020
+	   rgb
+       color_range
+	   Set the color range value in the frame header.  Note that any value
+	   imposed by the color space will take precedence over this value.
+
+	   tv
+	   pc
+
+   vp9_superframe
+       Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
+       fixes merging of split/segmented VP9 streams where the alt-ref frame
+       was split from its visible counterpart.
+
+   vp9_superframe_split
+       Split VP9 superframes into single frames.
+
+   vp9_raw_reorder
+       Given a VP9 stream with correct timestamps but possibly out of order,
+       insert additional show-existing-frame packets to correct the ordering.
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+						   FFMPEG-BITSTREAM-FILTERS(1)

ffmpeg-6.1-amd64-static/manpages/ffmpeg-devices.txt

@@ -0,0 +1,1904 @@
+FFMPEG-DEVICES(1)					     FFMPEG-DEVICES(1)
+
+NAME
+       ffmpeg-devices - FFmpeg devices
+
+DESCRIPTION
+       This document describes the input and output devices provided by the
+       libavdevice library.
+
+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 -option value in the FFmpeg tools, or
+       by setting the value explicitly in the device "AVFormatContext" options
+       or using the libavutil/opt.h API for programmatic use.
+
+INPUT DEVICES
+       Input devices are configured elements in FFmpeg which enable accessing
+       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=INDEV", or you can disable a particular input
+       device using the option "--disable-indev=INDEV".
+
+       The option "-devices" of the ff* tools will display the list of
+       supported input devices.
+
+       A description of the currently available input devices follows.
+
+   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:
+
+	       hw:<CARD>[,<DEV>[,<SUBDEV>]]
+
+       where the DEV and SUBDEV components are optional.
+
+       The three arguments (in order: CARD,DEV,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 /proc/asound/cards and /proc/asound/devices.
+
+       For example to capture with ffmpeg from an ALSA device with card id 0,
+       you may run the command:
+
+	       ffmpeg -f alsa -i hw:0 alsaout.wav
+
+       For more information see:
+       <http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>
+
+       Options
+
+       sample_rate
+	   Set the sample rate in Hz. Default is 48000.
+
+       channels
+	   Set the number of channels. Default is 2.
+
+   android_camera
+       Android camera input device.
+
+       This input devices uses the Android Camera2 NDK API which is available
+       on devices with API level 24+. The availability of android_camera is
+       autodetected during configuration.
+
+       This device allows capturing from all cameras on an Android device,
+       which are integrated into the Camera2 NDK API.
+
+       The available cameras are enumerated internally and can be selected
+       with the camera_index parameter. The input file string is discarded.
+
+       Generally the back facing camera has index 0 while the front facing
+       camera has index 1.
+
+       Options
+
+       video_size
+	   Set the video size given as a string such as 640x480 or hd720.
+	   Falls back to the first available configuration reported by Android
+	   if requested video size is not available or by default.
+
+       framerate
+	   Set the video framerate.  Falls back to the first available
+	   configuration reported by Android if requested framerate is not
+	   available or by default (-1).
+
+       camera_index
+	   Set the index of the camera to use. Default is 0.
+
+       input_queue_size
+	   Set the maximum number of frames to buffer. Default is 5.
+
+   avfoundation
+       AVFoundation input device.
+
+       AVFoundation is the currently recommended framework by Apple for
+       streamgrabbing on OSX >= 10.7 as well as on iOS.
+
+       The input filename has to be given in the following syntax:
+
+	       -i "[[VIDEO]:[AUDIO]]"
+
+       The first entry selects the video input while the latter selects the
+       audio input.  The stream has to be specified by the device name or the
+       device index as shown by the device list.  Alternatively, the video
+       and/or audio input device can be chosen by index using the
+
+	   B<-video_device_index E<lt>INDEXE<gt>>
+
+       and/or
+
+	   B<-audio_device_index E<lt>INDEXE<gt>>
+
+       , overriding any device name or index given in the input filename.
+
+       All available devices can be enumerated by using -list_devices true,
+       listing all device names and corresponding indices.
+
+       There are two device name aliases:
+
+       "default"
+	   Select the AVFoundation default device of the corresponding type.
+
+       "none"
+	   Do not record the corresponding media type.	This is equivalent to
+	   specifying an empty device name or index.
+
+       Options
+
+       AVFoundation supports the following options:
+
+       -list_devices <TRUE|FALSE>
+	   If set to true, a list of all available input devices is given
+	   showing all device names and indices.
+
+       -video_device_index <INDEX>
+	   Specify the video device by its index. Overrides anything given in
+	   the input filename.
+
+       -audio_device_index <INDEX>
+	   Specify the audio device by its index. Overrides anything given in
+	   the input filename.
+
+       -pixel_format <FORMAT>
+	   Request the video device to use a specific pixel format.  If the
+	   specified format is not supported, a list of available formats is
+	   given and the first one in this list is used instead. Available
+	   pixel formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le,
+	   rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
+	    bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16,
+	   yuv422p10, yuv444p10,
+	    yuv420p, nv12, yuyv422, gray"
+
+       -framerate
+	   Set the grabbing frame rate. Default is "ntsc", corresponding to a
+	   frame rate of "30000/1001".
+
+       -video_size
+	   Set the video frame size.
+
+       -capture_cursor
+	   Capture the mouse pointer. Default is 0.
+
+       -capture_mouse_clicks
+	   Capture the screen mouse clicks. Default is 0.
+
+       -capture_raw_data
+	   Capture the raw device data. Default is 0.  Using this option may
+	   result in receiving the underlying data delivered to the
+	   AVFoundation framework. E.g. for muxed devices that sends raw DV
+	   data to the framework (like tape-based camcorders), setting this
+	   option to false results in extracted video frames captured in the
+	   designated pixel format only. Setting this option to true results
+	   in receiving the raw DV stream untouched.
+
+       Examples
+
+       o   Print the list of AVFoundation supported devices and exit:
+
+		   $ ffmpeg -f avfoundation -list_devices true -i ""
+
+       o   Record video from video device 0 and audio from audio device 0 into
+	   out.avi:
+
+		   $ ffmpeg -f avfoundation -i "0:0" out.avi
+
+       o   Record video from video device 2 and audio from audio device 1 into
+	   out.avi:
+
+		   $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
+
+       o   Record video from the system default video device using the pixel
+	   format bgr0 and do not record any audio into out.avi:
+
+		   $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
+
+       o   Record raw DV data from a suitable input device and write the
+	   output into out.dv:
+
+		   $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv
+
+   bktr
+       BSD video input device.
+
+       Options
+
+       framerate
+	   Set the frame rate.
+
+       video_size
+	   Set the video frame size. Default is "vga".
+
+       standard
+	   Available values are:
+
+	   pal
+	   ntsc
+	   secam
+	   paln
+	   palm
+	   ntscj
+
+   decklink
+       The decklink input device provides capture capabilities for Blackmagic
+       DeckLink devices.
+
+       To enable this input device, you need the Blackmagic DeckLink SDK and
+       you need to configure with the appropriate "--extra-cflags" and
+       "--extra-ldflags".  On Windows, you need to run the IDL files through
+       widl.
+
+       DeckLink is very picky about the formats it supports. Pixel format of
+       the input can be set with raw_format.  Framerate and video size must be
+       determined for your device with -list_formats 1. Audio sample rate is
+       always 48 kHz and the number of channels can be 2, 8 or 16. Note that
+       all audio channels are bundled in one single audio track.
+
+       Options
+
+       list_devices
+	   If set to true, print a list of devices and exit.  Defaults to
+	   false. This option is deprecated, please use the "-sources" option
+	   of ffmpeg to list the available input devices.
+
+       list_formats
+	   If set to true, print a list of supported formats and exit.
+	   Defaults to false.
+
+       format_code <FourCC>
+	   This sets the input video format to the format given by the FourCC.
+	   To see the supported values of your device(s) use list_formats.
+	   Note that there is a FourCC 'pal ' that can also be used as pal (3
+	   letters).  Default behavior is autodetection of the input video
+	   format, if the hardware supports it.
+
+       raw_format
+	   Set the pixel format of the captured video.	Available values are:
+
+	   auto
+	       This is the default which means 8-bit YUV 422 or 8-bit ARGB if
+	       format autodetection is used, 8-bit YUV 422 otherwise.
+
+	   uyvy422
+	       8-bit YUV 422.
+
+	   yuv422p10
+	       10-bit YUV 422.
+
+	   argb
+	       8-bit RGB.
+
+	   bgra
+	       8-bit RGB.
+
+	   rgb10
+	       10-bit RGB.
+
+       teletext_lines
+	   If set to nonzero, an additional teletext stream will be captured
+	   from the vertical ancillary data. Both SD PAL (576i) and HD (1080i
+	   or 1080p) sources are supported. In case of HD sources, OP47
+	   packets are decoded.
+
+	   This option is a bitmask of the SD PAL VBI lines captured,
+	   specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB
+	   in the mask. Selected lines which do not contain teletext
+	   information will be ignored. You can use the special all constant
+	   to select all possible lines, or standard to skip lines 6, 318 and
+	   319, which are not compatible with all receivers.
+
+	   For SD sources, ffmpeg needs to be compiled with
+	   "--enable-libzvbi". For HD sources, on older (pre-4K) DeckLink card
+	   models you have to capture in 10 bit mode.
+
+       channels
+	   Defines number of audio channels to capture. Must be 2, 8 or 16.
+	   Defaults to 2.
+
+       duplex_mode
+	   Sets the decklink device duplex/profile mode. Must be unset, half,
+	   full, one_sub_device_full, one_sub_device_half,
+	   two_sub_device_full, four_sub_device_half Defaults to unset.
+
+	   Note: DeckLink SDK 11.0 have replaced the duplex property by a
+	   profile property.  For the DeckLink Duo 2 and DeckLink Quad 2, a
+	   profile is shared between any 2 sub-devices that utilize the same
+	   connectors. For the DeckLink 8K Pro, a profile is shared between
+	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.
+
+	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
+	   one_sub_device_full, one_sub_device_half, two_sub_device_full,
+	   four_sub_device_half
+
+	   Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
+	   full
+
+       timecode_format
+	   Timecode type to include in the frame and video stream metadata.
+	   Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any,
+	   vitc, vitc2, or serial.  Defaults to none (not included).
+
+	   In order to properly support 50/60 fps timecodes, the ordering of
+	   the queried timecode types for rp188any is HFR, VITC1, VITC2 and
+	   LTC for >30 fps content. Note that this is slightly different to
+	   the ordering used by the DeckLink API, which is HFR, VITC1, LTC,
+	   VITC2.
+
+       video_input
+	   Sets the video input source. Must be unset, sdi, hdmi, optical_sdi,
+	   component, composite or s_video.  Defaults to unset.
+
+       audio_input
+	   Sets the audio input source. Must be unset, embedded, aes_ebu,
+	   analog, analog_xlr, analog_rca or microphone. Defaults to unset.
+
+       video_pts
+	   Sets the video packet timestamp source. Must be video, audio,
+	   reference, wallclock or abs_wallclock.  Defaults to video.
+
+       audio_pts
+	   Sets the audio packet timestamp source. Must be video, audio,
+	   reference, wallclock or abs_wallclock.  Defaults to audio.
+
+       draw_bars
+	   If set to true, color bars are drawn in the event of a signal loss.
+	   Defaults to true.
+
+       queue_size
+	   Sets maximum input buffer size in bytes. If the buffering reaches
+	   this value, incoming frames will be dropped.  Defaults to
+	   1073741824.
+
+       audio_depth
+	   Sets the audio sample bit depth. Must be 16 or 32.  Defaults to 16.
+
+       decklink_copyts
+	   If set to true, timestamps are forwarded as they are without
+	   removing the initial offset.  Defaults to false.
+
+       timestamp_align
+	   Capture start time alignment in seconds. If set to nonzero, input
+	   frames are dropped till the system timestamp aligns with configured
+	   value.  Alignment difference of up to one frame duration is
+	   tolerated.  This is useful for maintaining input synchronization
+	   across N different hardware devices deployed for 'N-way'
+	   redundancy. The system time of different hardware devices should be
+	   synchronized with protocols such as NTP or PTP, before using this
+	   option.  Note that this method is not foolproof. In some border
+	   cases input synchronization may not happen due to thread scheduling
+	   jitters in the OS.  Either sync could go wrong by 1 frame or in a
+	   rarer case timestamp_align seconds.	Defaults to 0.
+
+       wait_for_tc (bool)
+	   Drop frames till a frame with timecode is received. Sometimes
+	   serial timecode isn't received with the first input frame. If that
+	   happens, the stored stream timecode will be inaccurate. If this
+	   option is set to true, input frames are dropped till a frame with
+	   timecode is received.  Option timecode_format must be specified.
+	   Defaults to false.
+
+       enable_klv(bool)
+	   If set to true, extracts KLV data from VANC and outputs KLV
+	   packets.  KLV VANC packets are joined based on MID and PSC fields
+	   and aggregated into one KLV packet.	Defaults to false.
+
+       Examples
+
+       o   List input devices:
+
+		   ffmpeg -sources decklink
+
+       o   List supported formats:
+
+		   ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
+
+       o   Capture video clip at 1080i50:
+
+		   ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
+
+       o   Capture video clip at 1080i50 10 bit:
+
+		   ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
+
+       o   Capture video clip at 1080i50 with 16 audio channels:
+
+		   ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
+
+   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:
+
+	       <TYPE>=<NAME>[:<TYPE>=<NAME>]
+
+       where TYPE can be either audio or video, and NAME is the device's name
+       or alternative name..
+
+       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.
+
+       video_size
+	   Set the video size in the captured video.
+
+       framerate
+	   Set the frame rate in the captured video.
+
+       sample_rate
+	   Set the sample rate (in Hz) of the captured audio.
+
+       sample_size
+	   Set the sample size (in bits) of the captured audio.
+
+       channels
+	   Set the number of channels in the captured audio.
+
+       list_devices
+	   If set to true, print a list of devices and exit.
+
+       list_options
+	   If set to true, print a list of selected device's options and exit.
+
+       video_device_number
+	   Set video device number for devices with the same name (starts at
+	   0, defaults to 0).
+
+       audio_device_number
+	   Set audio device number for devices with the same name (starts at
+	   0, defaults to 0).
+
+       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.
+
+       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
+	   <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>
+
+       video_pin_name
+	   Select video capture pin to use by name or alternative name.
+
+       audio_pin_name
+	   Select audio capture pin to use by name or alternative name.
+
+       crossbar_video_input_pin_number
+	   Select video input pin number for crossbar device. This will be
+	   routed to the crossbar device's Video Decoder output pin.  Note
+	   that changing this value can affect future invocations (sets a new
+	   default) until system reboot occurs.
+
+       crossbar_audio_input_pin_number
+	   Select audio input pin number for crossbar device. This will be
+	   routed to the crossbar device's Audio Decoder output pin.  Note
+	   that changing this value can affect future invocations (sets a new
+	   default) until system reboot occurs.
+
+       show_video_device_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to change video filter properties and
+	   configurations manually.  Note that for crossbar devices, adjusting
+	   values in this dialog may be needed at times to toggle between PAL
+	   (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing,
+	   etc.  Changing these values can enable different scan rates/frame
+	   rates and avoiding green bars at the bottom, flickering scan lines,
+	   etc.  Note that with some devices, changing these properties can
+	   also affect future invocations (sets new defaults) until system
+	   reboot occurs.
+
+       show_audio_device_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to change audio filter properties and
+	   configurations manually.
+
+       show_video_crossbar_connection_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to manually modify crossbar pin
+	   routings, when it opens a video device.
+
+       show_audio_crossbar_connection_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to manually modify crossbar pin
+	   routings, when it opens an audio device.
+
+       show_analog_tv_tuner_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to manually modify TV channels and
+	   frequencies.
+
+       show_analog_tv_tuner_audio_dialog
+	   If set to true, before capture starts, popup a display dialog to
+	   the end user, allowing them to manually modify TV audio (like mono
+	   vs. stereo, Language A,B or C).
+
+       audio_device_load
+	   Load an audio capture filter device from file instead of searching
+	   it by name. It may load additional parameters too, if the filter
+	   supports the serialization of its properties to.  To use this an
+	   audio capture source has to be specified, but it can be anything
+	   even fake one.
+
+       audio_device_save
+	   Save the currently used audio capture filter device and its
+	   parameters (if the filter supports it) to a file.  If a file with
+	   the same name exists it will be overwritten.
+
+       video_device_load
+	   Load a video capture filter device from file instead of searching
+	   it by name. It may load additional parameters too, if the filter
+	   supports the serialization of its properties to.  To use this a
+	   video capture source has to be specified, but it can be anything
+	   even fake one.
+
+       video_device_save
+	   Save the currently used video capture filter device and its
+	   parameters (if the filter supports it) to a file.  If a file with
+	   the same name exists it will be overwritten.
+
+       use_video_device_timestamps
+	   If set to false, the timestamp for video frames will be derived
+	   from the wallclock instead of the timestamp provided by the capture
+	   device. This allows working around devices that provide unreliable
+	   timestamps.
+
+       Examples
+
+       o   Print the list of DirectShow supported devices and exit:
+
+		   $ ffmpeg -list_devices true -f dshow -i dummy
+
+       o   Open video device Camera:
+
+		   $ ffmpeg -f dshow -i video="Camera"
+
+       o   Open second video device with name Camera:
+
+		   $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
+
+       o   Open video device Camera and audio device Microphone:
+
+		   $ ffmpeg -f dshow -i video="Camera":audio="Microphone"
+
+       o   Print the list of supported options in selected device and exit:
+
+		   $ ffmpeg -list_options true -f dshow -i video="Camera"
+
+       o   Specify pin names to capture by name or alternative name, specify
+	   alternative device name:
+
+		   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"
+
+       o   Configure a crossbar device, specifying crossbar pins, allow user
+	   to adjust video capture properties at startup:
+
+		   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
+			-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
+
+   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 /dev/fb0.
+
+       For more detailed information read the file
+       Documentation/fb/framebuffer.txt included in the Linux source tree.
+
+       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).
+
+       To record from the framebuffer device /dev/fb0 with ffmpeg:
+
+	       ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
+
+       You can take a single screenshot image with the command:
+
+	       ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
+
+       Options
+
+       framerate
+	   Set the frame rate. Default is 25.
+
+   gdigrab
+       Win32 GDI-based screen capture device.
+
+       This device allows you to capture a region of the display on Windows.
+
+       There are two options for the input filename:
+
+	       desktop
+
+       or
+
+	       title=<window_title>
+
+       The first option will capture the entire desktop, or a fixed region of
+       the desktop. The second option will instead capture the contents of a
+       single window, regardless of its position on the screen.
+
+       For example, to grab the entire desktop using ffmpeg:
+
+	       ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
+
+       Grab a 640x480 region at position "10,20":
+
+	       ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
+
+       Grab the contents of the window named "Calculator"
+
+	       ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
+
+       Options
+
+       draw_mouse
+	   Specify whether to draw the mouse pointer. Use the value 0 to not
+	   draw the pointer. Default value is 1.
+
+       framerate
+	   Set the grabbing frame rate. Default value is "ntsc", corresponding
+	   to a frame rate of "30000/1001".
+
+       show_region
+	   Show grabbed region on screen.
+
+	   If show_region is specified with 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.
+
+	   Note that show_region is incompatible with grabbing the contents of
+	   a single window.
+
+	   For example:
+
+		   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
+
+       video_size
+	   Set the video frame size. The default is to capture the full screen
+	   if desktop is selected, or the full window size if
+	   title=window_title is selected.
+
+       offset_x
+	   When capturing a region with video_size, set the distance from the
+	   left edge of the screen or desktop.
+
+	   Note that the offset calculation is from the top left corner of the
+	   primary monitor on Windows. If you have a monitor positioned to the
+	   left of your primary monitor, you will need to use a negative
+	   offset_x value to move the region to that monitor.
+
+       offset_y
+	   When capturing a region with video_size, set the distance from the
+	   top edge of the screen or desktop.
+
+	   Note that the offset calculation is from the top left corner of the
+	   primary monitor on Windows. If you have a monitor positioned above
+	   your primary monitor, you will need to use a negative offset_y
+	   value to move the region to that monitor.
+
+   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
+       "--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.
+
+       Options
+
+       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 auto,
+	   dv and hdv are supported.
+
+       dvbuffer
+	   Set maximum 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.
+
+       dvguid
+	   Select the capture device by specifying its 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.
+
+       Examples
+
+       o   Grab and show the input of a FireWire DV/HDV device.
+
+		   ffplay -f iec61883 -i auto
+
+       o   Grab and record the input of a FireWire DV/HDV device, using a
+	   packet buffer of 100000 packets if the source is HDV.
+
+		   ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg
+
+   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 client_name:input_N, where client_name is
+       the name provided by the application, and 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 jack_connect and
+       jack_disconnect programs, or do it through a graphical interface, for
+       example with qjackctl.
+
+       To list the JACK clients and their properties you can invoke the
+       command jack_lsp.
+
+       Follows an example which shows how to capture a JACK readable client
+       with ffmpeg.
+
+	       # 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
+
+       For more information read: <http://jackaudio.org/>
+
+       Options
+
+       channels
+	   Set the number of channels. Default is 2.
+
+   kmsgrab
+       KMS video input device.
+
+       Captures the KMS scanout framebuffer associated with a specified CRTC
+       or plane as a DRM object that can be passed to other hardware
+       functions.
+
+       Requires either DRM master or CAP_SYS_ADMIN to run.
+
+       If you don't understand what all of that means, you probably don't want
+       this.  Look at x11grab instead.
+
+       Options
+
+       device
+	   DRM device to capture on.  Defaults to /dev/dri/card0.
+
+       format
+	   Pixel format of the framebuffer.  This can be autodetected if you
+	   are running Linux 5.7 or later, but needs to be provided for
+	   earlier versions.  Defaults to bgr0, which is the most common
+	   format used by the Linux console and Xorg X server.
+
+       format_modifier
+	   Format modifier to signal on output frames.	This is necessary to
+	   import correctly into some APIs.  It can be autodetected if you are
+	   running Linux 5.7 or later, but will need to be provided explicitly
+	   when needed in earlier versions.  See the libdrm documentation for
+	   possible values.
+
+       crtc_id
+	   KMS CRTC ID to define the capture source.  The first active plane
+	   on the given CRTC will be used.
+
+       plane_id
+	   KMS plane ID to define the capture source.  Defaults to the first
+	   active plane found if neither crtc_id nor plane_id are specified.
+
+       framerate
+	   Framerate to capture at.  This is not synchronised to any page
+	   flipping or framebuffer changes - it just defines the interval at
+	   which the framebuffer is sampled.  Sampling faster than the
+	   framebuffer update rate will generate independent frames with the
+	   same content.  Defaults to 30.
+
+       Examples
+
+       o   Capture from the first active plane, download the result to normal
+	   frames and encode.  This will only work if the framebuffer is both
+	   linear and mappable - if not, the result may be scrambled or fail
+	   to download.
+
+		   ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
+
+       o   Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert
+	   to NV12 and encode as H.264.
+
+		   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
+
+       o   To capture only part of a plane the output can be cropped - this
+	   can be used to capture a single window, as long as it has a known
+	   absolute position and size.	For example, to capture and encode the
+	   middle quarter of a 1920x1080 plane:
+
+		   ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4
+
+   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.  The
+       filtergraph is specified through the option graph.
+
+       Options
+
+       graph
+	   Specify the filtergraph to use as input. Each video open output
+	   must be labelled by a unique string of the form "outN", where 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.
+
+	   The suffix "+subcc" can be appended to the output label to create
+	   an extra stream with the closed captions packets attached to that
+	   output (experimental; only for EIA-608 / CEA-708 for now).  The
+	   subcc streams are created after all the normal streams, in the
+	   order of the corresponding stream.  For example, if there is
