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2021-02-09 04:25:58 +01:00
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/*.1
/*.3
/*.html
/*.pod
/config.texi
/avoptions_codec.texi
/avoptions_format.texi
/fate.txt
/print_options

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LIBRARIES-$(CONFIG_AVUTIL) += libavutil
LIBRARIES-$(CONFIG_SWSCALE) += libswscale
LIBRARIES-$(CONFIG_SWRESAMPLE) += libswresample
LIBRARIES-$(CONFIG_AVCODEC) += libavcodec
LIBRARIES-$(CONFIG_AVFORMAT) += libavformat
LIBRARIES-$(CONFIG_AVDEVICE) += libavdevice
LIBRARIES-$(CONFIG_AVFILTER) += libavfilter
COMPONENTS-$(CONFIG_AVUTIL) += ffmpeg-utils
COMPONENTS-$(CONFIG_SWSCALE) += ffmpeg-scaler
COMPONENTS-$(CONFIG_SWRESAMPLE) += ffmpeg-resampler
COMPONENTS-$(CONFIG_AVCODEC) += ffmpeg-codecs ffmpeg-bitstream-filters
COMPONENTS-$(CONFIG_AVFORMAT) += ffmpeg-formats ffmpeg-protocols
COMPONENTS-$(CONFIG_AVDEVICE) += ffmpeg-devices
COMPONENTS-$(CONFIG_AVFILTER) += ffmpeg-filters
MANPAGES1 = $(AVPROGS-yes:%=doc/%.1) $(AVPROGS-yes:%=doc/%-all.1) $(COMPONENTS-yes:%=doc/%.1)
MANPAGES3 = $(LIBRARIES-yes:%=doc/%.3)
MANPAGES = $(MANPAGES1) $(MANPAGES3)
PODPAGES = $(AVPROGS-yes:%=doc/%.pod) $(AVPROGS-yes:%=doc/%-all.pod) $(COMPONENTS-yes:%=doc/%.pod) $(LIBRARIES-yes:%=doc/%.pod)
HTMLPAGES = $(AVPROGS-yes:%=doc/%.html) $(AVPROGS-yes:%=doc/%-all.html) $(COMPONENTS-yes:%=doc/%.html) $(LIBRARIES-yes:%=doc/%.html) \
doc/developer.html \
doc/faq.html \
doc/fate.html \
doc/general.html \
doc/git-howto.html \
doc/mailing-list-faq.html \
doc/nut.html \
doc/platform.html \
TXTPAGES = doc/fate.txt \
DOCS-$(CONFIG_HTMLPAGES) += $(HTMLPAGES)
DOCS-$(CONFIG_PODPAGES) += $(PODPAGES)
DOCS-$(CONFIG_MANPAGES) += $(MANPAGES)
DOCS-$(CONFIG_TXTPAGES) += $(TXTPAGES)
DOCS = $(DOCS-yes)
all-$(CONFIG_DOC): doc
doc: documentation
apidoc: doc/doxy/html
documentation: $(DOCS)
TEXIDEP = perl $(SRC_PATH)/doc/texidep.pl $(SRC_PATH) $< $@ >$(@:%=%.d)
doc/%.txt: TAG = TXT
doc/%.txt: doc/%.texi
$(Q)$(TEXIDEP)
$(M)makeinfo --force --no-headers -o $@ $< 2>/dev/null
GENTEXI = format codec
GENTEXI := $(GENTEXI:%=doc/avoptions_%.texi)
$(GENTEXI): TAG = GENTEXI
$(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
$(M)doc/print_options $* > $@
doc/%.html: TAG = HTML
doc/%-all.html: TAG = HTML
ifdef HAVE_MAKEINFO_HTML
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
else
doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
endif
doc/%.pod: TAG = POD
doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-not-all=yes -Idoc $< $@
doc/%-all.pod: TAG = POD
doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
$(Q)$(TEXIDEP)
$(M)perl $(SRC_PATH)/doc/texi2pod.pl -Dconfig-all=yes -Idoc $< $@
doc/%.1 doc/%.3: TAG = MAN
doc/%.1: doc/%.pod $(GENTEXI)
$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
doc/%.3: doc/%.pod $(GENTEXI)
$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
$(DOCS) doc/doxy/html: | doc/
DOXY_INPUT = $(INSTHEADERS)
DOXY_INPUT_DEPS = $(addprefix $(SRC_PATH)/, $(DOXY_INPUT)) ffbuild/config.mak
doc/doxy/html: TAG = DOXY
doc/doxy/html: $(SRC_PATH)/doc/Doxyfile $(SRC_PATH)/doc/doxy-wrapper.sh $(DOXY_INPUT_DEPS)
$(M)OUT_DIR=$$PWD/doc/doxy; cd $(SRC_PATH); ./doc/doxy-wrapper.sh $$OUT_DIR $< $(DOXYGEN) $(DOXY_INPUT);
install-doc: install-html install-man
install-html:
install-man:
ifdef CONFIG_HTMLPAGES
install-progs-$(CONFIG_DOC): install-html
install-html: $(HTMLPAGES)
$(Q)mkdir -p "$(DOCDIR)"
$(INSTALL) -m 644 $(HTMLPAGES) "$(DOCDIR)"
endif
ifdef CONFIG_MANPAGES
install-progs-$(CONFIG_DOC): install-man
install-man: $(MANPAGES)
$(Q)mkdir -p "$(MANDIR)/man1"
$(INSTALL) -m 644 $(MANPAGES1) "$(MANDIR)/man1"
$(Q)mkdir -p "$(MANDIR)/man3"
$(INSTALL) -m 644 $(MANPAGES3) "$(MANDIR)/man3"