+	   "out19+subcc", "out7+subcc" and up to "out42", the stream #43 is
+	   subcc for stream #7 and stream #44 is subcc for stream #19.
+
+	   If not specified defaults to the filename specified for the input
+	   device.
+
+       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 graph.
+
+       dumpgraph
+	   Dump graph to stderr.
+
+       Examples
+
+       o   Create a color video stream and play it back with ffplay:
+
+		   ffplay -f lavfi -graph "color=c=pink [out0]" dummy
+
+       o   As the previous example, but use filename for specifying the graph
+	   description, and omit the "out0" label:
+
+		   ffplay -f lavfi color=c=pink
+
+       o   Create three different video test filtered sources and play them:
+
+		   ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
+
+       o   Read an audio stream from a file using the amovie source and play
+	   it back with ffplay:
+
+		   ffplay -f lavfi "amovie=test.wav"
+
+       o   Read an audio stream and a video stream and play it back with
+	   ffplay:
+
+		   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
+
+       o   Dump decoded frames to images and closed captions to a file
+	   (experimental):
+
+		   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
+
+   libcdio
+       Audio-CD input device based on libcdio.
+
+       To enable this input device during configuration you need libcdio
+       installed on your system. It requires the configure option
+       "--enable-libcdio".
+
+       This device allows playing and grabbing from an Audio-CD.
+
+       For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you
+       may run the command:
+
+	       ffmpeg -f libcdio -i /dev/sr0 cd.wav
+
+       Options
+
+       speed
+	   Set drive reading speed. Default value is 0.
+
+	   The speed is specified CD-ROM speed units. The speed is set through
+	   the libcdio "cdio_cddap_speed_set" function. On many CD-ROM drives,
+	   specifying a value too large will result in using the fastest
+	   speed.
+
+       paranoia_mode
+	   Set paranoia recovery mode flags. It accepts one of the following
+	   values:
+
+	   disable
+	   verify
+	   overlap
+	   neverskip
+	   full
+
+	   Default value is disable.
+
+	   For more information about the available recovery modes, consult
+	   the paranoia project documentation.
+
+   libdc1394
+       IIDC1394 input device, based on libdc1394 and libraw1394.
+
+       Requires the configure option "--enable-libdc1394".
+
+       Options
+
+       framerate
+	   Set the frame rate. Default is "ntsc", corresponding to a frame
+	   rate of "30000/1001".
+
+       pixel_format
+	   Select the pixel format. Default is "uyvy422".
+
+       video_size
+	   Set the video size given as a string such as "640x480" or "hd720".
+	   Default is "qvga".
+
+   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 "--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
+       "--extra-cflags" and "--extra-ldflags" for allowing the build system to
+       locate the OpenAL headers and libraries.
+
+       An incomplete list of OpenAL implementations follows:
+
+       Creative
+	   The official Windows implementation, providing hardware
+	   acceleration with supported devices and software fallback.  See
+	   <http://openal.org/>.
+
+       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
+	   <http://kcat.strangesoft.net/openal.html>.
+
+       Apple
+	   OpenAL is part of Core Audio, the official Mac OS X Audio
+	   interface.  See
+	   <http://developer.apple.com/technologies/mac/audio-and-video.html>
+
+       This device allows one 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 list_devices.
+
+       Options
+
+       channels
+	   Set the number of channels in the captured audio. Only the values 1
+	   (monaural) and 2 (stereo) are currently supported.  Defaults to 2.
+
+       sample_size
+	   Set the sample size (in bits) of the captured audio. Only the
+	   values 8 and 16 are currently supported. Defaults to 16.
+
+       sample_rate
+	   Set the sample rate (in Hz) of the captured audio.  Defaults to
+	   44.1k.
+
+       list_devices
+	   If set to true, print a list of devices and exit.  Defaults to
+	   false.
+
+       Examples
+
+       Print the list of OpenAL supported devices and exit:
+
+	       $ ffmpeg -list_devices true -f openal -i dummy out.ogg
+
+       Capture from the OpenAL device DR-BT101 via PulseAudio:
+
+	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
+
+       Capture from the default device (note the empty string '' as filename):
+
+	       $ ffmpeg -f openal -i '' out.ogg
+
+       Capture from two devices simultaneously, writing to two different
+       files, within the same ffmpeg command:
+
+	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
+
+       Note: not all OpenAL implementations support multiple simultaneous
+       capture - try the latest OpenAL Soft if the above does not work.
+
+   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 /dev/dsp.
+
+       For example to grab from /dev/dsp using ffmpeg use the command:
+
+	       ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
+
+       For more information about OSS see:
+       <http://manuals.opensound.com/usersguide/dsp.html>
+
+       Options
+
+       sample_rate
+	   Set the sample rate in Hz. Default is 48000.
+
+       channels
+	   Set the number of channels. Default is 2.
+
+   pulse
+       PulseAudio input device.
+
+       To enable this output device you need to configure FFmpeg with
+       "--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 pactl list sources.
+
+       More information about PulseAudio can be found on
+       <http://www.pulseaudio.org>.
+
+       Options
+
+       server
+	   Connect to a specific PulseAudio server, specified by an IP
+	   address.  Default server is used when not provided.
+
+       name
+	   Specify the application name PulseAudio will use when showing
+	   active clients, by default it is the "LIBAVFORMAT_IDENT" string.
+
+       stream_name
+	   Specify the stream name PulseAudio will use when showing active
+	   streams, by default it is "record".
+
+       sample_rate
+	   Specify the samplerate in Hz, by default 48kHz is used.
+
+       channels
+	   Specify the channels in use, by default 2 (stereo) is set.
+
+       frame_size
+	   This option does nothing and is deprecated.
+
+       fragment_size
+	   Specify the size in bytes of the minimal buffering fragment in
+	   PulseAudio, it will affect the audio latency. By default it is set
+	   to 50 ms amount of data.
+
+       wallclock
+	   Set the initial PTS using the current time. Default is 1.
+
+       Examples
+
+       Record a stream from default device:
+
+	       ffmpeg -f pulse -i default /tmp/pulse.wav
+
+   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 /dev/audio0.
+
+       For example to grab from /dev/audio0 using ffmpeg use the command:
+
+	       ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
+
+       Options
+
+       sample_rate
+	   Set the sample rate in Hz. Default is 48000.
+
+       channels
+	   Set the number of channels. Default is 2.
+
+   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
+       "--enable-libv4l2" configure option), it is possible to use it with the
+       "-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
+       /dev/videoN, where N is a number associated to the device.
+
+       Video4Linux2 devices usually support a limited set of widthxheight
+       sizes and frame rates. You can check which are supported using
+       -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 -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 -timestamps abs or -ts abs option can be used to force
+       conversion into the real time clock.
+
+       Some usage examples of the video4linux2 device with ffmpeg and ffplay:
+
+       o   List supported formats for a video4linux2 device:
+
+		   ffplay -f video4linux2 -list_formats all /dev/video0
+
+       o   Grab and show the input of a video4linux2 device:
+
+		   ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
+
+       o   Grab and record the input of a video4linux2 device, leave the frame
+	   rate and size as previously set:
+
+		   ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
+
+       For more information about Video4Linux, check <http://linuxtv.org/>.
+
+       Options
+
+       standard
+	   Set the standard. Must be the name of a supported standard. To get
+	   a list of the supported standards, use the list_standards option.
+
+       channel
+	   Set the input channel number. Default to -1, which means using the
+	   previously selected channel.
+
+       video_size
+	   Set the video frame size. The argument must be a string in the form
+	   WIDTHxHEIGHT or a valid size abbreviation.
+
+       pixel_format
+	   Select the pixel format (only valid for raw video input).
+
+       input_format
+	   Set the preferred pixel format (for raw video) or a codec name.
+	   This option allows one to select the input format, when several are
+	   available.
+
+       framerate
+	   Set the preferred video frame rate.
+
+       list_formats
+	   List available formats (supported pixel formats, codecs, and frame
+	   sizes) and exit.
+
+	   Available values are:
+
+	   all Show all available (compressed and non-compressed) formats.
+
+	   raw Show only raw video (non-compressed) formats.
+
+	   compressed
+	       Show only compressed formats.
+
+       list_standards
+	   List supported standards and exit.
+
+	   Available values are:
+
+	   all Show all supported standards.
+
+       timestamps, ts
+	   Set type of timestamps for grabbed frames.
+
+	   Available values are:
+
+	   default
+	       Use timestamps from the kernel.
+
+	   abs Use absolute timestamps (wall clock).
+
+	   mono2abs
+	       Force conversion from monotonic to absolute timestamps.
+
+	   Default value is "default".
+
+       use_libv4l2
+	   Use libv4l2 (v4l-utils) conversion functions. Default is 0.
+
+   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.
+
+       Options
+
+       video_size
+	   Set the video frame size.
+
+       framerate
+	   Set the grabbing frame rate. Default value is "ntsc", corresponding
+	   to a frame rate of "30000/1001".
+
+   x11grab
+       X11 video input device.
+
+       To enable this input device during configuration you need libxcb
+       installed on your system. It will be automatically detected during
+       configuration.
+
+       This device allows one to capture a region of an X11 display.
+
+       The filename passed as input has the syntax:
+
+	       [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]
+
+       hostname:display_number.screen_number specifies the X11 display name of
+       the screen to grab from. hostname can be omitted, and defaults to
+       "localhost". The environment variable DISPLAY contains the default
+       display name.
+
+       x_offset and 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 xdpyinfo program for getting basic information about the
+       properties of your X11 display (e.g. grep for "name" or "dimensions").
+
+       For example to grab from :0.0 using ffmpeg:
+
+	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
+
+       Grab at position "10,20":
+
+	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+
+       Options
+
+       select_region
+	   Specify whether to select the grabbing area graphically using the
+	   pointer.  A value of 1 prompts the user to select the grabbing area
+	   graphically by clicking and dragging. A single click with no
+	   dragging will select the whole screen. A region with zero width or
+	   height will also select the whole screen. This option overwrites
+	   the video_size, grab_x, and grab_y options. Default value is 0.
+
+       draw_mouse
+	   Specify whether to draw the mouse pointer. A value of 0 specifies
+	   not to draw the pointer. Default value is 1.
+
+       follow_mouse
+	   Make the grabbed area follow the mouse. The argument can be
+	   "centered" or a number of pixels 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 PIXELS (greater than zero) to the edge of region.
+
+	   For example:
+
+		   ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
+
+	   To follow only when the mouse pointer reaches within 100 pixels to
+	   edge:
+
+		   ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
+
+       framerate
+	   Set the grabbing frame rate. Default value is "ntsc", corresponding
+	   to a frame rate of "30000/1001".
+
+       show_region
+	   Show grabbed region on screen.
+
+	   If show_region is specified with 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.
+
+       region_border
+	   Set the region border thickness if -show_region 1 is used.  Range
+	   is 1 to 128 and default is 3 (XCB-based x11grab only).
+
+	   For example:
+
+		   ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
+
+	   With follow_mouse:
+
+		   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
+
+       window_id
+	   Grab this window, instead of the whole screen. Default value is 0,
+	   which maps to the whole screen (root window).
+
+	   The id of a window can be found using the xwininfo program,
+	   possibly with options -tree and -root.
+
+	   If the window is later enlarged, the new area is not recorded.
+	   Video ends when the window is closed, unmapped (i.e., iconified) or
+	   shrunk beyond the video size (which defaults to the initial window
+	   size).
+
+	   This option disables options follow_mouse and select_region.
+
+       video_size
+	   Set the video frame size. Default is the full desktop or window.
+
+       grab_x
+       grab_y
+	   Set the grabbing region coordinates. They are expressed as offset
+	   from the top left corner of the X11 window and correspond to the
+	   x_offset and y_offset parameters in the device name. The default
+	   value for both options is 0.
+
+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=OUTDEV", or you can disable a particular input
+       device using the option "--disable-outdev=OUTDEV".
+
+       The option "-devices" of the ff* tools will display the list of enabled
+       output devices.
+
+       A description of the currently available output devices follows.
+
+   alsa
+       ALSA (Advanced Linux Sound Architecture) output device.
+
+       Examples
+
+       o   Play a file on default ALSA device:
+
+		   ffmpeg -i INPUT -f alsa default
+
+       o   Play a file on soundcard 1, audio device 7:
+
+		   ffmpeg -i INPUT -f alsa hw:1,7
+
+   AudioToolbox
+       AudioToolbox output device.
+
+       Allows native output to CoreAudio devices on OSX.
+
+       The output filename can be empty (or "-") to refer to the default
+       system output device or a number that refers to the device index as
+       shown using: "-list_devices true".
+
+       Alternatively, the audio input device can be chosen by index using the
+
+	   B<-audio_device_index E<lt>INDEXE<gt>>
+
+       , overriding any device name or index given in the input filename.
+
+       All available devices can be enumerated by using -list_devices true,
+       listing all device names, UIDs and corresponding indices.
+
+       Options
+
+       AudioToolbox supports the following options:
+
+       -audio_device_index <INDEX>
+	   Specify the audio device by its index. Overrides anything given in
+	   the output filename.
+
+       Examples
+
+       o   Print the list of supported devices and output a sine wave to the
+	   default device:
+
+		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -
+
+       o   Output a sine wave to the device with the index 2, overriding any
+	   output filename:
+
+		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -
+
+   caca
+       CACA output device.
+
+       This output device allows one 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
+       "--enable-libcaca".  libcaca is a graphics library that outputs text
+       instead of pixels.
+
+       For more information about libcaca, check:
+       <http://caca.zoy.org/wiki/libcaca>
+
+       Options
+
+       window_title
+	   Set the CACA window title, if not specified default to the filename
+	   specified for the output device.
+
+       window_size
+	   Set the CACA window size, can be a string of the form widthxheight
+	   or a video size abbreviation.  If not specified it defaults to the
+	   size of the input video.
+
+       driver
+	   Set display driver.
+
+       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 "-list_dither
+	   algorithms".
+
+       antialias
+	   Set antialias method. Antialiasing smoothens the rendered image and
+	   avoids the commonly seen staircase effect.  The accepted values are
+	   listed with "-list_dither antialiases".
+
+       charset
+	   Set which characters are going to be used when rendering text.  The
+	   accepted values are listed with "-list_dither charsets".
+
+       color
+	   Set color to be used when rendering text.  The accepted values are
+	   listed with "-list_dither colors".
+
+       list_drivers
+	   If set to true, print a list of available drivers and exit.
+
+       list_dither
+	   List available dither options related to the argument.  The
+	   argument must be one of "algorithms", "antialiases", "charsets",
+	   "colors".
+
+       Examples
+
+       o   The following command shows the ffmpeg output is an CACA window,
+	   forcing its size to 80x25:
+
+		   ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
+
+       o   Show the list of available drivers and exit:
+
+		   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
+
+       o   Show the list of available dither colors and exit:
+
+		   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
+
+   decklink
+       The decklink output device provides playback capabilities for
+       Blackmagic DeckLink devices.
+
+       To enable this output device, you need the Blackmagic DeckLink SDK and
+       you need to configure with the appropriate "--extra-cflags" and
+       "--extra-ldflags".  On Windows, you need to run the IDL files through
+       widl.
+
+       DeckLink is very picky about the formats it supports. Pixel format is
+       always uyvy422, framerate, field order and video size must be
+       determined for your device with -list_formats 1. Audio sample rate is
+       always 48 kHz.
+
+       Options
+
+       list_devices
+	   If set to true, print a list of devices and exit.  Defaults to
+	   false. This option is deprecated, please use the "-sinks" option of
+	   ffmpeg to list the available output devices.
+
+       list_formats
+	   If set to true, print a list of supported formats and exit.
+	   Defaults to false.
+
+       preroll
+	   Amount of time to preroll video in seconds.	Defaults to 0.5.
+
+       duplex_mode
+	   Sets the decklink device duplex/profile mode. Must be unset, half,
+	   full, one_sub_device_full, one_sub_device_half,
+	   two_sub_device_full, four_sub_device_half Defaults to unset.
+
+	   Note: DeckLink SDK 11.0 have replaced the duplex property by a
+	   profile property.  For the DeckLink Duo 2 and DeckLink Quad 2, a
+	   profile is shared between any 2 sub-devices that utilize the same
+	   connectors. For the DeckLink 8K Pro, a profile is shared between
+	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.
+
+	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
+	   one_sub_device_full, one_sub_device_half, two_sub_device_full,
+	   four_sub_device_half
+
+	   Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
+	   full
+
+       timing_offset
+	   Sets the genlock timing pixel offset on the used output.  Defaults
+	   to unset.
+
+       link
+	   Sets the SDI video link configuration on the used output. Must be
+	   unset, single link SDI, dual link SDI or quad link SDI.  Defaults
+	   to unset.
+
+       sqd Enable Square Division Quad Split mode for Quad-link SDI output.
+	   Must be unset, true or false.  Defaults to unset.
+
+       level_a
+	   Enable SMPTE Level A mode on the used output.  Must be unset, true
+	   or false.  Defaults to unset.
+
+       vanc_queue_size
+	   Sets maximum output buffer size in bytes for VANC data. If the
+	   buffering reaches this value, outgoing VANC data will be dropped.
+	   Defaults to 1048576.
+
+       Examples
+
+       o   List output devices:
+
+		   ffmpeg -sinks decklink
+
+       o   List supported formats:
+
+		   ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
+
+       o   Play video clip:
+
+		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
+
+       o   Play video clip with non-standard framerate or video size:
+
+		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
+
+   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 /dev/fb0.
+
+       For more detailed information read the file
+       Documentation/fb/framebuffer.txt included in the Linux source tree.
+
+       Options
+
+       xoffset
+       yoffset
+	   Set x/y coordinate of top left corner. Default is 0.
+
+       Examples
+
+       Play a file on framebuffer device /dev/fb0.  Required pixel format
+       depends on current framebuffer settings.
+
+	       ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
+
+       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).
+
+   opengl
+       OpenGL output device.
+
+       To enable this output device you need to configure FFmpeg with
+       "--enable-opengl".
+
+       This output device allows one to render to OpenGL context.  Context may
+       be provided by application or default SDL window is created.
+
+       When device renders to external context, application must implement
+       handlers for following messages: "AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" -
+       create OpenGL context on current thread.
+       "AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
+       "AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers.
+       "AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER" - destroy OpenGL context.
+       Application is also required to inform a device about current
+       resolution by sending "AV_APP_TO_DEV_WINDOW_SIZE" message.
+
+       Options
+
+       background
+	   Set background color. Black is a default.
+
+       no_window
+	   Disables default SDL window when set to non-zero value.
+	   Application must provide OpenGL context and both "window_size_cb"
+	   and "window_swap_buffers_cb" callbacks when set.
+
+       window_title
+	   Set the SDL window title, if not specified default to the filename
+	   specified for the output device.  Ignored when no_window is set.
+
+       window_size
+	   Set preferred window size, can be a string of the form widthxheight
+	   or a video size abbreviation.  If not specified it defaults to the
+	   size of the input video, downscaled according to the aspect ratio.
+	   Mostly usable when no_window is not set.
+
+       Examples
+
+       Play a file on SDL window using OpenGL rendering:
+
+	       ffmpeg  -i INPUT -f opengl "window title"
+
+   oss
+       OSS (Open Sound System) output device.
+
+   pulse
+       PulseAudio output device.
+
+       To enable this output device you need to configure FFmpeg with
+       "--enable-libpulse".
+
+       More information about PulseAudio can be found on
+       <http://www.pulseaudio.org>
+
+       Options
+
+       server
+	   Connect to a specific PulseAudio server, specified by an IP
+	   address.  Default server is used when not provided.
+
+       name
+	   Specify the application name PulseAudio will use when showing
+	   active clients, by default it is the "LIBAVFORMAT_IDENT" string.
+
+       stream_name
+	   Specify the stream name PulseAudio will use when showing active
+	   streams, by default it is set to the specified output name.
+
+       device
+	   Specify the device to use. Default device is used when not
+	   provided.  List of output devices can be obtained with command
+	   pactl list sinks.
+
+       buffer_size
+       buffer_duration
+	   Control the size and duration of the PulseAudio buffer. A small
+	   buffer gives more control, but requires more frequent updates.
+
+	   buffer_size specifies size in bytes while buffer_duration specifies
+	   duration in milliseconds.
+
+	   When both options are provided then the highest value is used
+	   (duration is recalculated to bytes using stream parameters). If
+	   they are set to 0 (which is default), the device will use the
+	   default PulseAudio duration value. By default PulseAudio set buffer
+	   duration to around 2 seconds.
+
+       prebuf
+	   Specify pre-buffering size in bytes. The server does not start with
+	   playback before at least prebuf bytes are available in the buffer.
+	   By default this option is initialized to the same value as
+	   buffer_size or buffer_duration (whichever is bigger).
+
+       minreq
+	   Specify minimum request size in bytes. The server does not request
+	   less than minreq bytes from the client, instead waits until the
+	   buffer is free enough to request more bytes at once. It is
+	   recommended to not set this option, which will initialize this to a
+	   value that is deemed sensible by the server.
+
+       Examples
+
+       Play a file on default device on default server:
+
+	       ffmpeg  -i INPUT -f pulse "stream name"
+
+   sdl
+       SDL (Simple DirectMedia Layer) output device.
+
+       "sdl2" can be used as alias for "sdl".
+
+       This output device allows one 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: <http://www.libsdl.org/>
+
+       Options
+
+       window_borderless
+	   Set SDL window border off.  Default value is 0 (enable window
+	   border).
+
+       window_enable_quit
+	   Enable quit action (using window button or keyboard key) when non-
+	   zero value is provided.  Default value is 1 (enable quit action).
+
+       window_fullscreen
+	   Set fullscreen mode when non-zero value is provided.  Default value
+	   is zero.
+
+       window_size
+	   Set the SDL window size, can be a string of the form widthxheight
+	   or a video size abbreviation.  If not specified it defaults to the
+	   size of the input video, downscaled according to the aspect ratio.
+
+       window_title
+	   Set the SDL window title, if not specified default to the filename
+	   specified for the output device.
+
+       window_x
+       window_y
+	   Set the position of the window on the screen.
+
+       Interactive commands
+
+       The window created by the device can be controlled through the
+       following interactive commands.
+
+       q, ESC
+	   Quit the device immediately.
+
+       Examples
+
+       The following command shows the ffmpeg output is an SDL window, forcing
+       its size to the qcif format:
+
+	       ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
+
+   sndio
+       sndio audio output device.
+
+   v4l2
+       Video4Linux2 output device.
+
+   xv
+       XV (XVideo) output device.
+
+       This output device allows one to show a video stream in a X Window
+       System window.
+
+       Options
+
+       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 hostname[:number[.screen_number]].
+
+	   hostname specifies the name of the host machine on which the
+	   display is physically attached. number specifies the number of the
+	   display server on that host machine. 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, "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.
+
+       window_id
+	   When set to non-zero value then device doesn't create new window,
+	   but uses existing one with provided window_id. By default this
+	   options is set to zero and device creates its own window.
+
+       window_size
+	   Set the created window size, can be a string of the form
+	   widthxheight or a video size abbreviation. If not specified it
+	   defaults to the size of the input video.  Ignored when window_id is
+	   set.
+
+       window_x
+       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.  Ignored when window_id is set.
+
+       window_title
+	   Set the window title, if not specified default to the filename
+	   specified for the output device. Ignored when window_id is set.
+
+       For more information about XVideo see <http://www.x.org/>.
+
+       Examples
+
+       o   Decode, display and encode video input with ffmpeg at the same
+	   time:
+
+		   ffmpeg -i INPUT OUTPUT -f xv display
+
+       o   Decode and display the input video to multiple X11 windows:
+
+		   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+							     FFMPEG-DEVICES(1)

ffmpeg-6.1-amd64-static/manpages/ffmpeg-protocols.txt

@@ -0,0 +1,1960 @@
+FFMPEG-PROTOCOLS(1)					   FFMPEG-PROTOCOLS(1)
+
+NAME
+       ffmpeg-protocols - FFmpeg protocols
+
+DESCRIPTION
+       This document describes the input and output protocols provided by the
+       libavformat library.
+
+PROTOCOL OPTIONS
+       The libavformat library provides some generic global options, which can
+       be set on all the protocols. In addition each protocol may support so-
+       called private options, which are specific for that component.
+
+       Options may be set by specifying -option value in the FFmpeg tools, or
+       by setting the value explicitly in the "AVFormatContext" options or
+       using the libavutil/opt.h API for programmatic use.
+
+       The list of supported options follows:
+
+       protocol_whitelist list (input)
+	   Set a ","-separated list of allowed protocols. "ALL" matches all
+	   protocols. Protocols prefixed by "-" are disabled.  All protocols
+	   are allowed by default but protocols used by an another protocol
+	   (nested protocols) are restricted to a per protocol subset.
+
+PROTOCOLS
+       Protocols are configured elements in FFmpeg that enable access to
+       resources that require specific protocols.
+
+       When you configure your FFmpeg build, all the supported protocols are
+       enabled by default. You can list all available ones using the configure
+       option "--list-protocols".
+
+       You can disable all the protocols using the configure option
+       "--disable-protocols", and selectively enable a protocol using the
+       option "--enable-protocol=PROTOCOL", or you can disable a particular
+       protocol using the option "--disable-protocol=PROTOCOL".
+
+       The option "-protocols" of the ff* tools will display the list of
+       supported protocols.
+
+       All protocols accept the following options:
+
+       rw_timeout
+	   Maximum time to wait for (network) read/write operations to
+	   complete, in microseconds.
+
+       A description of the currently available protocols follows.
+
+   amqp
+       Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker
+       based publish-subscribe communication protocol.
+
+       FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A
+       separate AMQP broker must also be run. An example open-source AMQP
+       broker is RabbitMQ.
+
+       After starting the broker, an FFmpeg client may stream data to the
+       broker using the command:
+
+	       ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]
+
+       Where hostname and port (default is 5672) is the address of the broker.
+       The client may also set a user/password for authentication. The default
+       for both fields is "guest". Name of virtual host on broker can be set
+       with vhost. The default value is "/".
+
+       Muliple subscribers may stream from the broker using the command:
+
+	       ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]
+
+       In RabbitMQ all data published to the broker flows through a specific
+       exchange, and each subscribing client has an assigned queue/buffer.
+       When a packet arrives at an exchange, it may be copied to a client's
+       queue depending on the exchange and routing_key fields.
+
+       The following options are supported:
+
+       exchange
+	   Sets the exchange to use on the broker. RabbitMQ has several
+	   predefined exchanges: "amq.direct" is the default exchange, where
+	   the publisher and subscriber must have a matching routing_key;
+	   "amq.fanout" is the same as a broadcast operation (i.e. the data is
+	   forwarded to all queues on the fanout exchange independent of the
+	   routing_key); and "amq.topic" is similar to "amq.direct", but
+	   allows for more complex pattern matching (refer to the RabbitMQ
+	   documentation).
+
+       routing_key
+	   Sets the routing key. The default value is "amqp". The routing key
+	   is used on the "amq.direct" and "amq.topic" exchanges to decide
+	   whether packets are written to the queue of a subscriber.
+
+       pkt_size
+	   Maximum size of each packet sent/received to the broker. Default is
+	   131072.  Minimum is 4096 and max is any large value (representable
+	   by an int). When receiving packets, this sets an internal buffer
+	   size in FFmpeg. It should be equal to or greater than the size of
+	   the published packets to the broker. Otherwise the received message
+	   may be truncated causing decoding errors.
+
+       connection_timeout
+	   The timeout in seconds during the initial connection to the broker.
+	   The default value is rw_timeout, or 5 seconds if rw_timeout is not
+	   set.
+
+       delivery_mode mode
+	   Sets the delivery mode of each message sent to broker.  The
+	   following values are accepted:
+
+	   persistent
+	       Delivery mode set to "persistent" (2). This is the default
+	       value.  Messages may be written to the broker's disk depending
+	       on its setup.
+
+	   non-persistent
+	       Delivery mode set to "non-persistent" (1).  Messages will stay
+	       in broker's memory unless the broker is under memory pressure.
+
+   async
+       Asynchronous data filling wrapper for input stream.
+
+       Fill data in a background thread, to decouple I/O operation from demux
+       thread.
+
+	       async:<URL>
+	       async:http://host/resource
+	       async:cache:http://host/resource
+
+   bluray
+       Read BluRay playlist.
+
+       The accepted options are:
+
+       angle
+	   BluRay angle
+
+       chapter
+	   Start chapter (1...N)
+
+       playlist
+	   Playlist to read (BDMV/PLAYLIST/?????.mpls)
+
+       Examples:
+
+       Read longest playlist from BluRay mounted to /mnt/bluray:
+
+	       bluray:/mnt/bluray
+
+       Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start
+       from chapter 2:
+
+	       -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
+
+   cache
+       Caching wrapper for input stream.
+
+       Cache the input stream to temporary file. It brings seeking capability
+       to live streams.
+
+       The accepted options are:
+
+       read_ahead_limit
+	   Amount in bytes that may be read ahead when seeking isn't
+	   supported. Range is -1 to INT_MAX.  -1 for unlimited. Default is
+	   65536.
+
+       URL Syntax is
+
+	       cache:<URL>
+
+   concat
+       Physical concatenation protocol.
+
+       Read and seek from many resources in sequence as if they were a unique
+       resource.
+
+       A URL accepted by this protocol has the syntax:
+
+	       concat:<URL1>|<URL2>|...|<URLN>
+
+       where URL1, URL2, ..., URLN are the urls of the resource to be
+       concatenated, each one possibly specifying a distinct protocol.
+
+       For example to read a sequence of files split1.mpeg, split2.mpeg,
+       split3.mpeg with ffplay use the command:
+
+	       ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
+
+       Note that you may need to escape the character "|" which is special for
+       many shells.
+
+   concatf
+       Physical concatenation protocol using a line break delimited list of
+       resources.
+
+       Read and seek from many resources in sequence as if they were a unique
+       resource.
+
+       A URL accepted by this protocol has the syntax:
+
+	       concatf:<URL>
+
+       where URL is the url containing a line break delimited list of
+       resources to be concatenated, each one possibly specifying a distinct
+       protocol. Special characters must be escaped with backslash or single
+       quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
+       manual.
+
+       For example to read a sequence of files split1.mpeg, split2.mpeg,
+       split3.mpeg listed in separate lines within a file split.txt with
+       ffplay use the command:
+
+	       ffplay concatf:split.txt
+
+       Where split.txt contains the lines:
+
+	       split1.mpeg
+	       split2.mpeg
+	       split3.mpeg
+
+   crypto
+       AES-encrypted stream reading protocol.
+
+       The accepted options are:
+
+       key Set the AES decryption key binary block from given hexadecimal
+	   representation.
+
+       iv  Set the AES decryption initialization vector binary block from
+	   given hexadecimal representation.
+
+       Accepted URL formats:
+
+	       crypto:<URL>
+	       crypto+<URL>
+
+   data
+       Data in-line in the URI. See
+       <http://en.wikipedia.org/wiki/Data_URI_scheme>.
+
+       For example, to convert a GIF file given inline with ffmpeg:
+
+	       ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
+
+   fd
+       File descriptor access protocol.
+
+       The accepted syntax is:
+
+	       fd: -fd <file_descriptor>
+
+       If fd is not specified, by default the stdout file descriptor will be
+       used for writing, stdin for reading. Unlike the pipe protocol, fd
+       protocol has seek support if it corresponding to a regular file. fd
+       protocol doesn't support pass file descriptor via URL for security.
+
+       This protocol accepts the following options:
+
+       blocksize
+	   Set I/O operation maximum block size, in bytes. Default value is
+	   "INT_MAX", which results in not limiting the requested block size.
+	   Setting this value reasonably low improves user termination request
+	   reaction time, which is valuable if data transmission is slow.
+
+       fd  Set file descriptor.
+
+   file
+       File access protocol.
+
+       Read from or write to a file.
+
+       A file URL can have the form:
+
+	       file:<filename>
+
+       where filename is the path of the file to read.
+
+       An URL that does not have a protocol prefix will be assumed to be a
+       file URL. Depending on the build, an URL that looks like a Windows path
+       with the drive letter at the beginning will also be assumed to be a
+       file URL (usually not the case in builds for unix-like systems).
+
+       For example to read from a file input.mpeg with ffmpeg use the command:
+
+	       ffmpeg -i file:input.mpeg output.mpeg
+
+       This protocol accepts the following options:
+
+       truncate
+	   Truncate existing files on write, if set to 1. A value of 0
+	   prevents truncating. Default value is 1.
+
+       blocksize
+	   Set I/O operation maximum block size, in bytes. Default value is
+	   "INT_MAX", which results in not limiting the requested block size.
+	   Setting this value reasonably low improves user termination request
+	   reaction time, which is valuable for files on slow medium.
+
+       follow
+	   If set to 1, the protocol will retry reading at the end of the
+	   file, allowing reading files that still are being written. In order
+	   for this to terminate, you either need to use the rw_timeout
+	   option, or use the interrupt callback (for API users).
+
+       seekable
+	   Controls if seekability is advertised on the file. 0 means non-
+	   seekable, -1 means auto (seekable for normal files, non-seekable
+	   for named pipes).
+
+	   Many demuxers handle seekable and non-seekable resources
+	   differently, overriding this might speed up opening certain files
+	   at the cost of losing some features (e.g. accurate seeking).
+
+   ftp
+       FTP (File Transfer Protocol).
+
+       Read from or write to remote resources using FTP protocol.
+
+       Following syntax is required.
+
+	       ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
+
+       This protocol accepts the following options.
+
+       timeout
+	   Set timeout in microseconds of socket I/O operations used by the
+	   underlying low level operation. By default it is set to -1, which
+	   means that the timeout is not specified.
+
+       ftp-user
+	   Set a user to be used for authenticating to the FTP server. This is
+	   overridden by the user in the FTP URL.
+
+       ftp-password
+	   Set a password to be used for authenticating to the FTP server.
+	   This is overridden by the password in the FTP URL, or by ftp-
+	   anonymous-password if no user is set.
+
+       ftp-anonymous-password
+	   Password used when login as anonymous user. Typically an e-mail
+	   address should be used.
+
+       ftp-write-seekable
+	   Control seekability of connection during encoding. If set to 1 the
+	   resource is supposed to be seekable, if set to 0 it is assumed not
+	   to be seekable. Default value is 0.
+
+       NOTE: Protocol can be used as output, but it is recommended to not do
+       it, unless special care is taken (tests, customized server
+       configuration etc.). Different FTP servers behave in different way
+       during seek operation. ff* tools may produce incomplete content due to
+       server limitations.
+
+   gopher
+       Gopher protocol.
+
+   gophers
+       Gophers protocol.
+
+       The Gopher protocol with TLS encapsulation.
+
+   hls
+       Read Apple HTTP Live Streaming compliant segmented stream as a uniform
+       one. The M3U8 playlists describing the segments can be remote HTTP
+       resources or local files, accessed using the standard file protocol.
+       The nested protocol is declared by specifying "+proto" after the hls
+       URI scheme name, where proto is either "file" or "http".
+
+	       hls+http://host/path/to/remote/resource.m3u8
+	       hls+file://path/to/local/resource.m3u8
+
+       Using this protocol is discouraged - the hls demuxer should work just
+       as well (if not, please report the issues) and is more complete.  To
+       use the hls demuxer instead, simply use the direct URLs to the m3u8
+       files.
+
+   http
+       HTTP (Hyper Text Transfer Protocol).
+
+       This protocol accepts the following options:
+
+       seekable
+	   Control seekability of connection. If set to 1 the resource is
+	   supposed to be seekable, if set to 0 it is assumed not to be
+	   seekable, if set to -1 it will try to autodetect if it is seekable.
+	   Default value is -1.
+
+       chunked_post
+	   If set to 1 use chunked Transfer-Encoding for posts, default is 1.
+
+       content_type
+	   Set a specific content type for the POST messages or for listen
+	   mode.
+
+       http_proxy
+	   set HTTP proxy to tunnel through e.g. http://example.com:1234
+
+       headers
+	   Set custom HTTP headers, can override built in default headers. The
+	   value must be a string encoding the headers.
+
+       multiple_requests
+	   Use persistent connections if set to 1, default is 0.
+
+       post_data
+	   Set custom HTTP post data.
+
+       referer
+	   Set the Referer header. Include 'Referer: URL' header in HTTP
+	   request.
+
+       user_agent
+	   Override the User-Agent header. If not specified the protocol will
+	   use a string describing the libavformat build. ("Lavf/<version>")
+
+       reconnect_at_eof
+	   If set then eof is treated like an error and causes reconnection,
+	   this is useful for live / endless streams.
+
+       reconnect_streamed
+	   If set then even streamed/non seekable streams will be reconnected
+	   on errors.
+
+       reconnect_on_network_error
+	   Reconnect automatically in case of TCP/TLS errors during connect.
+
+       reconnect_on_http_error
+	   A comma separated list of HTTP status codes to reconnect on. The
+	   list can include specific status codes (e.g. '503') or the strings
+	   '4xx' / '5xx'.
+
+       reconnect_delay_max
+	   Sets the maximum delay in seconds after which to give up
+	   reconnecting
+
+       mime_type
+	   Export the MIME type.
+
+       http_version
+	   Exports the HTTP response version number. Usually "1.0" or "1.1".
+
+       icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
+	   the server supports this, the metadata has to be retrieved by the
+	   application by reading the icy_metadata_headers and
+	   icy_metadata_packet options.  The default is 1.
+
+       icy_metadata_headers
+	   If the server supports ICY metadata, this contains the ICY-specific
+	   HTTP reply headers, separated by newline characters.
+
+       icy_metadata_packet
+	   If the server supports ICY metadata, and icy was set to 1, this
+	   contains the last non-empty metadata packet sent by the server. It
+	   should be polled in regular intervals by applications interested in
+	   mid-stream metadata updates.
+
+       cookies
+	   Set the cookies to be sent in future requests. The format of each
+	   cookie is the same as the value of a Set-Cookie HTTP response
+	   field. Multiple cookies can be delimited by a newline character.
+
+       offset
+	   Set initial byte offset.
+
+       end_offset
+	   Try to limit the request to bytes preceding this offset.
+
+       method
+	   When used as a client option it sets the HTTP method for the
+	   request.
+
+	   When used as a server option it sets the HTTP method that is going
+	   to be expected from the client(s).  If the expected and the
+	   received HTTP method do not match the client will be given a Bad
+	   Request response.  When unset the HTTP method is not checked for
+	   now. This will be replaced by autodetection in the future.
+
+       listen
+	   If set to 1 enables experimental HTTP server. This can be used to
+	   send data when used as an output option, or read data from a client
+	   with HTTP POST when used as an input option.  If set to 2 enables
+	   experimental multi-client HTTP server. This is not yet implemented
+	   in ffmpeg.c and thus must not be used as a command line option.
+
+		   # Server side (sending):
+		   ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
+
+		   # Client side (receiving):
+		   ffmpeg -i http://<server>:<port> -c copy somefile.ogg
+
+		   # Client can also be done with wget:
+		   wget http://<server>:<port> -O somefile.ogg
+
+		   # Server side (receiving):
+		   ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
+
+		   # Client side (sending):
+		   ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
+
+		   # Client can also be done with wget:
+		   wget --post-file=somefile.ogg http://<server>:<port>
+
+       send_expect_100
+	   Send an Expect: 100-continue header for POST. If set to 1 it will
+	   send, if set to 0 it won't, if set to -1 it will try to send if it
+	   is applicable. Default value is -1.
+
+       auth_type
+	   Set HTTP authentication type. No option for Digest, since this
+	   method requires getting nonce parameters from the server first and
+	   can't be used straight away like Basic.
+
+	   none
+	       Choose the HTTP authentication type automatically. This is the
+	       default.
+
+	   basic
+	       Choose the HTTP basic authentication.
+
+	       Basic authentication sends a Base64-encoded string that
+	       contains a user name and password for the client. Base64 is not
+	       a form of encryption and should be considered the same as
+	       sending the user name and password in clear text (Base64 is a
+	       reversible encoding).  If a resource needs to be protected,
+	       strongly consider using an authentication scheme other than
+	       basic authentication. HTTPS/TLS should be used with basic
+	       authentication.	Without these additional security
+	       enhancements, basic authentication should not be used to
+	       protect sensitive or valuable information.
+
+       HTTP Cookies
+
+       Some HTTP requests will be denied unless cookie values are passed in
+       with the request. The cookies option allows these cookies to be
+       specified. At the very least, each cookie must specify a value along
+       with a path and domain.	HTTP requests that match both the domain and
+       path will automatically include the cookie value in the HTTP Cookie
+       header field. Multiple cookies can be delimited by a newline.
+
+       The required syntax to play a stream specifying a cookie is:
+
+	       ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
+
+   Icecast
+       Icecast protocol (stream to Icecast servers)
+
+       This protocol accepts the following options:
+
+       ice_genre
+	   Set the stream genre.
+
+       ice_name
+	   Set the stream name.
+
+       ice_description
+	   Set the stream description.
+
+       ice_url
+	   Set the stream website URL.
+
+       ice_public
+	   Set if the stream should be public.	The default is 0 (not public).
+
+       user_agent
+	   Override the User-Agent header. If not specified a string of the
+	   form "Lavf/<version>" will be used.
+
+       password
+	   Set the Icecast mountpoint password.
+
+       content_type
+	   Set the stream content type. This must be set if it is different
+	   from audio/mpeg.
+
+       legacy_icecast
+	   This enables support for Icecast versions < 2.4.0, that do not
+	   support the HTTP PUT method but the SOURCE method.
+
+       tls Establish a TLS (HTTPS) connection to Icecast.
+
+	       icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
+
+   ipfs
+       InterPlanetary File System (IPFS) protocol support. One can access
+       files stored on the IPFS network through so-called gateways. These are
+       http(s) endpoints.  This protocol wraps the IPFS native protocols
+       (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
+       should) host their own node which means this protocol will use one's
+       local gateway to access files on the IPFS network.
+
+       This protocol accepts the following options:
+
+       gateway
+	   Defines the gateway to use. When not set, the protocol will first
+	   try locating the local gateway by looking at $IPFS_GATEWAY,
+	   $IPFS_PATH and "$HOME/.ipfs/", in that order.
+
+       One can use this protocol in 2 ways. Using IPFS:
+
+	       ffplay ipfs://<hash>
+
+       Or the IPNS protocol (IPNS is mutable IPFS):
+
+	       ffplay ipns://<hash>
+
+   mmst
+       MMS (Microsoft Media Server) protocol over TCP.
+
+   mmsh
+       MMS (Microsoft Media Server) protocol over HTTP.
+
+       The required syntax is:
+
+	       mmsh://<server>[:<port>][/<app>][/<playpath>]
+
+   md5
+       MD5 output protocol.
+
+       Computes the MD5 hash of the data to be written, and on close writes
+       this to the designated output or stdout if none is specified. It can be
+       used to test muxers without writing an actual file.
+
+       Some examples follow.
+
+	       # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
+	       ffmpeg -i input.flv -f avi -y md5:output.avi.md5
+
+	       # Write the MD5 hash of the encoded AVI file to stdout.
+	       ffmpeg -i input.flv -f avi -y md5:
+
+       Note that some formats (typically MOV) require the output protocol to
+       be seekable, so they will fail with the MD5 output protocol.
+
+   pipe
+       UNIX pipe access protocol.
+
+       Read and write from UNIX pipes.
+
+       The accepted syntax is:
+
+	       pipe:[<number>]
+
+       If fd isn't specified, number is the number corresponding to the file
+       descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr).
+       If number is not specified, by default the stdout file descriptor will