endif
uninstall: uninstall-doc
uninstall-doc: uninstall-html uninstall-man
uninstall-html:
$(RM) -r "$(DOCDIR)"
uninstall-man:
$(RM) $(addprefix "$(MANDIR)/man1/",$(AVPROGS-yes:%=%.1) $(AVPROGS-yes:%=%-all.1) $(COMPONENTS-yes:%=%.1))
$(RM) $(addprefix "$(MANDIR)/man3/",$(LIBRARIES-yes:%=%.3))
clean:: docclean
distclean:: docclean
$(RM) doc/config.texi
docclean::
$(RM) $(CLEANSUFFIXES:%=doc/%)
$(RM) $(TXTPAGES) doc/*.html doc/*.pod doc/*.1 doc/*.3 doc/avoptions_*.texi
$(RM) -r doc/doxy/html
-include $(wildcard $(DOCS:%=%.d))
.PHONY: apidoc doc documentation

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

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externals/ffmpeg/doc/bitstream_filters.texi vendored Executable file
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@chapter Bitstream Filters
@c man begin BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option @code{--list-bsfs}.
You can disable all the bitstream filters using the configure option
@code{--disable-bsfs}, and selectively enable any bitstream filter using
the option @code{--enable-bsf=BSF}, or you can disable a particular
bitstream filter using the option @code{--disable-bsf=BSF}.
The option @code{-bsfs} of the ff* tools will display the list of
all the supported bitstream filters included in your build.
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 '='.
@example
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
@end example
Below is a description of the currently available bitstream filters,
with their parameters, if any.
@section 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.
@section av1_metadata
Modify metadata embedded in an AV1 stream.
@table @option
@item td
Insert or remove temporal delimiter OBUs in all temporal units of the
stream.
@table @samp
@item insert
Insert a TD at the beginning of every TU which does not already have one.
@item remove
Remove the TD from the beginning of every TU which has one.
@end table
@item color_primaries
@item transfer_characteristics
@item matrix_coefficients
Set the color description fields in the stream (see AV1 section 6.4.2).
@item 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).
@table @samp
@item tv
Limited range.
@item pc
Full range.
@end table
@item 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.
@table @samp
@item vertical
Left position (matching the default in MPEG-2 and H.264).
@item colocated
Top-left position.
@end table
@item tick_rate
Set the tick rate (@emph{num_units_in_display_tick / time_scale}) in
the timing info in the sequence header.
@item num_ticks_per_picture
Set the number of ticks in each picture, to indicate that the stream
has a fixed framerate. Ignored if @option{tick_rate} is not also set.
@item delete_padding
Deletes Padding OBUs.
@end table
@section chomp
Remove zero padding at the end of a packet.
@section dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as
DTS-HD.
@section 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.
@table @option
@item freq
The additional argument specifies which packets should be filtered.
It accepts the values:
@table @samp
@item k
@item keyframe
add extradata to all key packets
@item e
@item all
add extradata to all packets
@end table
@end table
If not specified it is assumed @samp{k}.
For example the following @command{ffmpeg} command forces a global
header (thus disabling individual packet headers) in the H.264 packets
generated by the @code{libx264} encoder, but corrects them by adding
the header stored in extradata to the key packets:
@example
ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts
@end example
@section eac3_core
Extract the core from a E-AC-3 stream, dropping extra channels.
@section 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.
@table @option
@item remove
When this option is enabled, the long-term headers are removed from the
bitstream after extraction.
@end table
@section filter_units
Remove units with types in or not in a given set from the stream.
@table @option
@item 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 '-'.
@item remove_types
Identical to @option{pass_types}, except the units in the given set
removed and all others passed through.
@end table
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:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT
@end example
To remove all AUDs, SEI and filler from an H.265 stream:
@example
ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT
@end example
@section hapqa_extract
Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an HAPQ or an HAPAlphaOnly file.
@table @option
@item texture
Specifies the texture to keep.
@table @option
@item color
@item alpha
@end table
@end table
Convert HAPQA to HAPQ
@example
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
@end example
Convert HAPQA to HAPAlphaOnly
@example
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
@end example
@section h264_metadata
Modify metadata embedded in an H.264 stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio of the stream in the VUI parameters.
@item overscan_appropriate_flag
Set whether the stream is suitable for display using overscan
or not (see H.264 section E.2.1).
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.264 section E.2.1 and
table E-2).
@item colour_primaries
@item transfer_characteristics
@item 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).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.264 section
E.2.1 and figure E-1).
@item tick_rate
Set the tick rate (num_units_in_tick / time_scale) 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).
@item 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).
@item crop_left
@item crop_right
@item crop_top
@item 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).
@item sei_user_data
Insert a string as SEI unregistered user data. The argument must
be of the form @emph{UUID+string}, where the UUID is as hex digits
possibly separated by hyphens, and the string can be anything.
For example, @samp{086f3693-b7b3-4f2c-9653-21492feee5b8+hello} will
insert the string ``hello'' associated with the given UUID.
@item delete_filler
Deletes both filler NAL units and filler SEI messages.
@item 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, @samp{4.2}), a
level_idc value (for example, @samp{42}), or the special name @samp{auto}
indicating that the filter should attempt to guess the level from the
input stream properties.
@end table
@section h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code
prefixed mode (as defined in the Annex B of the ITU-T H.264
specification).
This is required by some streaming formats, typically the MPEG-2
transport stream format (muxer @code{mpegts}).
For example to remux an MP4 file containing an H.264 stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw H.264 (muxer @code{h264}) output formats.
@section 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.
A new single global PPS is created, and all of the redundant PPSs
within the stream are removed.
@section hevc_metadata
Modify metadata embedded in an HEVC stream.
@table @option
@item aud
Insert or remove AUD NAL units in all access units of the stream.
@table @samp
@item insert
@item remove
@end table
@item sample_aspect_ratio
Set the sample aspect ratio in the stream in the VUI parameters.
@item video_format
@item video_full_range_flag
Set the video format in the stream (see H.265 section E.3.1 and
table E.2).
@item colour_primaries
@item transfer_characteristics
@item 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).
@item chroma_sample_loc_type
Set the chroma sample location in the stream (see H.265 section
E.3.1 and figure E.1).
@item tick_rate
Set the tick rate in the VPS and VUI parameters (num_units_in_tick /
time_scale). Combined with @option{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.
@item 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 @option{tick_rate} is not also set.
@item crop_left
@item crop_right
@item crop_top
@item 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).
@item 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, @samp{5.1}), a
@emph{general_level_idc} value (for example, @samp{153} for level 5.1),
or the special name @samp{auto} indicating that the filter should
attempt to guess the level from the input stream properties.
@end table
@section 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 @code{mpegts}).
For example to remux an MP4 file containing an HEVC stream to mpegts
format with @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
@end example
Please note that this filter is auto-inserted for MPEG-TS (muxer
@code{mpegts}) and raw HEVC/H.265 (muxer @code{h265} or
@code{hevc}) output formats.
@section 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
@option{-tag:v}.
For example, to remux 30 MB/sec NTSC IMX to MOV:
@example
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
@end example
@section mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a
JPEG image. The individual frames can be extracted without loss,
e.g. by
@example
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
@end example
Unfortunately, these chunks are incomplete JPEG images, because
they lack the DHT segment required for decoding. Quoting from
@url{http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml}:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001,
commented that "MJPEG, or at least the MJPEG in AVIs having the
MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* --
Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2,
and it must use basic Huffman encoding, not arithmetic or
progressive. . . . You can indeed extract the MJPEG frames and
decode them with a regular JPEG decoder, but you have to prepend
the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in
the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG
stream (carrying the AVI1 header ID and lacking a DHT segment) to
produce fully qualified JPEG images.