+       be used for writing, stdin for reading.
+
+       For example to read from stdin with ffmpeg:
+
+	       cat test.wav | ffmpeg -i pipe:0
+	       # ...this is the same as...
+	       cat test.wav | ffmpeg -i pipe:
+
+       For writing to stdout with ffmpeg:
+
+	       ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
+	       # ...this is the same as...
+	       ffmpeg -i test.wav -f avi pipe: | cat > test.avi
+
+       This protocol accepts the following options:
+
+       blocksize
+	   Set I/O operation maximum block size, in bytes. Default value is
+	   "INT_MAX", which results in not limiting the requested block size.
+	   Setting this value reasonably low improves user termination request
+	   reaction time, which is valuable if data transmission is slow.
+
+       fd  Set file descriptor.
+
+       Note that some formats (typically MOV), require the output protocol to
+       be seekable, so they will fail with the pipe output protocol.
+
+   prompeg
+       Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
+
+       The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
+       mechanism for MPEG-2 Transport Streams sent over RTP.
+
+       This protocol must be used in conjunction with the "rtp_mpegts" muxer
+       and the "rtp" protocol.
+
+       The required syntax is:
+
+	       -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
+
+       The destination UDP ports are "port + 2" for the column FEC stream and
+       "port + 4" for the row FEC stream.
+
+       This protocol accepts the following options:
+
+       l=n The number of columns (4-20, LxD <= 100)
+
+       d=n The number of rows (4-20, LxD <= 100)
+
+       Example usage:
+
+	       -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
+
+   rist
+       Reliable Internet Streaming Transport protocol
+
+       The accepted options are:
+
+       rist_profile
+	   Supported values:
+
+	   simple
+	   main
+	       This one is default.
+
+	   advanced
+       buffer_size
+	   Set internal RIST buffer size in milliseconds for retransmission of
+	   data.  Default value is 0 which means the librist default (1 sec).
+	   Maximum value is 30 seconds.
+
+       fifo_size
+	   Size of the librist receiver output fifo in number of packets. This
+	   must be a power of 2.  Defaults to 8192 (vs the librist default of
+	   1024).
+
+       overrun_nonfatal=1|0
+	   Survive in case of librist fifo buffer overrun. Default value is 0.
+
+       pkt_size
+	   Set maximum packet size for sending data. 1316 by default.
+
+       log_level
+	   Set loglevel for RIST logging messages. You only need to set this
+	   if you explicitly want to enable debug level messages or packet
+	   loss simulation, otherwise the regular loglevel is respected.
+
+       secret
+	   Set override of encryption secret, by default is unset.
+
+       encryption
+	   Set encryption type, by default is disabled.  Acceptable values are
+	   128 and 256.
+
+   rtmp
+       Real-Time Messaging Protocol.
+
+       The Real-Time Messaging Protocol (RTMP) is used for streaming
+       multimedia content across a TCP/IP network.
+
+       The required syntax is:
+
+	       rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
+
+       The accepted parameters are:
+
+       username
+	   An optional username (mostly for publishing).
+
+       password
+	   An optional password (mostly for publishing).
+
+       server
+	   The address of the RTMP server.
+
+       port
+	   The number of the TCP port to use (by default is 1935).
+
+       app It is the name of the application to access. It usually corresponds
+	   to the path where the application is installed on the RTMP server
+	   (e.g. /ondemand/, /flash/live/, etc.). You can override the value
+	   parsed from the URI through the "rtmp_app" option, too.
+
+       playpath
+	   It is the path or name of the resource to play with reference to
+	   the application specified in app, may be prefixed by "mp4:". You
+	   can override the value parsed from the URI through the
+	   "rtmp_playpath" option, too.
+
+       listen
+	   Act as a server, listening for an incoming connection.
+
+       timeout
+	   Maximum time to wait for the incoming connection. Implies listen.
+
+       Additionally, the following parameters can be set via command line
+       options (or in code via "AVOption"s):
+
+       rtmp_app
+	   Name of application to connect on the RTMP server. This option
+	   overrides the parameter specified in the URI.
+
+       rtmp_buffer
+	   Set the client buffer time in milliseconds. The default is 3000.
+
+       rtmp_conn
+	   Extra arbitrary AMF connection parameters, parsed from a string,
+	   e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each
+	   value is prefixed by a single character denoting the type, B for
+	   Boolean, N for number, S for string, O for object, or Z for null,
+	   followed by a colon. For Booleans the data must be either 0 or 1
+	   for FALSE or TRUE, respectively.  Likewise for Objects the data
+	   must be 0 or 1 to end or begin an object, respectively. Data items
+	   in subobjects may be named, by prefixing the type with 'N' and
+	   specifying the name before the value (i.e. "NB:myFlag:1"). This
+	   option may be used multiple times to construct arbitrary AMF
+	   sequences.
+
+       rtmp_enhanced_codecs
+	   Specify the list of codecs the client advertises to support in an
+	   enhanced RTMP stream. This option should be set to a comma
+	   separated list of fourcc values, like "hvc1,av01,vp09" for multiple
+	   codecs or "hvc1" for only one codec. The specified list will be
+	   presented in the "fourCcLive" property of the Connect Command
+	   Message.
+
+       rtmp_flashver
+	   Version of the Flash plugin used to run the SWF player. The default
+	   is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
+	   (compatible; <libavformat version>).)
+
+       rtmp_flush_interval
+	   Number of packets flushed in the same request (RTMPT only). The
+	   default is 10.
+
+       rtmp_live
+	   Specify that the media is a live stream. No resuming or seeking in
+	   live streams is possible. The default value is "any", which means
+	   the subscriber first tries to play the live stream specified in the
+	   playpath. If a live stream of that name is not found, it plays the
+	   recorded stream. The other possible values are "live" and
+	   "recorded".
+
+       rtmp_pageurl
+	   URL of the web page in which the media was embedded. By default no
+	   value will be sent.
+
+       rtmp_playpath
+	   Stream identifier to play or to publish. This option overrides the
+	   parameter specified in the URI.
+
+       rtmp_subscribe
+	   Name of live stream to subscribe to. By default no value will be
+	   sent.  It is only sent if the option is specified or if rtmp_live
+	   is set to live.
+
+       rtmp_swfhash
+	   SHA256 hash of the decompressed SWF file (32 bytes).
+
+       rtmp_swfsize
+	   Size of the decompressed SWF file, required for SWFVerification.
+
+       rtmp_swfurl
+	   URL of the SWF player for the media. By default no value will be
+	   sent.
+
+       rtmp_swfverify
+	   URL to player swf file, compute hash/size automatically.
+
+       rtmp_tcurl
+	   URL of the target stream. Defaults to proto://host[:port]/app.
+
+       tcp_nodelay=1|0
+	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
+
+	   Remark: Writing to the socket is currently not optimized to
+	   minimize system calls and reduces the efficiency / effect of
+	   TCP_NODELAY.
+
+       For example to read with ffplay a multimedia resource named "sample"
+       from the application "vod" from an RTMP server "myserver":
+
+	       ffplay rtmp://myserver/vod/sample
+
+       To publish to a password protected server, passing the playpath and app
+       names separately:
+
+	       ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
+
+   rtmpe
+       Encrypted Real-Time Messaging Protocol.
+
+       The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
+       streaming multimedia content within standard cryptographic primitives,
+       consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
+       pair of RC4 keys.
+
+   rtmps
+       Real-Time Messaging Protocol over a secure SSL connection.
+
+       The Real-Time Messaging Protocol (RTMPS) is used for streaming
+       multimedia content across an encrypted connection.
+
+   rtmpt
+       Real-Time Messaging Protocol tunneled through HTTP.
+
+       The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
+       for streaming multimedia content within HTTP requests to traverse
+       firewalls.
+
+   rtmpte
+       Encrypted Real-Time Messaging Protocol tunneled through HTTP.
+
+       The Encrypted Real-Time Messaging Protocol tunneled through HTTP
+       (RTMPTE) is used for streaming multimedia content within HTTP requests
+       to traverse firewalls.
+
+   rtmpts
+       Real-Time Messaging Protocol tunneled through HTTPS.
+
+       The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
+       used for streaming multimedia content within HTTPS requests to traverse
+       firewalls.
+
+   libsmbclient
+       libsmbclient permits one to manipulate CIFS/SMB network resources.
+
+       Following syntax is required.
+
+	       smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
+
+       This protocol accepts the following options.
+
+       timeout
+	   Set timeout in milliseconds of socket I/O operations used by the
+	   underlying low level operation. By default it is set to -1, which
+	   means that the timeout is not specified.
+
+       truncate
+	   Truncate existing files on write, if set to 1. A value of 0
+	   prevents truncating. Default value is 1.
+
+       workgroup
+	   Set the workgroup used for making connections. By default workgroup
+	   is not specified.
+
+       For more information see: <http://www.samba.org/>.
+
+   libssh
+       Secure File Transfer Protocol via libssh
+
+       Read from or write to remote resources using SFTP protocol.
+
+       Following syntax is required.
+
+	       sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
+
+       This protocol accepts the following options.
+
+       timeout
+	   Set timeout of socket I/O operations used by the underlying low
+	   level operation. By default it is set to -1, which means that the
+	   timeout is not specified.
+
+       truncate
+	   Truncate existing files on write, if set to 1. A value of 0
+	   prevents truncating. Default value is 1.
+
+       private_key
+	   Specify the path of the file containing private key to use during
+	   authorization.  By default libssh searches for keys in the ~/.ssh/
+	   directory.
+
+       Example: Play a file stored on remote server.
+
+	       ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
+
+   librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
+       Real-Time Messaging Protocol and its variants supported through
+       librtmp.
+
+       Requires the presence of the librtmp headers and library during
+       configuration. You need to explicitly configure the build with
+       "--enable-librtmp". If enabled this will replace the native RTMP
+       protocol.
+
+       This protocol provides most client functions and a few server functions
+       needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
+       (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these
+       encrypted types (RTMPTE, RTMPTS).
+
+       The required syntax is:
+
+	       <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
+
+       where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
+       "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
+       server, port, app and playpath have the same meaning as specified for
+       the RTMP native protocol.  options contains a list of space-separated
+       options of the form key=val.
+
+       See the librtmp manual page (man 3 librtmp) for more information.
+
+       For example, to stream a file in real-time to an RTMP server using
+       ffmpeg:
+
+	       ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
+
+       To play the same stream using ffplay:
+
+	       ffplay "rtmp://myserver/live/mystream live=1"
+
+   rtp
+       Real-time Transport Protocol.
+
+       The required syntax for an RTP URL is:
+       rtp://hostname[:port][?option=val...]
+
+       port specifies the RTP port to use.
+
+       The following URL options are supported:
+
+       ttl=n
+	   Set the TTL (Time-To-Live) value (for multicast only).
+
+       rtcpport=n
+	   Set the remote RTCP port to n.
+
+       localrtpport=n
+	   Set the local RTP port to n.
+
+       localrtcpport=n'
+	   Set the local RTCP port to n.
+
+       pkt_size=n
+	   Set max packet size (in bytes) to n.
+
+       buffer_size=size
+	   Set the maximum UDP socket buffer size in bytes.
+
+       connect=0|1
+	   Do a "connect()" on the UDP socket (if set to 1) or not (if set to
+	   0).
+
+       sources=ip[,ip]
+	   List allowed source IP addresses.
+
+       block=ip[,ip]
+	   List disallowed (blocked) source IP addresses.
+
+       write_to_source=0|1
+	   Send packets to the source address of the latest received packet
+	   (if set to 1) or to a default remote address (if set to 0).
+
+       localport=n
+	   Set the local RTP port to n.
+
+       localaddr=addr
+	   Local IP address of a network interface used for sending packets or
+	   joining multicast groups.
+
+       timeout=n
+	   Set timeout (in microseconds) of socket I/O operations to n.
+
+	   This is a deprecated option. Instead, localrtpport should be used.
+
+       Important notes:
+
+       1.  If rtcpport is not set the RTCP port will be set to the RTP port
+	   value plus 1.
+
+       2.  If localrtpport (the local RTP port) is not set any available port
+	   will be used for the local RTP and RTCP ports.
+
+       3.  If localrtcpport (the local RTCP port) is not set it will be set to
+	   the local RTP port value plus 1.
+
+   rtsp
+       Real-Time Streaming Protocol.
+
+       RTSP is not technically a protocol handler in libavformat, it is a
+       demuxer and muxer. The demuxer supports both normal RTSP (with data
+       transferred over RTP; this is used by e.g. Apple and Microsoft) and
+       Real-RTSP (with data transferred over RDT).
+
+       The muxer can be used to send a stream using RTSP ANNOUNCE to a server
+       supporting it (currently Darwin Streaming Server and Mischa
+       Spiegelmock's <https://github.com/revmischa/rtsp-server>).
+
+       The required syntax for a RTSP url is:
+
+	       rtsp://<hostname>[:<port>]/<path>
+
+       Options can be set on the ffmpeg/ffplay command line, or set in code
+       via "AVOption"s or in "avformat_open_input".
+
+       Muxer
+
+       The following options are supported.
+
+       rtsp_transport
+	   Set RTSP transport protocols.
+
+	   It accepts the following values:
+
+	   udp Use UDP as lower transport protocol.
+
+	   tcp Use TCP (interleaving within the RTSP control channel) as lower
+	       transport protocol.
+
+	   Default value is 0.
+
+       rtsp_flags
+	   Set RTSP flags.
+
+	   The following values are accepted:
+
+	   latm
+	       Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.
+
+	   rfc2190
+	       Use RFC 2190 packetization instead of RFC 4629 for H.263.
+
+	   skip_rtcp
+	       Don't send RTCP sender reports.
+
+	   h264_mode0
+	       Use mode 0 for H.264 in RTP.
+
+	   send_bye
+	       Send RTCP BYE packets when finishing.
+
+	   Default value is 0.
+
+       min_port
+	   Set minimum local UDP port. Default value is 5000.
+
+       max_port
+	   Set maximum local UDP port. Default value is 65000.
+
+       buffer_size
+	   Set the maximum socket buffer size in bytes.
+
+       pkt_size
+	   Set max send packet size (in bytes). Default value is 1472.
+
+       Demuxer
+
+       The following options are supported.
+
+       initial_pause
+	   Do not start playing the stream immediately if set to 1. Default
+	   value is 0.
+
+       rtsp_transport
+	   Set RTSP transport protocols.
+
+	   It accepts the following values:
+
+	   udp Use UDP as lower transport protocol.
+
+	   tcp Use TCP (interleaving within the RTSP control channel) as lower
+	       transport protocol.
+
+	   udp_multicast
+	       Use UDP multicast as lower transport protocol.
+
+	   http
+	       Use HTTP tunneling as lower transport protocol, which is useful
+	       for passing proxies.
+
+	   https
+	       Use HTTPs tunneling as lower transport protocol, which is
+	       useful for passing proxies and widely used for security
+	       consideration.
+
+	   Multiple lower transport protocols may be specified, in that case
+	   they are tried one at a time (if the setup of one fails, the next
+	   one is tried).  For the muxer, only the tcp and udp options are
+	   supported.
+
+       rtsp_flags
+	   Set RTSP flags.
+
+	   The following values are accepted:
+
+	   filter_src
+	       Accept packets only from negotiated peer address and port.
+
+	   listen
+	       Act as a server, listening for an incoming connection.
+
+	   prefer_tcp
+	       Try TCP for RTP transport first, if TCP is available as RTSP
+	       RTP transport.
+
+	   satip_raw
+	       Export raw MPEG-TS stream instead of demuxing. The flag will
+	       simply write out the raw stream, with the original PAT/PMT/PIDs
+	       intact.
+
+	   Default value is none.
+
+       allowed_media_types
+	   Set media types to accept from the server.
+
+	   The following flags are accepted:
+
+	   video
+	   audio
+	   data
+	   subtitle
+
+	   By default it accepts all media types.
+
+       min_port
+	   Set minimum local UDP port. Default value is 5000.
+
+       max_port
+	   Set maximum local UDP port. Default value is 65000.
+
+       listen_timeout
+	   Set maximum timeout (in seconds) to establish an initial
+	   connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
+	   Default is -1 which means an infinite timeout when listen mode is
+	   set.
+
+       reorder_queue_size
+	   Set number of packets to buffer for handling of reordered packets.
+
+       timeout
+	   Set socket TCP I/O timeout in microseconds.
+
+       user_agent
+	   Override User-Agent header. If not specified, it defaults to the
+	   libavformat identifier string.
+
+       buffer_size
+	   Set the maximum socket buffer size in bytes.
+
+       When receiving data over UDP, the demuxer tries to reorder received
+       packets (since they may arrive out of order, or packets may get lost
+       totally). This can be disabled by setting the maximum demuxing delay to
+       zero (via the "max_delay" field of AVFormatContext).
+
+       When watching multi-bitrate Real-RTSP streams with ffplay, the streams
+       to display can be chosen with "-vst" n and "-ast" n for video and audio
+       respectively, and can be switched on the fly by pressing "v" and "a".
+
+       Examples
+
+       The following examples all make use of the ffplay and ffmpeg tools.
+
+       o   Watch a stream over UDP, with a max reordering delay of 0.5
+	   seconds:
+
+		   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
+
+       o   Watch a stream tunneled over HTTP:
+
+		   ffplay -rtsp_transport http rtsp://server/video.mp4
+
+       o   Send a stream in realtime to a RTSP server, for others to watch:
+
+		   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
+
+       o   Receive a stream in realtime:
+
+		   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
+
+   sap
+       Session Announcement Protocol (RFC 2974). This is not technically a
+       protocol handler in libavformat, it is a muxer and demuxer.  It is used
+       for signalling of RTP streams, by announcing the SDP for the streams
+       regularly on a separate port.
+
+       Muxer
+
+       The syntax for a SAP url given to the muxer is:
+
+	       sap://<destination>[:<port>][?<options>]
+
+       The RTP packets are sent to destination on port port, or to port 5004
+       if no port is specified.  options is a "&"-separated list. The
+       following options are supported:
+
+       announce_addr=address
+	   Specify the destination IP address for sending the announcements
+	   to.	If omitted, the announcements are sent to the commonly used
+	   SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
+	   or ff0e::2:7ffe if destination is an IPv6 address.
+
+       announce_port=port
+	   Specify the port to send the announcements on, defaults to 9875 if
+	   not specified.
+
+       ttl=ttl
+	   Specify the time to live value for the announcements and RTP
+	   packets, defaults to 255.
+
+       same_port=0|1
+	   If set to 1, send all RTP streams on the same port pair. If zero
+	   (the default), all streams are sent on unique ports, with each
+	   stream on a port 2 numbers higher than the previous.  VLC/Live555
+	   requires this to be set to 1, to be able to receive the stream.
+	   The RTP stack in libavformat for receiving requires all streams to
+	   be sent on unique ports.
+
+       Example command lines follow.
+
+       To broadcast a stream on the local subnet, for watching in VLC:
+
+	       ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1
+
+       Similarly, for watching in ffplay:
+
+	       ffmpeg -re -i <input> -f sap sap://224.0.0.255
+
+       And for watching in ffplay, over IPv6:
+
+	       ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
+
+       Demuxer
+
+       The syntax for a SAP url given to the demuxer is:
+
+	       sap://[<address>][:<port>]
+
+       address is the multicast address to listen for announcements on, if
+       omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
+       port that is listened on, 9875 if omitted.
+
+       The demuxers listens for announcements on the given address and port.
+       Once an announcement is received, it tries to receive that particular
+       stream.
+
+       Example command lines follow.
+
+       To play back the first stream announced on the normal SAP multicast
+       address:
+
+	       ffplay sap://
+
+       To play back the first stream announced on one the default IPv6 SAP
+       multicast address:
+
+	       ffplay sap://[ff0e::2:7ffe]
+
+   sctp
+       Stream Control Transmission Protocol.
+
+       The accepted URL syntax is:
+
+	       sctp://<host>:<port>[?<options>]
+
+       The protocol accepts the following options:
+
+       listen
+	   If set to any value, listen for an incoming connection. Outgoing
+	   connection is done by default.
+
+       max_streams
+	   Set the maximum number of streams. By default no limit is set.
+
+   srt
+       Haivision Secure Reliable Transport Protocol via libsrt.
+
+       The supported syntax for a SRT URL is:
+
+	       srt://<hostname>:<port>[?<options>]
+
+       options contains a list of &-separated options of the form key=val.
+
+       or
+
+	       <options> srt://<hostname>:<port>
+
+       options contains a list of '-key val' options.
+
+       This protocol accepts the following options.
+
+       connect_timeout=milliseconds
+	   Connection timeout; SRT cannot connect for RTT > 1500 msec (2
+	   handshake exchanges) with the default connect timeout of 3 seconds.
+	   This option applies to the caller and rendezvous connection modes.
+	   The connect timeout is 10 times the value set for the rendezvous
+	   mode (which can be used as a workaround for this connection problem
+	   with earlier versions).
+
+       ffs=bytes
+	   Flight Flag Size (Window Size), in bytes. FFS is actually an
+	   internal parameter and you should set it to not less than
+	   recv_buffer_size and mss. The default value is relatively large,
+	   therefore unless you set a very large receiver buffer, you do not
+	   need to change this option. Default value is 25600.
+
+       inputbw=bytes/seconds
+	   Sender nominal input rate, in bytes per seconds. Used along with
+	   oheadbw, when maxbw is set to relative (0), to calculate maximum
+	   sending rate when recovery packets are sent along with the main
+	   media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
+	   while maxbw is set to relative (0), the actual input rate is
+	   evaluated inside the library. Default value is 0.
+
+       iptos=tos
+	   IP Type of Service. Applies to sender only. Default value is 0xB8.
+
+       ipttl=ttl
+	   IP Time To Live. Applies to sender only. Default value is 64.
+
+       latency=microseconds
+	   Timestamp-based Packet Delivery Delay.  Used to absorb bursts of
+	   missed packet retransmissions.  This flag sets both rcvlatency and
+	   peerlatency to the same value. Note that prior to version 1.3.0
+	   this is the only flag to set the latency, however this is
+	   effectively equivalent to setting peerlatency, when side is sender
+	   and rcvlatency when side is receiver, and the bidirectional stream
+	   sending is not supported.
+
+       listen_timeout=microseconds
+	   Set socket listen timeout.
+
+       maxbw=bytes/seconds
+	   Maximum sending bandwidth, in bytes per seconds.  -1 infinite
+	   (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
+	   absolute limit value Default value is 0 (relative)
+
+       mode=caller|listener|rendezvous
+	   Connection mode.  caller opens client connection.  listener starts
+	   server to listen for incoming connections.  rendezvous use Rendez-
+	   Vous connection mode.  Default value is caller.
+
+       mss=bytes
+	   Maximum Segment Size, in bytes. Used for buffer allocation and rate
+	   calculation using a packet counter assuming fully filled packets.
+	   The smallest MSS between the peers is used. This is 1500 by default
+	   in the overall internet.  This is the maximum size of the UDP
+	   packet and can be only decreased, unless you have some unusual
+	   dedicated network settings. Default value is 1500.
+
+       nakreport=1|0
+	   If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
+	   periodically until a lost packet is retransmitted or intentionally
+	   dropped. Default value is 1.
+
+       oheadbw=percents
+	   Recovery bandwidth overhead above input rate, in percents.  See
+	   inputbw. Default value is 25%.
+
+       passphrase=string
+	   HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
+	   79 characters. The passphrase is the shared secret between the
+	   sender and the receiver. It is used to generate the Key Encrypting
+	   Key using PBKDF2 (Password-Based Key Derivation Function). It is
+	   used only if pbkeylen is non-zero. It is used on the receiver only
+	   if the received data is encrypted.  The configured passphrase
+	   cannot be recovered (write-only).
+
+       enforced_encryption=1|0
+	   If true, both connection parties must have the same password set
+	   (including empty, that is, with no encryption). If the password
+	   doesn't match or only one side is unencrypted, the connection is
+	   rejected. Default is true.
+
+       kmrefreshrate=packets
+	   The number of packets to be transmitted after which the encryption
+	   key is switched to a new key. Default is -1.  -1 means auto
+	   (0x1000000 in srt library). The range for this option is integers
+	   in the 0 - "INT_MAX".
+
+       kmpreannounce=packets
+	   The interval between when a new encryption key is sent and when
+	   switchover occurs. This value also applies to the subsequent
+	   interval between when switchover occurs and when the old encryption
+	   key is decommissioned. Default is -1.  -1 means auto (0x1000 in srt
+	   library). The range for this option is integers in the 0 -
+	   "INT_MAX".
+
+       snddropdelay=microseconds
+	   The sender's extra delay before dropping packets. This delay is
+	   added to the default drop delay time interval value.
+
+	   Special value -1: Do not drop packets on the sender at all.
+
+       payload_size=bytes
+	   Sets the maximum declared size of a packet transferred during the
+	   single call to the sending function in Live mode. Use 0 if this
+	   value isn't used (which is default in file mode).  Default is -1
+	   (automatic), which typically means MPEG-TS; if you are going to use
+	   SRT to send any different kind of payload, such as, for example,
+	   wrapping a live stream in very small frames, then you can use a
+	   bigger maximum frame size, though not greater than 1456 bytes.
+
+       pkt_size=bytes
+	   Alias for payload_size.
+
+       peerlatency=microseconds
+	   The latency value (as described in rcvlatency) that is set by the
+	   sender side as a minimum value for the receiver.
+
+       pbkeylen=bytes
+	   Sender encryption key length, in bytes.  Only can be set to 0, 16,
+	   24 and 32.  Enable sender encryption if not 0.  Not required on
+	   receiver (set to 0), key size obtained from sender in HaiCrypt
+	   handshake.  Default value is 0.
+
+       rcvlatency=microseconds
+	   The time that should elapse since the moment when the packet was
+	   sent and the moment when it's delivered to the receiver application
+	   in the receiving function.  This time should be a buffer time large
+	   enough to cover the time spent for sending, unexpectedly extended
+	   RTT time, and the time needed to retransmit the lost UDP packet.
+	   The effective latency value will be the maximum of this options'
+	   value and the value of peerlatency set by the peer side. Before
+	   version 1.3.0 this option is only available as latency.
+
+       recv_buffer_size=bytes
+	   Set UDP receive buffer size, expressed in bytes.
+
+       send_buffer_size=bytes
+	   Set UDP send buffer size, expressed in bytes.
+
+       timeout=microseconds
+	   Set raise error timeouts for read, write and connect operations.
+	   Note that the SRT library has internal timeouts which can be
+	   controlled separately, the value set here is only a cap on those.
+
+       tlpktdrop=1|0
+	   Too-late Packet Drop. When enabled on receiver, it skips missing
+	   packets that have not been delivered in time and delivers the
+	   following packets to the application when their time-to-play has
+	   come. It also sends a fake ACK to the sender. When enabled on
+	   sender and enabled on the receiving peer, the sender drops the
+	   older packets that have no chance of being delivered in time. It
+	   was automatically enabled in the sender if the receiver supports
+	   it.
+
+       sndbuf=bytes
+	   Set send buffer size, expressed in bytes.
+
+       rcvbuf=bytes
+	   Set receive buffer size, expressed in bytes.
+
+	   Receive buffer must not be greater than ffs.
+
+       lossmaxttl=packets
+	   The value up to which the Reorder Tolerance may grow. When Reorder
+	   Tolerance is > 0, then packet loss report is delayed until that
+	   number of packets come in. Reorder Tolerance increases every time a
+	   "belated" packet has come, but it wasn't due to retransmission
+	   (that is, when UDP packets tend to come out of order), with the
+	   difference between the latest sequence and this packet's sequence,
+	   and not more than the value of this option. By default it's 0,
+	   which means that this mechanism is turned off, and the loss report
+	   is always sent immediately upon experiencing a "gap" in sequences.
+
+       minversion
+	   The minimum SRT version that is required from the peer. A
+	   connection to a peer that does not satisfy the minimum version
+	   requirement will be rejected.
+
+	   The version format in hex is 0xXXYYZZ for x.y.z in human readable
+	   form.
+
+       streamid=string
+	   A string limited to 512 characters that can be set on the socket
+	   prior to connecting. This stream ID will be able to be retrieved by
+	   the listener side from the socket that is returned from srt_accept
+	   and was connected by a socket with that set stream ID. SRT does not
+	   enforce any special interpretation of the contents of this string.
+	   This option doesnXt make sense in Rendezvous connection; the result
+	   might be that simply one side will override the value from the
+	   other side and itXs the matter of luck which one would win
+
+       srt_streamid=string
+	   Alias for streamid to avoid conflict with ffmpeg command line
+	   option.
+
+       smoother=live|file
+	   The type of Smoother used for the transmission for that socket,
+	   which is responsible for the transmission and congestion control.
+	   The Smoother type must be exactly the same on both connecting
+	   parties, otherwise the connection is rejected.
+
+       messageapi=1|0
+	   When set, this socket uses the Message API, otherwise it uses
+	   Buffer API. Note that in live mode (see transtype) thereXs only
+	   message API available. In File mode you can chose to use one of two
+	   modes:
+
+	   Stream API (default, when this option is false). In this mode you
+	   may send as many data as you wish with one sending instruction, or
+	   even use dedicated functions that read directly from a file. The
+	   internal facility will take care of any speed and congestion
+	   control. When receiving, you can also receive as many data as
+	   desired, the data not extracted will be waiting for the next call.
+	   There is no boundary between data portions in the Stream mode.
+
+	   Message API. In this mode your single sending instruction passes
+	   exactly one piece of data that has boundaries (a message). Contrary
+	   to Live mode, this message may span across multiple UDP packets and
+	   the only size limitation is that it shall fit as a whole in the
+	   sending buffer. The receiver shall use as large buffer as necessary
+	   to receive the message, otherwise the message will not be given up.
+	   When the message is not complete (not all packets received or there
+	   was a packet loss) it will not be given up.
+
+       transtype=live|file
+	   Sets the transmission type for the socket, in particular, setting
+	   this option sets multiple other parameters to their default values
+	   as required for a particular transmission type.
+
+	   live: Set options as for live transmission. In this mode, you
+	   should send by one sending instruction only so many data that fit
+	   in one UDP packet, and limited to the value defined first in
+	   payload_size (1316 is default in this mode). There is no speed
+	   control in this mode, only the bandwidth control, if configured, in
+	   order to not exceed the bandwidth with the overhead transmission
+	   (retransmitted and control packets).
+
+	   file: Set options as for non-live transmission. See messageapi for
+	   further explanations
+
+       linger=seconds
+	   The number of seconds that the socket waits for unsent data when
+	   closing.  Default is -1. -1 means auto (off with 0 seconds in live
+	   mode, on with 180 seconds in file mode). The range for this option
+	   is integers in the 0 - "INT_MAX".
+
+       tsbpd=1|0
+	   When true, use Timestamp-based Packet Delivery mode. The default
+	   behavior depends on the transmission type: enabled in live mode,
+	   disabled in file mode.
+
+       For more information see: <https://github.com/Haivision/srt>.
+
+   srtp
+       Secure Real-time Transport Protocol.
+
+       The accepted options are:
+
+       srtp_in_suite
+       srtp_out_suite
+	   Select input and output encoding suites.
+
+	   Supported values:
+
+	   AES_CM_128_HMAC_SHA1_80
+	   SRTP_AES128_CM_HMAC_SHA1_80
+	   AES_CM_128_HMAC_SHA1_32
+	   SRTP_AES128_CM_HMAC_SHA1_32
+       srtp_in_params
+       srtp_out_params
+	   Set input and output encoding parameters, which are expressed by a
+	   base64-encoded representation of a binary block. The first 16 bytes
+	   of this binary block are used as master key, the following 14 bytes
+	   are used as master salt.
+
+   subfile
+       Virtually extract a segment of a file or another stream.  The
+       underlying stream must be seekable.
+
+       Accepted options:
+
+       start
+	   Start offset of the extracted segment, in bytes.
+
+       end End offset of the extracted segment, in bytes.  If set to 0,
+	   extract till end of file.
+
+       Examples:
+
+       Extract a chapter from a DVD VOB file (start and end sectors obtained
+       externally and multiplied by 2048):
+
+	       subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
+
+       Play an AVI file directly from a TAR archive:
+
+	       subfile,,start,183241728,end,366490624,,:archive.tar
+
+       Play a MPEG-TS file from start offset till end:
+
+	       subfile,,start,32815239,end,0,,:video.ts
+
+   tee
+       Writes the output to multiple protocols. The individual outputs are
+       separated by |
+
+	       tee:file://path/to/local/this.avi|file://path/to/local/that.avi
+
+   tcp
+       Transmission Control Protocol.
+
+       The required syntax for a TCP url is:
+
+	       tcp://<hostname>:<port>[?<options>]
+
+       options contains a list of &-separated options of the form key=val.
+
+       The list of supported options follows.
+
+       listen=2|1|0
+	   Listen for an incoming connection. 0 disables listen, 1 enables
+	   listen in single client mode, 2 enables listen in multi-client
+	   mode. Default value is 0.
+
+       local_addr=addr
+	   Local IP address of a network interface used for tcp socket
+	   connect.
+
+       local_port=port
+	   Local port used for tcp socket connect.
+
+       timeout=microseconds
+	   Set raise error timeout, expressed in microseconds.
+
+	   This option is only relevant in read mode: if no data arrived in
+	   more than this time interval, raise error.
+
+       listen_timeout=milliseconds
+	   Set listen timeout, expressed in milliseconds.
+
+       recv_buffer_size=bytes
+	   Set receive buffer size, expressed bytes.
+
+       send_buffer_size=bytes
+	   Set send buffer size, expressed bytes.
+
+       tcp_nodelay=1|0
+	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
+
+	   Remark: Writing to the socket is currently not optimized to
+	   minimize system calls and reduces the efficiency / effect of
+	   TCP_NODELAY.
+
+       tcp_mss=bytes
+	   Set maximum segment size for outgoing TCP packets, expressed in
+	   bytes.
+
+       The following example shows how to setup a listening TCP connection
+       with ffmpeg, which is then accessed with ffplay:
+
+	       ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
+	       ffplay tcp://<hostname>:<port>
+
+   tls
+       Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
+
+       The required syntax for a TLS/SSL url is:
+
+	       tls://<hostname>:<port>[?<options>]
+
+       The following parameters can be set via command line options (or in
+       code via "AVOption"s):
+
+       ca_file, cafile=filename
+	   A file containing certificate authority (CA) root certificates to
+	   treat as trusted. If the linked TLS library contains a default this
+	   might not need to be specified for verification to work, but not
+	   all libraries and setups have defaults built in.  The file must be
+	   in OpenSSL PEM format.
+
+       tls_verify=1|0
+	   If enabled, try to verify the peer that we are communicating with.
+	   Note, if using OpenSSL, this currently only makes sure that the
+	   peer certificate is signed by one of the root certificates in the
+	   CA database, but it does not validate that the certificate actually
+	   matches the host name we are trying to connect to. (With other
+	   backends, the host name is validated as well.)
+
+	   This is disabled by default since it requires a CA database to be
+	   provided by the caller in many cases.
+
+       cert_file, cert=filename
+	   A file containing a certificate to use in the handshake with the
+	   peer.  (When operating as server, in listen mode, this is more
+	   often required by the peer, while client certificates only are
+	   mandated in certain setups.)
+
+       key_file, key=filename
+	   A file containing the private key for the certificate.
+
+       listen=1|0
+	   If enabled, listen for connections on the provided port, and assume
+	   the server role in the handshake instead of the client role.
+
+       http_proxy
+	   The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
+	   The proxy must support the CONNECT method.
+
+       Example command lines:
+
+       To create a TLS/SSL server that serves an input stream.
+
+	       ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
+
+       To play back a stream from the TLS/SSL server using ffplay:
+
+	       ffplay tls://<hostname>:<port>
+
+   udp
+       User Datagram Protocol.
+
+       The required syntax for an UDP URL is:
+
+	       udp://<hostname>:<port>[?<options>]
+
+       options contains a list of &-separated options of the form key=val.
+
+       In case threading is enabled on the system, a circular buffer is used
+       to store the incoming data, which allows one to reduce loss of data due
+       to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
+       options are related to this buffer.
+
+       The list of supported options follows.
+
+       buffer_size=size
+	   Set the UDP maximum socket buffer size in bytes. This is used to
+	   set either the receive or send buffer size, depending on what the
+	   socket is used for.	Default is 32 KB for output, 384 KB for input.
+	   See also fifo_size.
+
+       bitrate=bitrate
+	   If set to nonzero, the output will have the specified constant
+	   bitrate if the input has enough packets to sustain it.
+
+       burst_bits=bits
+	   When using bitrate this specifies the maximum number of bits in
+	   packet bursts.
+
+       localport=port
+	   Override the local UDP port to bind with.
+
+       localaddr=addr
+	   Local IP address of a network interface used for sending packets or
+	   joining multicast groups.
+
+       pkt_size=size
+	   Set the size in bytes of UDP packets.
+
+       reuse=1|0
+	   Explicitly allow or disallow reusing UDP sockets.
+
+       ttl=ttl
+	   Set the time to live value (for multicast only).
+
+       connect=1|0
+	   Initialize the UDP socket with "connect()". In this case, the
+	   destination address can't be changed with ff_udp_set_remote_url
+	   later.  If the destination address isn't known at the start, this
+	   option can be specified in ff_udp_set_remote_url, too.  This allows
+	   finding out the source address for the packets with getsockname,
+	   and makes writes return with AVERROR(ECONNREFUSED) if "destination
+	   unreachable" is received.  For receiving, this gives the benefit of
+	   only receiving packets from the specified peer address/port.
+
+       sources=address[,address]
+	   Only receive packets sent from the specified addresses. In case of
+	   multicast, also subscribe to multicast traffic coming from these
+	   addresses only.
+
+       block=address[,address]
+	   Ignore packets sent from the specified addresses. In case of
+	   multicast, also exclude the source addresses in the multicast
+	   subscription.
+
+       fifo_size=units
+	   Set the UDP receiving circular buffer size, expressed as a number
+	   of packets with size of 188 bytes. If not specified defaults to
+	   7*4096.
+
+       overrun_nonfatal=1|0
+	   Survive in case of UDP receiving circular buffer overrun. Default
+	   value is 0.
+
+       timeout=microseconds
+	   Set raise error timeout, expressed in microseconds.
+
+	   This option is only relevant in read mode: if no data arrived in
+	   more than this time interval, raise error.
+
+       broadcast=1|0
+	   Explicitly allow or disallow UDP broadcasting.
+
+	   Note that broadcasting may not work properly on networks having a
+	   broadcast storm protection.
+
+       Examples
+
+       o   Use ffmpeg to stream over UDP to a remote endpoint:
+
+		   ffmpeg -i <input> -f <format> udp://<hostname>:<port>
+
+       o   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
+	   packets, using a large input buffer:
+
+		   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
+
+       o   Use ffmpeg to receive over UDP from a remote endpoint:
+
+		   ffmpeg -i udp://[<multicast-address>]:<port> ...
+
+   unix
+       Unix local socket
+
+       The required syntax for a Unix socket URL is:
+
+	       unix://<filepath>
+
+       The following parameters can be set via command line options (or in
+       code via "AVOption"s):
+
+       timeout
+	   Timeout in ms.
+
+       listen
+	   Create the Unix socket in listening mode.
+
+   zmq
+       ZeroMQ asynchronous messaging using the libzmq library.
+
+       This library supports unicast streaming to multiple clients without
+       relying on an external server.
+
+       The required syntax for streaming or connecting to a stream is:
+
+	       zmq:tcp://ip-address:port
+
+       Example: Create a localhost stream on port 5555:
+
+	       ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555
+
+       Multiple clients may connect to the stream using:
+
+	       ffplay zmq:tcp://127.0.0.1:5555
+
+       Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
+       pattern.  The server side binds to a port and publishes data. Clients
+       connect to the server (via IP address/port) and subscribe to the
+       stream. The order in which the server and client start generally does
+       not matter.
+
+       ffmpeg must be compiled with the --enable-libzmq option to support this
+       protocol.
+
+       Options can be set on the ffmpeg/ffplay command line. The following
+       options are supported:
+
+       pkt_size
+	   Forces the maximum packet size for sending/receiving data. The
+	   default value is 131,072 bytes. On the server side, this sets the
+	   maximum size of sent packets via ZeroMQ. On the clients, it sets an
+	   internal buffer size for receiving packets. Note that pkt_size on
+	   the clients should be equal to or greater than pkt_size on the
+	   server. Otherwise the received message may be truncated causing
+	   decoding errors.
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+							   FFMPEG-PROTOCOLS(1)