@example
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
@end example
@section mjpegadump
Add an MJPEG A header to the bitstream, to enable decoding by
Quicktime.
@anchor{mov2textsub}
@section mov2textsub
Extract a representable text file from MOV subtitles, stripping the
metadata header from each subtitle packet.
See also the @ref{text2movsub} filter.
@section mp3decomp
Decompress non-standard compressed MP3 audio headers.
@section mpeg2_metadata
Modify metadata embedded in an MPEG-2 stream.
@table @option
@item display_aspect_ratio
Set the display aspect ratio in the stream.
The following fixed values are supported:
@table @option
@item 4/3
@item 16/9
@item 221/100
@end table
Any other value will result in square pixels being signalled instead
(see H.262 section 6.3.3 and table 6-3).
@item 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).
@item video_format
Set the video format in the stream (see H.262 section 6.3.6 and
table 6-6).
@item colour_primaries
@item transfer_characteristics
@item 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).
@end table
@section 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 @command{ffmpeg}, you can use the command:
@example
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
@end example
@section 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:
@table @option
@item amount
A numeral string, whose value is related to how often output bytes will
be modified. Therefore, values below or equal to 0 are forbidden, and
the lower the more frequent bytes will be modified, with 1 meaning
every byte is modified.
@item dropamount
A numeral string, whose value is related to how often packets will be dropped.
Therefore, values below or equal to 0 are forbidden, and the lower the more
frequent packets will be dropped, with 1 meaning every packet is dropped.
@end table
The following example applies the modification to every byte but does not drop
any packets.
@example
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
@end example
@section null
This bitstream filter passes the packets through unchanged.
@section 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 @ref{asetnsamples,,asetnsamples audio
filter,ffmpeg-filters} but works on audio packets instead of audio frames.
@table @option
@item nb_out_samples, n
Set the number of samples per each output audio packet. The number is intended
as the number of samples @emph{per each channel}. Default value is 1024.
@item 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 @option{frame_rate}) as the previous ones. Default value is 1.
@item 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 @option{nb_out_samples}.
@end table
You can generate the well known 1602-1601-1602-1601-1602 pattern of 48kHz audio
for NTSC frame rate using the @option{frame_rate} option.
@example
ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -
@end example
@section prores_metadata
Modify color property metadata embedded in prores stream.
@table @option
@item color_primaries
Set the color primaries.
Available values are:
@table @samp
@item auto
Keep the same color primaries property (default).
@item unknown
@item bt709
@item bt470bg
BT601 625
@item smpte170m
BT601 525
@item bt2020
@item smpte431
DCI P3
@item smpte432
P3 D65
@end table
@item transfer_characteristics
Set the color transfer.
Available values are:
@table @samp
@item auto
Keep the same transfer characteristics property (default).
@item unknown
@item bt709
BT 601, BT 709, BT 2020
@item smpte2084
SMPTE ST 2084
@item arib-std-b67
ARIB STD-B67
@end table
@item matrix_coefficients
Set the matrix coefficient.
Available values are:
@table @samp
@item auto
Keep the same colorspace property (default).
@item unknown
@item bt709
@item smpte170m
BT 601
@item bt2020nc
@end table
@end table
Set Rec709 colorspace for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov
@end example
Set Hybrid Log-Gamma parameters for each frame of the file
@example
ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov
@end example
@section remove_extra
Remove extradata from packets.
It accepts the following parameter:
@table @option
@item freq
Set which frame types to remove extradata from.
@table @samp
@item k
Remove extradata from non-keyframes only.
@item keyframe
Remove extradata from keyframes only.
@item e, all
Remove extradata from all frames.
@end table
@end table
@anchor{text2movsub}
@section text2movsub
Convert text subtitles to MOV subtitles (as used by the @code{mov_text}
codec) with metadata headers.
See also the @ref{mov2textsub} filter.
@section 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.
@section truehd_core
Extract the core from a TrueHD stream, dropping ATMOS data.
@section vp9_metadata
Modify metadata embedded in a VP9 stream.
@table @option
@item 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.
@table @samp
@item unknown
@item bt601
@item bt709
@item smpte170
@item smpte240
@item bt2020
@item rgb
@end table
@item 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.
@table @samp
@item tv
@item pc
@end table
@end table
@section 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.
@section vp9_superframe_split
Split VP9 superframes into single frames.
@section 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.
@c man end BITSTREAM FILTERS

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FFmpeg currently uses a custom build system, this text attempts to document
some of its obscure features and options.