ffmpeg-6.1-amd64-static/manpages/ffmpeg-resampler.txt

@@ -0,0 +1,259 @@
+FFMPEG-RESAMPLER(1)					   FFMPEG-RESAMPLER(1)
+
+NAME
+       ffmpeg-resampler - FFmpeg Resampler
+
+DESCRIPTION
+       The FFmpeg resampler provides a high-level interface to the
+       libswresample library audio resampling utilities. In particular it
+       allows one to perform audio resampling, audio channel layout
+       rematrixing, and convert audio format and packing layout.
+
+RESAMPLER OPTIONS
+       The audio resampler supports the following named options.
+
+       Options may be set by specifying -option value in the FFmpeg tools,
+       option=value for the aresample filter, by setting the value explicitly
+       in the "SwrContext" options or using the libavutil/opt.h API for
+       programmatic use.
+
+       uchl, used_chlayout
+	   Set used input channel layout. Default is unset. This option is
+	   only used for special remapping.
+
+       isr, in_sample_rate
+	   Set the input sample rate. Default value is 0.
+
+       osr, out_sample_rate
+	   Set the output sample rate. Default value is 0.
+
+       isf, in_sample_fmt
+	   Specify the input sample format. It is set by default to "none".
+
+       osf, out_sample_fmt
+	   Specify the output sample format. It is set by default to "none".
+
+       tsf, internal_sample_fmt
+	   Set the internal sample format. Default value is "none".  This will
+	   automatically be chosen when it is not explicitly set.
+
+       ichl, in_chlayout
+       ochl, out_chlayout
+	   Set the input/output channel layout.
+
+	   See the Channel Layout section in the ffmpeg-utils(1) manual for
+	   the required syntax.
+
+       clev, center_mix_level
+	   Set the center mix level. It is a value expressed in deciBel, and
+	   must be in the interval [-32,32].
+
+       slev, surround_mix_level
+	   Set the surround mix level. It is a value expressed in deciBel, and
+	   must be in the interval [-32,32].
+
+       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].
+
+       rmvol, rematrix_volume
+	   Set rematrix volume. Default value is 1.0.
+
+       rematrix_maxval
+	   Set maximum output value for rematrixing.  This can be used to
+	   prevent clipping vs. preventing volume reduction.  A value of 1.0
+	   prevents clipping.
+
+       flags, swr_flags
+	   Set flags used by the converter. Default value is 0.
+
+	   It supports the following individual flags:
+
+	   res force resampling, this flag forces resampling to be used even
+	       when the input and output sample rates match.
+
+       dither_scale
+	   Set the dither scale. Default value is 1.
+
+       dither_method
+	   Set dither method. Default value is 0.
+
+	   Supported values:
+
+	   rectangular
+	       select rectangular dither
+
+	   triangular
+	       select triangular dither
+
+	   triangular_hp
+	       select triangular dither with high pass
+
+	   lipshitz
+	       select Lipshitz noise shaping dither.
+
+	   shibata
+	       select Shibata noise shaping dither.
+
+	   low_shibata
+	       select low Shibata noise shaping dither.
+
+	   high_shibata
+	       select high Shibata noise shaping dither.
+
+	   f_weighted
+	       select f-weighted noise shaping dither
+
+	   modified_e_weighted
+	       select modified-e-weighted noise shaping dither
+
+	   improved_e_weighted
+	       select improved-e-weighted noise shaping dither
+
+       resampler
+	   Set resampling engine. Default value is swr.
+
+	   Supported values:
+
+	   swr select the native SW Resampler; filter options precision and
+	       cheby are not applicable in this case.
+
+	   soxr
+	       select the SoX Resampler (where available); compensation, and
+	       filter options filter_size, phase_shift, exact_rational,
+	       filter_type & kaiser_beta, are not applicable in this case.
+
+       filter_size
+	   For swr only, set resampling filter size, default value is 32.
+
+       phase_shift
+	   For swr only, set resampling phase shift, default value is 10, and
+	   must be in the interval [0,30].
+
+       linear_interp
+	   Use linear interpolation when enabled (the default). Disable it if
+	   you want to preserve speed instead of quality when exact_rational
+	   fails.
+
+       exact_rational
+	   For swr only, when enabled, try to use exact phase_count based on
+	   input and output sample rate. However, if it is larger than "1 <<
+	   phase_shift", the phase_count will be "1 << phase_shift" as
+	   fallback. Default is enabled.
+
+       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).
+
+       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'.
+
+       cheby
+	   For soxr only, selects passband rolloff none (Chebyshev) & higher-
+	   precision approximation for 'irrational' ratios. Default value is
+	   0.
+
+       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.
+
+       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.
+
+       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
+	   (min_comp = "FLT_MAX").
+
+       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 min_comp.  The default is 0.1.
+
+       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.
+
+       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.
+
+       matrix_encoding
+	   Select matrixed stereo encoding.
+
+	   It accepts the following values:
+
+	   none
+	       select none
+
+	   dolby
+	       select Dolby
+
+	   dplii
+	       select Dolby Pro Logic II
+
+	   Default value is "none".
+
+       filter_type
+	   For swr only, select resampling filter type. This only affects
+	   resampling operations.
+
+	   It accepts the following values:
+
+	   cubic
+	       select cubic
+
+	   blackman_nuttall
+	       select Blackman Nuttall windowed sinc
+
+	   kaiser
+	       select Kaiser windowed sinc
+
+       kaiser_beta
+	   For swr only, set Kaiser window beta value. Must be a double float
+	   value in the interval [2,16], default value is 9.
+
+       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.
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+							   FFMPEG-RESAMPLER(1)

ffmpeg-6.1-amd64-static/manpages/ffmpeg-scaler.txt

@@ -0,0 +1,156 @@
+FFMPEG-SCALER(1)					      FFMPEG-SCALER(1)
+
+NAME
+       ffmpeg-scaler - FFmpeg video scaling and pixel format converter
+
+DESCRIPTION
+       The FFmpeg rescaler provides a high-level interface to the libswscale
+       library image conversion utilities. In particular it allows one to
+       perform image rescaling and pixel format conversion.
+
+SCALER OPTIONS
+       The video scaler supports the following named options.
+
+       Options may be set by specifying -option value in the FFmpeg tools,
+       with a few API-only exceptions noted below.  For programmatic use, they
+       can be set explicitly in the "SwsContext" options or through the
+       libavutil/opt.h API.
+
+       sws_flags
+	   Set the scaler flags. This is also used to set the scaling
+	   algorithm. Only a single algorithm should be selected. Default
+	   value is bicubic.
+
+	   It accepts the following values:
+
+	   fast_bilinear
+	       Select fast bilinear scaling algorithm.
+
+	   bilinear
+	       Select bilinear scaling algorithm.
+
+	   bicubic
+	       Select bicubic scaling algorithm.
+
+	   experimental
+	       Select experimental scaling algorithm.
+
+	   neighbor
+	       Select nearest neighbor rescaling algorithm.
+
+	   area
+	       Select averaging area rescaling algorithm.
+
+	   bicublin
+	       Select bicubic scaling algorithm for the luma component,
+	       bilinear for chroma components.
+
+	   gauss
+	       Select Gaussian rescaling algorithm.
+
+	   sinc
+	       Select sinc rescaling algorithm.
+
+	   lanczos
+	       Select Lanczos rescaling algorithm. The default width (alpha)
+	       is 3 and can be changed by setting "param0".
+
+	   spline
+	       Select natural bicubic spline rescaling algorithm.
+
+	   print_info
+	       Enable printing/debug logging.
+
+	   accurate_rnd
+	       Enable accurate rounding.
+
+	   full_chroma_int
+	       Enable full chroma interpolation.
+
+	   full_chroma_inp
+	       Select full chroma input.
+
+	   bitexact
+	       Enable bitexact output.
+
+       srcw (API only)
+	   Set source width.
+
+       srch (API only)
+	   Set source height.
+
+       dstw (API only)
+	   Set destination width.
+
+       dsth (API only)
+	   Set destination height.
+
+       src_format (API only)
+	   Set source pixel format (must be expressed as an integer).
+
+       dst_format (API only)
+	   Set destination pixel format (must be expressed as an integer).
+
+       src_range (boolean)
+	   If value is set to 1, indicates source is full range. Default value
+	   is 0, which indicates source is limited range.
+
+       dst_range (boolean)
+	   If value is set to 1, enable full range for destination. Default
+	   value is 0, which enables limited range.
+
+       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.
+
+       sws_dither
+	   Set the dithering algorithm. Accepts one of the following values.
+	   Default value is auto.
+
+	   auto
+	       automatic choice
+
+	   none
+	       no dithering
+
+	   bayer
+	       bayer dither
+
+	   ed  error diffusion dither
+
+	   a_dither
+	       arithmetic dither, based using addition
+
+	   x_dither
+	       arithmetic dither, based using xor (more random/less apparent
+	       patterning that a_dither).
+
+       alphablend
+	   Set the alpha blending to use when the input has alpha but the
+	   output does not.  Default value is none.
+
+	   uniform_color
+	       Blend onto a uniform background color
+
+	   checkerboard
+	       Blend onto a checkerboard
+
+	   none
+	       No blending
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+							      FFMPEG-SCALER(1)

ffmpeg-6.1-amd64-static/manpages/ffmpeg-utils.txt

@@ -0,0 +1,1256 @@
+FFMPEG-UTILS(1) 					       FFMPEG-UTILS(1)
+
+NAME
+       ffmpeg-utils - FFmpeg utilities
+
+DESCRIPTION
+       This document describes some generic features and utilities provided by
+       the libavutil library.
+
+SYNTAX
+       This section documents the syntax and formats employed by the FFmpeg
+       libraries and tools.
+
+   Quoting and escaping
+       FFmpeg adopts the following quoting and escaping mechanism, unless
+       explicitly specified. The following rules are applied:
+
+       o   ' and \ are special characters (respectively used for quoting and
+	   escaping). In addition to them, there might be other special
+	   characters depending on the specific syntax where the escaping and
+	   quoting are employed.
+
+       o   A special character is escaped by prefixing it with a \.
+
+       o   All characters enclosed between '' are included literally in the
+	   parsed string. The quote character ' itself cannot be quoted, so
+	   you may need to close the quote and escape it.
+
+       o   Leading and trailing whitespaces, unless escaped or quoted, are
+	   removed from the parsed string.
+
+       Note that you may need to add a second level of escaping when using the
+       command line or a script, which depends on the syntax of the adopted
+       shell language.
+
+       The function "av_get_token" defined in libavutil/avstring.h can be used
+       to parse a token quoted or escaped according to the rules defined
+       above.
+
+       The tool tools/ffescape in the FFmpeg source tree can be used to
+       automatically quote or escape a string in a script.
+
+       Examples
+
+       o   Escape the string "Crime d'Amour" containing the "'" special
+	   character:
+
+		   Crime d\'Amour
+
+       o   The string above contains a quote, so the "'" needs to be escaped
+	   when quoting it:
+
+		   'Crime d'\''Amour'
+
+       o   Include leading or trailing whitespaces using quoting:
+
+		   '  this string starts and ends with whitespaces  '
+
+       o   Escaping and quoting can be mixed together:
+
+		   ' The string '\'string\'' is a string '
+
+       o   To include a literal \ you can use either escaping or quoting:
+
+		   'c:\foo' can be written as c:\\foo
+
+   Date
+       The accepted syntax is:
+
+	       [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
+	       now
+
+       If the value is "now" it takes the current time.
+
+       Time is local time unless Z is appended, in which case it is
+       interpreted as UTC.  If the year-month-day part is not specified it
+       takes the current year-month-day.
+
+   Time duration
+       There are two accepted syntaxes for expressing time duration.
+
+	       [-][<HH>:]<MM>:<SS>[.<m>...]
+
+       HH expresses the number of hours, MM the number of minutes for a
+       maximum of 2 digits, and SS the number of seconds for a maximum of 2
+       digits. The m at the end expresses decimal value for SS.
+
+       or
+
+	       [-]<S>+[.<m>...][s|ms|us]
+
+       S expresses the number of seconds, with the optional decimal part m.
+       The optional literal suffixes s, ms or us indicate to interpret the
+       value as seconds, milliseconds or microseconds, respectively.
+
+       In both expressions, the optional - indicates negative duration.
+
+       Examples
+
+       The following examples are all valid time duration:
+
+       55  55 seconds
+
+       0.2 0.2 seconds
+
+       200ms
+	   200 milliseconds, that's 0.2s
+
+       200000us
+	   200000 microseconds, that's 0.2s
+
+       12:03:45
+	   12 hours, 03 minutes and 45 seconds
+
+       23.189
+	   23.189 seconds
+
+   Video size
+       Specify the size of the sourced video, it may be a string of the form
+       widthxheight, or the name of a size abbreviation.
+
+       The following abbreviations are recognized:
+
+       ntsc
+	   720x480
+
+       pal 720x576
+
+       qntsc
+	   352x240
+
+       qpal
+	   352x288
+
+       sntsc
+	   640x480
+
+       spal
+	   768x576
+
+       film
+	   352x240
+
+       ntsc-film
+	   352x240
+
+       sqcif
+	   128x96
+
+       qcif
+	   176x144
+
+       cif 352x288
+
+       4cif
+	   704x576
+
+       16cif
+	   1408x1152
+
+       qqvga
+	   160x120
+
+       qvga
+	   320x240
+
+       vga 640x480
+
+       svga
+	   800x600
+
+       xga 1024x768
+
+       uxga
+	   1600x1200
+
+       qxga
+	   2048x1536
+
+       sxga
+	   1280x1024
+
+       qsxga
+	   2560x2048
+
+       hsxga
+	   5120x4096
+
+       wvga
+	   852x480
+
+       wxga
+	   1366x768
+
+       wsxga
+	   1600x1024
+
+       wuxga
+	   1920x1200
+
+       woxga
+	   2560x1600
+
+       wqsxga
+	   3200x2048
+
+       wquxga
+	   3840x2400
+
+       whsxga
+	   6400x4096
+
+       whuxga
+	   7680x4800
+
+       cga 320x200
+
+       ega 640x350
+
+       hd480
+	   852x480
+
+       hd720
+	   1280x720
+
+       hd1080
+	   1920x1080
+
+       2k  2048x1080
+
+       2kflat
+	   1998x1080
+
+       2kscope
+	   2048x858
+
+       4k  4096x2160
+
+       4kflat
+	   3996x2160
+
+       4kscope
+	   4096x1716
+
+       nhd 640x360
+
+       hqvga
+	   240x160
+
+       wqvga
+	   400x240
+
+       fwqvga
+	   432x240
+
+       hvga
+	   480x320
+
+       qhd 960x540
+
+       2kdci
+	   2048x1080
+
+       4kdci
+	   4096x2160
+
+       uhd2160
+	   3840x2160
+
+       uhd4320
+	   7680x4320
+
+   Video rate
+       Specify the frame rate of a video, expressed as the number of frames
+       generated per second. It has to be a string in the format
+       frame_rate_num/frame_rate_den, an integer number, a float number or a
+       valid video frame rate abbreviation.
+
+       The following abbreviations are recognized:
+
+       ntsc
+	   30000/1001
+
+       pal 25/1
+
+       qntsc
+	   30000/1001
+
+       qpal
+	   25/1
+
+       sntsc
+	   30000/1001
+
+       spal
+	   25/1
+
+       film
+	   24/1
+
+       ntsc-film
+	   24000/1001
+
+   Ratio
+       A ratio can be expressed as an expression, or in the form
+       numerator:denominator.
+
+       Note that a ratio with infinite (1/0) or negative value is considered
+       valid, so you should check on the returned value if you want to exclude
+       those values.
+
+       The undefined value can be expressed using the "0:0" string.
+
+   Color
+       It can be the name of a color as defined below (case insensitive match)
+       or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string
+       representing the alpha component.
+
+       The alpha component may be a string composed by "0x" followed by an
+       hexadecimal number or a decimal number between 0.0 and 1.0, which
+       represents the opacity value (0x00 or 0.0 means completely transparent,
+       0xff or 1.0 completely opaque). If the alpha component is not specified
+       then 0xff is assumed.
+
+       The string random will result in a random color.
+
+       The following names of colors are recognized:
+
+       AliceBlue
+	   0xF0F8FF
+
+       AntiqueWhite
+	   0xFAEBD7
+
+       Aqua
+	   0x00FFFF
+
+       Aquamarine
+	   0x7FFFD4
+
+       Azure
+	   0xF0FFFF
+
+       Beige
+	   0xF5F5DC
+
+       Bisque
+	   0xFFE4C4
+
+       Black
+	   0x000000
+
+       BlanchedAlmond
+	   0xFFEBCD
+
+       Blue
+	   0x0000FF
+
+       BlueViolet
+	   0x8A2BE2
+
+       Brown
+	   0xA52A2A
+
+       BurlyWood
+	   0xDEB887
+
+       CadetBlue
+	   0x5F9EA0
+
+       Chartreuse
+	   0x7FFF00
+
+       Chocolate
+	   0xD2691E
+
+       Coral
+	   0xFF7F50
+
+       CornflowerBlue
+	   0x6495ED
+
+       Cornsilk
+	   0xFFF8DC
+
+       Crimson
+	   0xDC143C
+
+       Cyan
+	   0x00FFFF
+
+       DarkBlue
+	   0x00008B
+
+       DarkCyan
+	   0x008B8B
+
+       DarkGoldenRod
+	   0xB8860B
+
+       DarkGray
+	   0xA9A9A9
+
+       DarkGreen
+	   0x006400
+
+       DarkKhaki
+	   0xBDB76B
+
+       DarkMagenta
+	   0x8B008B
+
+       DarkOliveGreen
+	   0x556B2F
+
+       Darkorange
+	   0xFF8C00
+
+       DarkOrchid
+	   0x9932CC
+
+       DarkRed
+	   0x8B0000
+
+       DarkSalmon
+	   0xE9967A
+
+       DarkSeaGreen
+	   0x8FBC8F
+
+       DarkSlateBlue
+	   0x483D8B
+
+       DarkSlateGray
+	   0x2F4F4F
+
+       DarkTurquoise
+	   0x00CED1
+
+       DarkViolet
+	   0x9400D3
+
+       DeepPink
+	   0xFF1493
+
+       DeepSkyBlue
+	   0x00BFFF
+
+       DimGray
+	   0x696969
+
+       DodgerBlue
+	   0x1E90FF
+
+       FireBrick
+	   0xB22222
+
+       FloralWhite
+	   0xFFFAF0
+
+       ForestGreen
+	   0x228B22
+
+       Fuchsia
+	   0xFF00FF
+
+       Gainsboro
+	   0xDCDCDC
+
+       GhostWhite
+	   0xF8F8FF
+
+       Gold
+	   0xFFD700
+
+       GoldenRod
+	   0xDAA520
+
+       Gray
+	   0x808080
+
+       Green
+	   0x008000
+
+       GreenYellow
+	   0xADFF2F
+
+       HoneyDew
+	   0xF0FFF0
+
+       HotPink
+	   0xFF69B4
+
+       IndianRed
+	   0xCD5C5C
+
+       Indigo
+	   0x4B0082
+
+       Ivory
+	   0xFFFFF0
+
+       Khaki
+	   0xF0E68C
+
+       Lavender
+	   0xE6E6FA
+
+       LavenderBlush
+	   0xFFF0F5
+
+       LawnGreen
+	   0x7CFC00
+
+       LemonChiffon
+	   0xFFFACD
+
+       LightBlue
+	   0xADD8E6
+
+       LightCoral
+	   0xF08080
+
+       LightCyan
+	   0xE0FFFF
+
+       LightGoldenRodYellow
+	   0xFAFAD2
+
+       LightGreen
+	   0x90EE90
+
+       LightGrey
+	   0xD3D3D3
+
+       LightPink
+	   0xFFB6C1
+
+       LightSalmon
+	   0xFFA07A
+
+       LightSeaGreen
+	   0x20B2AA
+
+       LightSkyBlue
+	   0x87CEFA
+
+       LightSlateGray
+	   0x778899
+
+       LightSteelBlue
+	   0xB0C4DE
+
+       LightYellow
+	   0xFFFFE0
+
+       Lime
+	   0x00FF00
+
+       LimeGreen
+	   0x32CD32
+
+       Linen
+	   0xFAF0E6
+
+       Magenta
+	   0xFF00FF
+
+       Maroon
+	   0x800000
+
+       MediumAquaMarine
+	   0x66CDAA
+
+       MediumBlue
+	   0x0000CD
+
+       MediumOrchid
+	   0xBA55D3
+
+       MediumPurple
+	   0x9370D8
+
+       MediumSeaGreen
+	   0x3CB371
+
+       MediumSlateBlue
+	   0x7B68EE
+
+       MediumSpringGreen
+	   0x00FA9A
+
+       MediumTurquoise
+	   0x48D1CC
+
+       MediumVioletRed
+	   0xC71585
+
+       MidnightBlue
+	   0x191970
+
+       MintCream
+	   0xF5FFFA
+
+       MistyRose
+	   0xFFE4E1
+
+       Moccasin
+	   0xFFE4B5
+
+       NavajoWhite
+	   0xFFDEAD
+
+       Navy
+	   0x000080
+
+       OldLace
+	   0xFDF5E6
+
+       Olive
+	   0x808000
+
+       OliveDrab
+	   0x6B8E23
+
+       Orange
+	   0xFFA500
+
+       OrangeRed
+	   0xFF4500
+
+       Orchid
+	   0xDA70D6
+
+       PaleGoldenRod
+	   0xEEE8AA
+
+       PaleGreen
+	   0x98FB98
+
+       PaleTurquoise
+	   0xAFEEEE
+
+       PaleVioletRed
+	   0xD87093
+
+       PapayaWhip
+	   0xFFEFD5
+
+       PeachPuff
+	   0xFFDAB9
+
+       Peru
+	   0xCD853F
+
+       Pink
+	   0xFFC0CB
+
+       Plum
+	   0xDDA0DD
+
+       PowderBlue
+	   0xB0E0E6
+
+       Purple
+	   0x800080
+
+       Red 0xFF0000
+
+       RosyBrown
+	   0xBC8F8F
+
+       RoyalBlue
+	   0x4169E1
+
+       SaddleBrown
+	   0x8B4513
+
+       Salmon
+	   0xFA8072
+
+       SandyBrown
+	   0xF4A460
+
+       SeaGreen
+	   0x2E8B57
+
+       SeaShell
+	   0xFFF5EE
+
+       Sienna
+	   0xA0522D
+
+       Silver
+	   0xC0C0C0
+
+       SkyBlue
+	   0x87CEEB
+
+       SlateBlue
+	   0x6A5ACD
+
+       SlateGray
+	   0x708090
+
+       Snow
+	   0xFFFAFA
+
+       SpringGreen
+	   0x00FF7F
+
+       SteelBlue
+	   0x4682B4
+
+       Tan 0xD2B48C
+
+       Teal
+	   0x008080
+
+       Thistle
+	   0xD8BFD8
+
+       Tomato
+	   0xFF6347
+
+       Turquoise
+	   0x40E0D0
+
+       Violet
+	   0xEE82EE
+
+       Wheat
+	   0xF5DEB3
+
+       White
+	   0xFFFFFF
+
+       WhiteSmoke
+	   0xF5F5F5
+
+       Yellow
+	   0xFFFF00
+
+       YellowGreen
+	   0x9ACD32
+
+   Channel Layout
+       A channel layout specifies the spatial disposition of the channels in a
+       multi-channel audio stream. To specify a channel layout, FFmpeg makes
+       use of a special syntax.
+
+       Individual channels are identified by an id, as given by the table
+       below:
+
+       FL  front left
+
+       FR  front right
+
+       FC  front center
+
+       LFE low frequency
+
+       BL  back left
+
+       BR  back right
+
+       FLC front left-of-center
+
+       FRC front right-of-center
+
+       BC  back center
+
+       SL  side left
+
+       SR  side right
+
+       TC  top center
+
+       TFL top front left
+
+       TFC top front center
+
+       TFR top front right
+
+       TBL top back left
+
+       TBC top back center
+
+       TBR top back right
+
+       DL  downmix left
+
+       DR  downmix right
+
+       WL  wide left
+
+       WR  wide right
+
+       SDL surround direct left
+
+       SDR surround direct right
+
+       LFE2
+	   low frequency 2
+
+       Standard channel layout compositions can be specified by using the
+       following identifiers:
+
+       mono
+	   FC
+
+       stereo
+	   FL+FR
+
+       2.1 FL+FR+LFE
+
+       3.0 FL+FR+FC
+
+       3.0(back)
+	   FL+FR+BC
+
+       4.0 FL+FR+FC+BC
+
+       quad
+	   FL+FR+BL+BR
+
+       quad(side)
+	   FL+FR+SL+SR
+
+       3.1 FL+FR+FC+LFE
+
+       5.0 FL+FR+FC+BL+BR
+
+       5.0(side)
+	   FL+FR+FC+SL+SR
+
+       4.1 FL+FR+FC+LFE+BC
+
+       5.1 FL+FR+FC+LFE+BL+BR
+
+       5.1(side)
+	   FL+FR+FC+LFE+SL+SR
+
+       6.0 FL+FR+FC+BC+SL+SR
+
+       6.0(front)
+	   FL+FR+FLC+FRC+SL+SR
+
+       3.1.2
+	   FL+FR+FC+LFE+TFL+TFR
+
+       hexagonal
+	   FL+FR+FC+BL+BR+BC
+
+       6.1 FL+FR+FC+LFE+BC+SL+SR
+
+       6.1 FL+FR+FC+LFE+BL+BR+BC
+
+       6.1(front)
+	   FL+FR+LFE+FLC+FRC+SL+SR
+
+       7.0 FL+FR+FC+BL+BR+SL+SR
+
+       7.0(front)
+	   FL+FR+FC+FLC+FRC+SL+SR
+
+       7.1 FL+FR+FC+LFE+BL+BR+SL+SR
+
+       7.1(wide)
+	   FL+FR+FC+LFE+BL+BR+FLC+FRC
+
+       7.1(wide-side)
+	   FL+FR+FC+LFE+FLC+FRC+SL+SR
+
+       5.1.2
+	   FL+FR+FC+LFE+BL+BR+TFL+TFR
+
+       octagonal
+	   FL+FR+FC+BL+BR+BC+SL+SR
+
+       cube
+	   FL+FR+BL+BR+TFL+TFR+TBL+TBR
+
+       5.1.4
+	   FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR
+
+       7.1.2
+	   FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR
+
+       7.1.4
+	   FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR
+
+       hexadecagonal
+	   FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR
+
+       downmix
+	   DL+DR
+
+       22.2
+	   FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR
+
+       A custom channel layout can be specified as a sequence of terms,
+       separated by '+'.  Each term can be:
+
+       o   the name of a single channel (e.g. FL, FR, FC, LFE, etc.), each
+	   optionally containing a custom name after a '@', (e.g. FL@Left,
+	   FR@Right, FC@Center, LFE@Low_Frequency, etc.)
+
+       A standard channel layout can be specified by the following:
+
+       o   the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
+
+       o   the name of a standard channel layout (e.g. mono, stereo, 4.0,
+	   quad, 5.0, etc.)
+
+       o   a number of channels, in decimal, followed by 'c', yielding the
+	   default channel layout for that number of channels (see the
+	   function "av_channel_layout_default"). Note that not all channel
+	   counts have a default layout.
+
+       o   a number of channels, in decimal, followed by 'C', yielding an
+	   unknown channel layout with the specified number of channels. Note
+	   that not all channel layout specification strings support unknown
+	   channel layouts.
+
+       o   a channel layout mask, in hexadecimal starting with "0x" (see the
+	   "AV_CH_*" macros in libavutil/channel_layout.h.
+
+       Before libavutil version 53 the trailing character "c" to specify a
+       number of channels was optional, but now it is required, while a
+       channel layout mask can also be specified as a decimal number (if and
+       only if not followed by "c" or "C").
+
+       See also the function "av_channel_layout_from_string" defined in
+       libavutil/channel_layout.h.
+
+EXPRESSION EVALUATION
+       When evaluating an arithmetic expression, FFmpeg uses an internal
+       formula evaluator, implemented through the libavutil/eval.h interface.
+
+       An expression may contain unary, binary operators, constants, and
+       functions.
+
+       Two expressions expr1 and expr2 can be combined to form another
+       expression "expr1;expr2".  expr1 and expr2 are evaluated in turn, and
+       the new expression evaluates to the value of expr2.
+
+       The following binary operators are available: "+", "-", "*", "/", "^".
+
+       The following unary operators are available: "+", "-".
+
+       The following functions are available:
+
+       abs(x)
+	   Compute absolute value of x.
+
+       acos(x)
+	   Compute arccosine of x.
+
+       asin(x)
+	   Compute arcsine of x.
+
+       atan(x)
+	   Compute arctangent of x.
+
+       atan2(x, y)
+	   Compute principal value of the arc tangent of y/x.
+
+       between(x, min, max)
+	   Return 1 if x is greater than or equal to min and lesser than or
+	   equal to max, 0 otherwise.
+
+       bitand(x, y)
+       bitor(x, y)
+	   Compute bitwise and/or operation on x and y.
+
+	   The results of the evaluation of x and y are converted to integers
+	   before executing the bitwise operation.
+
+	   Note that both the conversion to integer and the conversion back to
+	   floating point can lose precision. Beware of unexpected results for
+	   large numbers (usually 2^53 and larger).
+
+       ceil(expr)
+	   Round the value of expression expr upwards to the nearest integer.
+	   For example, "ceil(1.5)" is "2.0".
+
+       clip(x, min, max)
+	   Return the value of x clipped between min and max.
+
+       cos(x)
+	   Compute cosine of x.
+
+       cosh(x)
+	   Compute hyperbolic cosine of x.
+
+       eq(x, y)
+	   Return 1 if x and y are equivalent, 0 otherwise.
+
+       exp(x)
+	   Compute exponential of x (with base "e", the Euler's number).
+
+       floor(expr)
+	   Round the value of expression expr downwards to the nearest
+	   integer. For example, "floor(-1.5)" is "-2.0".
+
+       gauss(x)
+	   Compute Gauss function of x, corresponding to "exp(-x*x/2) /
+	   sqrt(2*PI)".
+
+       gcd(x, y)
+	   Return the greatest common divisor of x and y. If both x and y are
+	   0 or either or both are less than zero then behavior is undefined.
+
+       gt(x, y)
+	   Return 1 if x is greater than y, 0 otherwise.
+
+       gte(x, y)
+	   Return 1 if x is greater than or equal to y, 0 otherwise.
+
+       hypot(x, y)
+	   This function is similar to the C function with the same name; it
+	   returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right
+	   triangle with sides of length x and y, or the distance of the point
+	   (x, y) from the origin.
+
+       if(x, y)
+	   Evaluate x, and if the result is non-zero return the result of the
+	   evaluation of y, return 0 otherwise.
+
+       if(x, y, z)
+	   Evaluate x, and if the result is non-zero return the evaluation
+	   result of y, otherwise the evaluation result of z.
+
+       ifnot(x, y)
+	   Evaluate x, and if the result is zero return the result of the
+	   evaluation of y, return 0 otherwise.
+
+       ifnot(x, y, z)
+	   Evaluate x, and if the result is zero return the evaluation result
+	   of y, otherwise the evaluation result of z.
+
+       isinf(x)
+	   Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
+
+       isnan(x)
+	   Return 1.0 if x is NAN, 0.0 otherwise.
+
+       ld(var)
+	   Load the value of the internal variable with number var, which was
+	   previously stored with st(var, expr).  The function returns the
+	   loaded value.
+
+       lerp(x, y, z)
+	   Return linear interpolation between x and y by amount of z.
+
+       log(x)
+	   Compute natural logarithm of x.
+
+       lt(x, y)
+	   Return 1 if x is lesser than y, 0 otherwise.
+
+       lte(x, y)
+	   Return 1 if x is lesser than or equal to y, 0 otherwise.
+
+       max(x, y)
+	   Return the maximum between x and y.
+
+       min(x, y)
+	   Return the minimum between x and y.
+
+       mod(x, y)
+	   Compute the remainder of division of x by y.
+
+       not(expr)
+	   Return 1.0 if expr is zero, 0.0 otherwise.
+
+       pow(x, y)
+	   Compute the power of x elevated y, it is equivalent to "(x)^(y)".
+
+       print(t)
+       print(t, l)
+	   Print the value of expression t with loglevel l. If l is not
+	   specified then a default log level is used.	Returns the value of
+	   the expression printed.
+
+	   Prints t with loglevel l
+
+       random(x)
+	   Return a pseudo random value between 0.0 and 1.0. x is the index of
+	   the internal variable which will be used to save the seed/state.
+
+       root(expr, max)
+	   Find an input value for which the function represented by expr with
+	   argument ld(0) is 0 in the interval 0..max.
+
+	   The expression in expr must denote a continuous function or the
+	   result is undefined.
+
+	   ld(0) is used to represent the function input value, which means
+	   that the given expression will be evaluated multiple times with
+	   various input values that the expression can access through ld(0).
+	   When the expression evaluates to 0 then the corresponding input
+	   value will be returned.
+
+       round(expr)
+	   Round the value of expression expr to the nearest integer. For
+	   example, "round(1.5)" is "2.0".
+
+       sgn(x)
+	   Compute sign of x.
+
+       sin(x)
+	   Compute sine of x.
+
+       sinh(x)
+	   Compute hyperbolic sine of x.
+
+       sqrt(expr)
+	   Compute the square root of expr. This is equivalent to "(expr)^.5".
+
+       squish(x)
+	   Compute expression "1/(1 + exp(4*x))".
+
+       st(var, expr)
+	   Store the value of the expression expr in an internal variable. var
+	   specifies the number of the variable where to store the value, and
+	   it is a value ranging from 0 to 9. The function returns the value
+	   stored in the internal variable.  Note, Variables are currently not
+	   shared between expressions.
+
+       tan(x)
+	   Compute tangent of x.
+
+       tanh(x)
+	   Compute hyperbolic tangent of x.
+
+       taylor(expr, x)
+       taylor(expr, x, id)
+	   Evaluate a Taylor series at x, given an expression representing the
+	   "ld(id)"-th derivative of a function at 0.
+
+	   When the series does not converge the result is undefined.
+
+	   ld(id) is used to represent the derivative order in expr, which
+	   means that the given expression will be evaluated multiple times
+	   with various input values that the expression can access through
+	   "ld(id)". If id is not specified then 0 is assumed.
+
+	   Note, when you have the derivatives at y instead of 0,
+	   "taylor(expr, x-y)" can be used.
+
+       time(0)
+	   Return the current (wallclock) time in seconds.
+
+       trunc(expr)
+	   Round the value of expression expr towards zero to the nearest
+	   integer. For example, "trunc(-1.5)" is "-1.0".
+
+       while(cond, expr)
+	   Evaluate expression expr while the expression cond is non-zero, and
+	   returns the value of the last expr evaluation, or NAN if cond was
+	   always false.
+
+       The following constants are available:
+
+       PI  area of the unit disc, approximately 3.14
+
+       E   exp(1) (Euler's number), approximately 2.718
+
+       PHI golden ratio (1+sqrt(5))/2, approximately 1.618
+
+       Assuming that an expression is considered "true" if it has a non-zero
+       value, note that:
+
+       "*" works like AND
+
+       "+" works like OR
+
+       For example the construct:
+
+	       if (A AND B) then C
+
+       is equivalent to:
+
+	       if(A*B, C)
+
+       In your C code, you can extend the list of unary and binary functions,
+       and define recognized constants, so that they are available for your
+       expressions.
+
+       The evaluator also recognizes the International System unit prefixes.
+       If 'i' is appended after the prefix, binary prefixes are used, which
+       are based on powers of 1024 instead of powers of 1000.  The 'B' postfix
+       multiplies the value by 8, and can be appended after a unit prefix or
+       used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as
+       number postfix.
+
+       The list of available International System prefixes follows, with
+       indication of the corresponding powers of 10 and of 2.
+
+       y   10^-24 / 2^-80
+
+       z   10^-21 / 2^-70
+
+       a   10^-18 / 2^-60
+
+       f   10^-15 / 2^-50
+
+       p   10^-12 / 2^-40
+
+       n   10^-9 / 2^-30
+
+       u   10^-6 / 2^-20
+
+       m   10^-3 / 2^-10
+
+       c   10^-2
+
+       d   10^-1
+
+       h   10^2
+
+       k   10^3 / 2^10
+
+       K   10^3 / 2^10
+
+       M   10^6 / 2^20
+
+       G   10^9 / 2^30
+
+       T   10^12 / 2^40
+
+       P   10^15 / 2^50
+
+       E   10^18 / 2^60
+
+       Z   10^21 / 2^70
+
+       Y   10^24 / 2^80
+
+SEE ALSO
+       ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+							       FFMPEG-UTILS(1)