Makefile variables:
V
Disable the default terse mode, the full command issued by make and its
output will be shown on the screen.
DBG
Preprocess x86 external assembler files to a .dbg.asm file in the object
directory, which then gets compiled. Helps in developing those assembler
files.
DESTDIR
Destination directory for the install targets, useful to prepare packages
or install FFmpeg in cross-environments.
GEN
Set to ‘1’ to generate the missing or mismatched references.
Makefile targets:
all
Default target, builds all the libraries and the executables.
fate
Run the fate test suite, note that you must have installed it.
fate-list
List all fate/regression test targets.
install
Install headers, libraries and programs.
examples
Build all examples located in doc/examples.
checkheaders
Check headers dependencies.
alltools
Build all tools in tools directory.
config
Reconfigure the project with the current configuration.
tools/target_dec_<decoder>_fuzzer
Build fuzzer to fuzz the specified decoder.
tools/target_bsf_<filter>_fuzzer
Build fuzzer to fuzz the specified bitstream filter.
Useful standard make commands:
make -t <target>
Touch all files that otherwise would be built, this is useful to reduce
unneeded rebuilding when changing headers, but note that you must force rebuilds
of files that actually need it by hand then.
make -j<num>
Rebuild with multiple jobs at the same time. Faster on multi processor systems.
make -k
Continue build in case of errors, this is useful for the regression tests
sometimes but note that it will still not run all reg tests.

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@chapter Decoders
@c man begin DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of
multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding @code{--enable-lib} option. You can list all
available decoders using the configure option @code{--list-decoders}.
You can disable all the decoders with the configure option
@code{--disable-decoders} and selectively enable / disable single decoders
with the options @code{--enable-decoder=@var{DECODER}} /
@code{--disable-decoder=@var{DECODER}}.
The option @code{-decoders} of the ff* tools will display the list of
enabled decoders.
@c man end DECODERS
@chapter Video Decoders
@c man begin VIDEO DECODERS
A description of some of the currently available video decoders
follows.
@section rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
@subsection Options
@table @option
@item top @var{top_field_first}
Specify the assumed field type of the input video.
@table @option
@item -1
the video is assumed to be progressive (default)
@item 0
bottom-field-first is assumed
@item 1
top-field-first is assumed
@end table
@end table
@section libdav1d
dav1d AV1 decoder.
libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
Requires the presence of the libdav1d headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libdav1d}.
@subsection Options
The following options are supported by the libdav1d wrapper.
@table @option
@item framethreads
Set amount of frame threads to use during decoding. The default value is 0 (autodetect).
@item tilethreads
Set amount of tile threads to use during decoding. The default value is 0 (autodetect).
@item filmgrain
Apply film grain to the decoded video if present in the bitstream. Defaults to the
internal default of the library.
@item oppoint
Select an operating point of a scalable AV1 bitstream (0 - 31). Defaults to the
internal default of the library.
@item alllayers
Output all spatial layers of a scalable AV1 bitstream. The default value is false.
@end table
@section libdavs2
AVS2-P2/IEEE1857.4 video decoder wrapper.
This decoder allows libavcodec to decode AVS2 streams with davs2 library.
@c man end VIDEO DECODERS
@chapter Audio Decoders
@c man begin AUDIO DECODERS
A description of some of the currently available audio decoders
follows.
@section ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as
the undocumented RealAudio 3 (a.k.a. dnet).
@subsection AC-3 Decoder Options
@table @option
@item -drc_scale @var{value}
Dynamic Range Scale Factor. The factor to apply to dynamic range values
from the AC-3 stream. This factor is applied exponentially.
There are 3 notable scale factor ranges:
@table @option
@item drc_scale == 0
DRC disabled. Produces full range audio.
@item 0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value.
Audio reproduction is between full range and full compression.
@item drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically.
Loud sounds are fully compressed. Soft sounds are enhanced.
@end table
@end table
@section flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
@subsection FLAC Decoder options
@table @option
@item -use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values
(like the default value). This option makes it possible to decode such streams
correctly by using lavc's old buggy lpc logic for decoding.
@end table
@section ffwavesynth
Internal wave synthesizer.
This decoder generates wave patterns according to predefined sequences. Its
use is purely internal and the format of the data it accepts is not publicly
documented.
@section libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with @code{--enable-libcelt}.
@section libgsm
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with @code{--enable-libgsm}.
This decoder supports both the ordinary GSM and the Microsoft variant.