ffmpeg-6.1-amd64-static/manpages/ffprobe.txt

@@ -0,0 +1,976 @@
+FFPROBE(1)							    FFPROBE(1)
+
+NAME
+       ffprobe - ffprobe media prober
+
+SYNOPSIS
+       ffprobe [options] input_url
+
+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 url is specified in input, ffprobe will try to open and probe the
+       url content. If the url cannot be opened or recognized as a multimedia
+       file, a positive exit code is returned.
+
+       If no output is specified as output with o ffprobe will write to
+       stdout.
+
+       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 output_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 sections.
+
+       Metadata tags stored in the container or in the streams are recognized
+       and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
+       section.
+
+OPTIONS
+       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 multiples, 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.
+
+   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. "-codec:a:1 ac3" contains the
+       "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 "-b:a 128k"
+       matches all audio streams.
+
+       An empty stream specifier matches all streams. For example, "-codec
+       copy" or "-codec: copy" would copy all the streams without reencoding.
+
+       Possible forms of stream specifiers are:
+
+       stream_index
+	   Matches the stream with this index. E.g. "-threads:1 4" would set
+	   the thread count for the second stream to 4. If stream_index is
+	   used as an additional stream specifier (see below), then it selects
+	   stream number stream_index from the matching streams. Stream
+	   numbering is based on the order of the streams as detected by
+	   libavformat except when a program ID is also specified. In this
+	   case it is based on the ordering of the streams in the program.
+
+       stream_type[:additional_stream_specifier]
+	   stream_type is one of following: 'v' or 'V' for video, 'a' for
+	   audio, 's' for subtitle, 'd' for data, and 't' for attachments. 'v'
+	   matches all video streams, 'V' only matches video streams which are
+	   not attached pictures, video thumbnails or cover arts. If
+	   additional_stream_specifier is used, then it matches streams which
+	   both have this type and match the additional_stream_specifier.
+	   Otherwise, it matches all streams of the specified type.
+
+       p:program_id[:additional_stream_specifier]
+	   Matches streams which are in the program with the id program_id. If
+	   additional_stream_specifier is used, then it matches streams which
+	   both are part of the program and match the
+	   additional_stream_specifier.
+
+       #stream_id or i:stream_id
+	   Match the stream by stream id (e.g. PID in MPEG-TS container).
+
+       m:key[:value]
+	   Matches streams with the metadata tag key having the specified
+	   value. If value is not given, matches streams that contain the
+	   given tag with any value.
+
+       u   Matches streams with usable configuration, the codec must be
+	   defined and the essential information such as video dimension or
+	   audio sample rate must be present.
+
+	   Note that in ffmpeg, matching by metadata will only work properly
+	   for input files.
+
+   Generic options
+       These options are shared amongst the ff* tools.
+
+       -L  Show license.
+
+       -h, -?, -help, --help [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 arg are:
+
+	   long
+	       Print advanced tool options in addition to the basic tool
+	       options.
+
+	   full
+	       Print complete list of options, including shared and private
+	       options for encoders, decoders, demuxers, muxers, filters, etc.
+
+	   decoder=decoder_name
+	       Print detailed information about the decoder named
+	       decoder_name. Use the -decoders option to get a list of all
+	       decoders.
+
+	   encoder=encoder_name
+	       Print detailed information about the encoder named
+	       encoder_name. Use the -encoders option to get a list of all
+	       encoders.
+
+	   demuxer=demuxer_name
+	       Print detailed information about the demuxer named
+	       demuxer_name. Use the -formats option to get a list of all
+	       demuxers and muxers.
+
+	   muxer=muxer_name
+	       Print detailed information about the muxer named muxer_name.
+	       Use the -formats option to get a list of all muxers and
+	       demuxers.
+
+	   filter=filter_name
+	       Print detailed information about the filter named filter_name.
+	       Use the -filters option to get a list of all filters.
+
+	   bsf=bitstream_filter_name
+	       Print detailed information about the bitstream filter named
+	       bitstream_filter_name.  Use the -bsfs option to get a list of
+	       all bitstream filters.
+
+	   protocol=protocol_name
+	       Print detailed information about the protocol named
+	       protocol_name.  Use the -protocols option to get a list of all
+	       protocols.
+
+       -version
+	   Show version.
+
+       -buildconf
+	   Show the build configuration, one option per line.
+
+       -formats
+	   Show available formats (including devices).
+
+       -demuxers
+	   Show available demuxers.
+
+       -muxers
+	   Show available muxers.
+
+       -devices
+	   Show available devices.
+
+       -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.
+
+       -decoders
+	   Show available decoders.
+
+       -encoders
+	   Show all available encoders.
+
+       -bsfs
+	   Show available bitstream filters.
+
+       -protocols
+	   Show available protocols.
+
+       -filters
+	   Show available libavfilter filters.
+
+       -pix_fmts
+	   Show available pixel formats.
+
+       -sample_fmts
+	   Show available sample formats.
+
+       -layouts
+	   Show channel names and standard channel layouts.
+
+       -dispositions
+	   Show stream dispositions.
+
+       -colors
+	   Show recognized color names.
+
+       -sources device[,opt1=val1[,opt2=val2]...]
+	   Show autodetected sources of the input device.  Some devices may
+	   provide system-dependent source names that cannot be autodetected.
+	   The returned list cannot be assumed to be always complete.
+
+		   ffmpeg -sources pulse,server=192.168.0.4
+
+       -sinks device[,opt1=val1[,opt2=val2]...]
+	   Show autodetected sinks of the output device.  Some devices may
+	   provide system-dependent sink names that cannot be autodetected.
+	   The returned list cannot be assumed to be always complete.
+
+		   ffmpeg -sinks pulse,server=192.168.0.4
+
+       -loglevel [flags+]loglevel | -v [flags+]loglevel
+	   Set logging level and flags used by the library.
+
+	   The optional flags prefix can consist of the following values:
+
+	   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.
+
+	   level
+	       Indicates that log output should add a "[level]" prefix to each
+	       message line. This can be used as an alternative to log
+	       coloring, e.g. when dumping the log to file.
+
+	   Flags can also be used alone by adding a '+'/'-' prefix to
+	   set/reset a single flag without affecting other flags or changing
+	   loglevel. When setting both flags and loglevel, a '+' separator is
+	   expected between the last flags value and before loglevel.
+
+	   loglevel is a string or a number containing one of the following
+	   values:
+
+	   quiet, -8
+	       Show nothing at all; be silent.
+
+	   panic, 0
+	       Only show fatal errors which could lead the process to crash,
+	       such as an assertion failure. This is not currently used for
+	       anything.
+
+	   fatal, 8
+	       Only show fatal errors. These are errors after which the
+	       process absolutely cannot continue.
+
+	   error, 16
+	       Show all errors, including ones which can be recovered from.
+
+	   warning, 24
+	       Show all warnings and errors. Any message related to possibly
+	       incorrect or unexpected events will be shown.
+
+	   info, 32
+	       Show informative messages during processing. This is in
+	       addition to warnings and errors. This is the default value.
+
+	   verbose, 40
+	       Same as "info", except more verbose.
+
+	   debug, 48
+	       Show everything, including debugging information.
+
+	   trace, 56
+
+	   For example to enable repeated log output, add the "level" prefix,
+	   and set loglevel to "verbose":
+
+		   ffmpeg -loglevel repeat+level+verbose -i input output
+
+	   Another example that enables repeated log output without affecting
+	   current state of "level" prefix flag or loglevel:
+
+		   ffmpeg [...] -loglevel +repeat
+
+	   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
+	   AV_LOG_FORCE_NOCOLOR, or can be forced setting the environment
+	   variable AV_LOG_FORCE_COLOR.
+
+       -report
+	   Dump full command line and log output to a file named
+	   "program-YYYYMMDD-HHMMSS.log" in the current directory.  This file
+	   can be useful for bug reports.  It also implies "-loglevel debug".
+
+	   Setting the environment variable FFREPORT to any value has the same
+	   effect. If the value is a ':'-separated key=value sequence, these
+	   options will affect the report; option 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 options are recognized:
+
+	   file
+	       set the file name to use for the report; %p is expanded to the
+	       name of the program, %t is expanded to a timestamp, "%%" is
+	       expanded to a plain "%"
+
+	   level
+	       set the log verbosity level using a numerical value (see
+	       "-loglevel").
+
+	   For example, to output a report to a file named ffreport.log using
+	   a log level of 32 (alias for log level "info"):
+
+		   FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
+
+	   Errors in parsing the environment variable are not fatal, and will
+	   not appear in the report.
+
+       -hide_banner
+	   Suppress printing banner.
+
+	   All FFmpeg tools will normally show a copyright notice, build
+	   options and library versions. This option can be used to suppress
+	   printing this information.
+
+       -cpuflags flags (global)
+	   Allows setting and clearing cpu flags. This option is intended for
+	   testing. Do not use it unless you know what you're doing.
+
+		   ffmpeg -cpuflags -sse+mmx ...
+		   ffmpeg -cpuflags mmx ...
+		   ffmpeg -cpuflags 0 ...
+
+	   Possible flags for this option are:
+
+	   x86
+	       mmx
+	       mmxext
+	       sse
+	       sse2
+	       sse2slow
+	       sse3
+	       sse3slow
+	       ssse3
+	       atom
+	       sse4.1
+	       sse4.2
+	       avx
+	       avx2
+	       xop
+	       fma3
+	       fma4
+	       3dnow
+	       3dnowext
+	       bmi1
+	       bmi2
+	       cmov
+	   ARM
+	       armv5te
+	       armv6
+	       armv6t2
+	       vfp
+	       vfpv3
+	       neon
+	       setend
+	   AArch64
+	       armv8
+	       vfp
+	       neon
+	   PowerPC
+	       altivec
+	   Specific Processors
+	       pentium2
+	       pentium3
+	       pentium4
+	       k6
+	       k62
+	       athlon
+	       athlonxp
+	       k8
+       -cpucount count (global)
+	   Override detection of CPU count. This option is intended for
+	   testing. Do not use it unless you know what you're doing.
+
+		   ffmpeg -cpucount 2
+
+       -max_alloc bytes
+	   Set the maximum size limit for allocating a block on the heap by
+	   ffmpeg's family of malloc functions. Exercise extreme caution when
+	   using this option. Don't use if you do not understand the full
+	   consequence of doing so.  Default is INT_MAX.
+
+   AVOptions
+       These options are provided directly by the libavformat, libavdevice and
+       libavcodec libraries. To see the list of available AVOptions, use the
+       -help option. They are separated into two categories:
+
+       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.
+
+       private
+	   These options are specific to the given container, device or codec.
+	   Private options are listed under their corresponding
+	   containers/devices/codecs.
+
+       For example to write an ID3v2.3 header instead of a default ID3v2.4 to
+       an MP3 file, use the id3v2_version private option of the MP3 muxer:
+
+	       ffmpeg -i input.flac -id3v2_version 3 out.mp3
+
+       All codec AVOptions are per-stream, and thus a stream specifier should
+       be attached to them:
+
+	       ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4
+
+       In the above example, a multichannel audio stream is mapped twice for
+       output.	The first instance is encoded with codec ac3 and bitrate 640k.
+       The second instance is downmixed to 2 channels and encoded with codec
+       aac. A bitrate of 128k is specified for it using absolute index of the
+       output stream.
+
+       Note: the -nooption syntax cannot be used for boolean AVOptions, use
+       -option 0/-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.
+
+   Main options
+       -f format
+	   Force format to use.
+
+       -unit
+	   Show the unit of the displayed values.
+
+       -prefix
+	   Use SI prefixes for the displayed values.  Unless the
+	   "-byte_binary_prefix" option is used all the prefixes are decimal.
+
+       -byte_binary_prefix
+	   Force the use of binary prefixes for byte values.
+
+       -sexagesimal
+	   Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
+
+       -pretty
+	   Prettify the format of the displayed values, it corresponds to the
+	   options "-unit -prefix -byte_binary_prefix -sexagesimal".
+
+       -output_format, -of, -print_format writer_name[=writer_options]
+	   Set the output printing format.
+
+	   writer_name specifies the name of the writer, and writer_options
+	   specifies the options to be passed to the writer.
+
+	   For example for printing the output in JSON format, specify:
+
+		   -output_format json
+
+	   For more details on the available output printing formats, see the
+	   Writers section below.
+
+       -sections
+	   Print sections structure and section information, and exit. The
+	   output is not meant to be parsed by a machine.
+
+       -select_streams stream_specifier
+	   Select only the streams specified by stream_specifier. This option
+	   affects only the options related to streams (e.g. "show_streams",
+	   "show_packets", etc.).
+
+	   For example to show only audio streams, you can use the command:
+
+		   ffprobe -show_streams -select_streams a INPUT
+
+	   To show only video packets belonging to the video stream with index
+	   1:
+
+		   ffprobe -show_packets -select_streams v:1 INPUT
+
+       -show_data
+	   Show payload data, as a hexadecimal and ASCII dump. Coupled with
+	   -show_packets, it will dump the packets' data. Coupled with
+	   -show_streams, it will dump the codec extradata.
+
+	   The dump is printed as the "data" field. It may contain newlines.
+
+       -show_data_hash algorithm
+	   Show a hash of payload data, for packets with -show_packets and for
+	   codec extradata with -show_streams.
+
+       -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".
+
+       -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".
+
+       -show_format_entry name
+	   Like -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 "show_entries" instead.
+
+       -show_entries section_entries
+	   Set list of entries to show.
+
+	   Entries are specified according to the following syntax.
+	   section_entries contains a list of section entries separated by
+	   ":". 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 ",".
+
+	   If section name is specified but is followed by no "=", 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 "=" 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:
+
+		   <LOCAL_SECTION_ENTRIES> ::= <SECTION_ENTRY_NAME>[,<LOCAL_SECTION_ENTRIES>]
+		   <SECTION_ENTRY>	   ::= <SECTION_NAME>[=[<LOCAL_SECTION_ENTRIES>]]
+		   <SECTION_ENTRIES>	   ::= <SECTION_ENTRY>[:<SECTION_ENTRIES>]
+
+	   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:
+
+		   packet=pts_time,duration_time,stream_index : stream=index,codec_type
+
+	   To show all the entries in the section "format", but only the codec
+	   type in the section "stream", specify the argument:
+
+		   format : stream=codec_type
+
+	   To show all the tags in the stream and format sections:
+
+		   stream_tags : format_tags
+
+	   To show only the "title" tag (if available) in the stream sections:
+
+		   stream_tags=title
+
+       -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".
+
+       -show_frames
+	   Show information about each frame and subtitle contained in the
+	   input multimedia stream.
+
+	   The information for each single frame is printed within a dedicated
+	   section with name "FRAME" or "SUBTITLE".
+
+       -show_log loglevel
+	   Show logging information from the decoder about each frame
+	   according to the value set in loglevel, (see "-loglevel"). This
+	   option requires "-show_frames".
+
+	   The information for each log message is printed within a dedicated
+	   section with name "LOG".
+
+       -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".
+
+       -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".
+
+       -show_chapters
+	   Show information about chapters stored in the format.
+
+	   Each chapter is printed within a dedicated section with name
+	   "CHAPTER".
+
+       -count_frames
+	   Count the number of frames per stream and report it in the
+	   corresponding stream section.
+
+       -count_packets
+	   Count the number of packets per stream and report it in the
+	   corresponding stream section.
+
+       -read_intervals read_intervals
+	   Read only the specified intervals. read_intervals must be a
+	   sequence of interval specifications separated by ",".  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 absolute 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:
+
+		   <INTERVAL>  ::= [<START>|+<START_OFFSET>][%[<END>|+<END_OFFSET>]]
+		   <INTERVALS> ::= <INTERVAL>[,<INTERVALS>]
+
+	   A few examples follow.
+
+	   o   Seek to time 10, read packets until 20 seconds after the found
+	       seek point, then seek to position "01:30" (1 minute and thirty
+	       seconds) and read packets until position "01:45".
+
+		       10%+20,01:30%01:45
+
+	   o   Read only 42 packets after seeking to position "01:23":
+
+		       01:23%+#42
+
+	   o   Read only the first 20 seconds from the start:
+
+		       %+20
+
+	   o   Read from the start until position "02:30":
+
+		       %02:30
+
+       -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.
+
+       -show_program_version
+	   Show information related to program version.
+
+	   Version information is printed within a section with name
+	   "PROGRAM_VERSION".
+
+       -show_library_versions
+	   Show information related to library versions.
+
+	   Version information for each library is printed within a section
+	   with name "LIBRARY_VERSION".
+
+       -show_versions
+	   Show information related to program and library versions. This is
+	   the equivalent of setting both -show_program_version and
+	   -show_library_versions options.
+
+       -show_pixel_formats
+	   Show information about all pixel formats supported by FFmpeg.
+
+	   Pixel format information for each format is printed within a
+	   section with name "PIXEL_FORMAT".
+
+       -show_optional_fields value
+	   Some writers viz. JSON and XML, omit the printing of fields with
+	   invalid or non-applicable values, while other writers always print
+	   them. This option enables one to control this behaviour.  Valid
+	   values are "always"/1, "never"/0 and "auto"/"-1".  Default is auto.
+
+       -bitexact
+	   Force bitexact output, useful to produce output which is not
+	   dependent on the specific build.
+
+       -i input_url
+	   Read input_url.
+
+       -o output_url
+	   Write output to output_url. If not specified, the output is sent to
+	   stdout.
+
+WRITERS
+       A writer defines the output format adopted by 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 key=value pairs,
+       separated by ":".
+
+       All writers support the following options:
+
+       string_validation, sv
+	   Set string validation mode.
+
+	   The following values are accepted.
+
+	   fail
+	       The writer will fail immediately in case an invalid string
+	       (UTF-8) sequence or code point is found in the input. This is
+	       especially useful to validate input metadata.
+
+	   ignore
+	       Any validation error will be ignored. This will result in
+	       possibly broken output, especially with the json or xml writer.
+
+	   replace
+	       The writer will substitute invalid UTF-8 sequences or code
+	       points with the string specified with the
+	       string_validation_replacement.
+
+	   Default value is replace.
+
+       string_validation_replacement, svr
+	   Set replacement string to use in case string_validation is set to
+	   replace.
+
+	   In case the option is not specified, the writer will assume the
+	   empty string, that is it will remove the invalid sequences from the
+	   input strings.
+
+       A description of the currently available writers follows.
+
+   default
+       Default format.
+
+       Print each section in the form:
+
+	       [SECTION]
+	       key1=val1
+	       ...
+	       keyN=valN
+	       [/SECTION]
+
+       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.
+
+       nokey, nk
+	   If set to 1 specify not to print the key of each field. Default
+	   value is 0.
+
+       noprint_wrappers, nw
+	   If set to 1 specify not to print the section header and footer.
+	   Default value is 0.
+
+   compact, csv
+       Compact and CSV format.
+
+       The "csv" writer is equivalent to "compact", but supports different
+       defaults.
+
+       Each section is printed on a single line.  If no option is specified,
+       the output has the form:
+
+	       section|key1=val1| ... |keyN=valN
+
+       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.
+
+       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 "csv" writer).
+
+       nokey, nk
+	   If set to 1 specify not to print the key of each field. Its default
+	   value is 0 (1 for the "csv" writer).
+
+       escape, e
+	   Set the escape mode to use, default to "c" ("csv" for the "csv"
+	   writer).
+
+	   It can assume one of the following values:
+
+	   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 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 SEP is converted to \SEP.
+
+	   csv Perform CSV-like escaping, as described in RFC4180.  Strings
+	       containing a newline (\n), a carriage return (\r), a double
+	       quote ("), or SEP are enclosed in double-quotes.
+
+	   none
+	       Perform no escaping.
+
+       print_section, p
+	   Print the section name at the beginning of each line if the value
+	   is 1, disable it with value set to 0. Default value is 1.
+
+   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 sep_char
+       option).
+
+       The description of the accepted options follows.
+
+       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 ..
+
+       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.
+
+   ini
+       INI format output.
+
+       Print output in an INI based format.
+
+       The following conventions are adopted:
+
+       o   all key and values are UTF-8
+
+       o   . is the subgroup separator
+
+       o   newline, \t, \f, \b and the following characters are escaped
+
+       o   \ is the escape character
+
+       o   # is the comment indicator
+
+       o   = is the key/value separator
+
+       o   : is not used but usually parsed as key/value separator
+
+       This writer accepts options as a list of key=value pairs, separated by
+       :.
+
+       The description of the accepted options follows.
+
+       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.
+
+   json
+       JSON based format.
+
+       Each section is printed using JSON notation.
+
+       The description of the accepted options follows.
+
+       compact, c
+	   If set to 1 enable compact output, that is each section will be
+	   printed on a single line. Default value is 0.
+
+       For more information about JSON, see <http://www.json.org/>.
+
+   xml
+       XML based format.
+
+       The XML output is described in the XML schema description file
+       ffprobe.xsd installed in the FFmpeg datadir.
+
+       An updated version of the schema can be retrieved at the 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 ffprobe.xsd schema
+       only when no special global output options (unit, prefix,
+       byte_binary_prefix, sexagesimal etc.) are specified.
+
+       The description of the accepted options follows.
+
+       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.
+
+       xsd_strict, x
+	   If set to 1 perform more checks for ensuring that the output is XSD
+	   compliant. Default value is 0.  This option automatically sets
+	   fully_qualified to 1.
+
+       For more information about the XML format, see
+       <https://www.w3.org/XML/>.
+
+TIMECODE
+       ffprobe supports Timecode extraction:
+
+       o   MPEG1/2 timecode is extracted from the GOP, and is available in the
+	   video stream details (-show_streams, see timecode).
+
+       o   MOV timecode is extracted from tmcd track, so is available in the
+	   tmcd stream metadata (-show_streams, see TAG:timecode).
+
+       o   DV, GXF and AVI timecodes are available in format metadata
+	   (-show_format, see TAG:timecode).
+
+SEE ALSO
+       ffprobe-all(1), ffmpeg(1), ffplay(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)
+
+AUTHORS
+       The FFmpeg developers.
+
+       For details about the authorship, see the Git history of the project
+       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
+       the FFmpeg source directory, or browsing the online repository at
+       <https://git.ffmpeg.org/ffmpeg>.
+
+       Maintainers for the specific components are listed in the file
+       MAINTAINERS in the source code tree.
+
+								    FFPROBE(1)

ffmpeg-6.1-amd64-static/model/000-PLEASE-README.TXT

@@ -0,0 +1,4 @@
+The vmaf filter expects this directory of models to be preset at /usr/local/share/model
+
+If you wish to use this filter, run "sudo cp -r model /usr/local/share/" from the directory
+containing the static binaries. If not, you can ignore this and do nothing.

ffmpeg-6.1-amd64-static/model/other_models/model_V8a.model

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ffmpeg-6.1-amd64-static/model/other_models/nflx_v1.json

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+{
+    "param_dict": {
+        "C": 1.0,
+        "norm_type": "clip_0to1",
+        "score_clip": [
+            0.0,
+            100.0
+        ],
+        "cache_size": 200,
+        "nu": 0.5,
+        "gamma": 0.85
+    },
+    "model_dict": {
+        "norm_type": "linear_rescale",
+        "score_clip": [
+            0.0,
+            100.0
+        ],
+        "feature_names": [
+            "VMAF_feature_adm_score",
+            "VMAF_feature_ansnr_score",
+            "VMAF_feature_motion_score",
+            "VMAF_feature_vif_score"
+        ],
+        "intercepts": [
+            -0.1909090909090909,
+            -1.635828565827225,
+            -0.5027725296167747,
+            -0.022214587359292954,
+            -0.12191917348723096
+        ],
+        "model_type": "LIBSVMNUSVR",
+        "slopes": [
+            0.010909090909090908,
+            2.635828565827225,
+            0.030306790717580585,
+            0.06846153126171134,
+            1.121919173487231
+        ],
+        "model": "svm_type nu_svr\nkernel_type rbf\ngamma 0.85\nnr_class 2\ntotal_sv 87\nrho -0.678599\nSV\n-1 1:0.500758 2:0.44434935 3:0.80360073 4:0.25504335 \n-1 1:0.66395488 2:0.49545656 3:0.80360073 4:0.30315084 \n-1 1:0.68710445 2:0.51616008 3:0.80360073 4:0.32840241 \n1 1:0.80552743 2:0.57886981 3:0.80360073 4:0.3952589 \n1 1:0.93943344 2:0.67382416 3:0.80360073 4:0.48875504 \n-0.6005168509118413 1:0.44111817 2:0.13943156 3:0.060660457 4:0.23164915 \n0.3001652364483464 1:0.79372033 2:0.31017378 3:0.060660457 4:0.4749309 \n1 1:0.91710352 2:0.45437437 3:0.060660457 4:0.66549345 \n0.2347757543790557 1:0.99910085 2:0.62307047 3:0.060660457 4:0.94344783 \n0.9879940111050362 1:0.70228825 2:0.41065868 3:0.25800309 4:0.16020672 \n1 1:0.71225044 2:0.41933192 3:0.25800309 4:0.16684779 \n1 1:0.79683949 2:0.43826809 3:0.25800309 4:0.17816881 \n1 1:0.81652281 2:0.45300881 3:0.25800309 4:0.19016999 \n-1 1:0.93569958 2:0.48157676 3:0.25800309 4:0.21208644 \n-0.05952502774634448 1:0.93913288 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+    }
+}

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+{
+    "param_dict": {
+        "C": 4.0,
+        "norm_type": "none",
+        "score_clip": [
+            0.0,
+            100.0
+        ],
+        "custom_clip_0to1_map": {
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+                0.5
+            ]
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+    },
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+            0.0,
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+        ],
+        "feature_names": [
+            "VMAF_feature_adm2_score",
+            "VMAF_feature_motion_score",
+            "VMAF_feature_vif_scale0_score",
+            "VMAF_feature_vif_scale1_score",
+            "VMAF_feature_vif_scale2_score",
+            "VMAF_feature_vif_scale3_score"
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