@section libilbc
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libilbc}.
@subsection Options
The following option is supported by the libilbc wrapper.
@table @option
@item enhance
Enable the enhancement of the decoded audio when set to 1. The default
value is 0 (disabled).
@end table
@section libopencore-amrnb
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrnb}.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB
without this library.
@section libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with @code{--enable-libopencore-amrwb}.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB
without this library.
@section libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libopus}.
An FFmpeg native decoder for Opus exists, so users can decode Opus
without this library.
@c man end AUDIO DECODERS
@chapter Subtitles Decoders
@c man begin SUBTILES DECODERS
@section libaribb24
ARIB STD-B24 caption decoder.
Implements profiles A and C of the ARIB STD-B24 standard.
@subsection libaribb24 Decoder Options
@table @option
@item -aribb24-base-path @var{path}
Sets the base path for the libaribb24 library. This is utilized for reading of
configuration files (for custom unicode conversions), and for dumping of
non-text symbols as images under that location.
Unset by default.
@item -aribb24-skip-ruby-text @var{boolean}
Tells the decoder wrapper to skip text blocks that contain half-height ruby
text.
Enabled by default.
@end table
@section dvbsub
@subsection Options
@table @option
@item compute_clut
@table @option
@item -1
Compute clut if no matching CLUT is in the stream.
@item 0
Never compute CLUT
@item 1
Always compute CLUT and override the one provided in the stream.
@end table
@item dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.
@end table
@section dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
also be found in VobSub file pairs and in some Matroska files.
@subsection Options
@table @option
@item palette
Specify the global palette used by the bitmaps. When stored in VobSub, the
palette is normally specified in the index file; in Matroska, the palette is
stored in the codec extra-data in the same format as in VobSub. In DVDs, the
palette is stored in the IFO file, and therefore not available when reading
from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by commas, for example @code{0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
@item ifo_palette
Specify the IFO file from which the global palette is obtained.
(experimental)
@item forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced
and non-forced subtitles in the same track. Setting this flag to @code{1}
will only keep the forced subtitles. Default value is @code{0}.
@end table
@section libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
subtitles. Requires the presence of the libzvbi headers and library during
configuration. You need to explicitly configure the build with
@code{--enable-libzvbi}.
@subsection Options
@table @option
@item txt_page
List of teletext page numbers to decode. Pages that do not match the specified
list are dropped. You may use the special @code{*} string to match all pages,
or @code{subtitle} to match all subtitle pages.
Default value is *.
@item txt_default_region
Set default character set used for decoding, a value between 0 and 87 (see
ETS 300 706, Section 15, Table 32). Default value is -1, which does not
override the libzvbi default. This option is needed for some legacy level 1.0
transmissions which cannot signal the proper charset.
@item txt_chop_top
Discards the top teletext line. Default value is 1.
@item txt_format
Specifies the format of the decoded subtitles.
@table @option
@item bitmap
The default format, you should use this for teletext pages, because certain
graphics and colors cannot be expressed in simple text or even ASS.
@item text
Simple text based output without formatting.
@item ass
Formatted ASS output, subtitle pages and teletext pages are returned in
different styles, subtitle pages are stripped down to text, but an effort is
made to keep the text alignment and the formatting.
@end table
@item txt_left
X offset of generated bitmaps, default is 0.
@item txt_top
Y offset of generated bitmaps, default is 0.
@item txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated
text. This option is useful for teletext based subtitles where empty spaces may
be present at the start or at the end of the lines or empty lines may be
present between the subtitle lines because of double-sized teletext characters.
Default value is 1.
@item txt_duration
Sets the display duration of the decoded teletext pages or subtitles in
milliseconds. Default value is -1 which means infinity or until the next
subtitle event comes.
@item txt_transparent
Force transparent background of the generated teletext bitmaps. Default value
is 0 which means an opaque background.
@item txt_opacity
Sets the opacity (0-255) of the teletext background. If
@option{txt_transparent} is not set, it only affects characters between a start
box and an end box, typically subtitles. Default value is 0 if
@option{txt_transparent} is set, 255 otherwise.
@end table
@c man end SUBTILES DECODERS

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a.summary-letter {
text-decoration: none;
}
a {
color: #2D6198;
}
a:visited {
color: #884488;
}
#banner {
background-color: white;
position: relative;
text-align: center;
}
#banner img {
margin-bottom: 1px;
margin-top: 5px;
}
#body {
margin-left: 1em;
margin-right: 1em;
}
body {
background-color: #313131;
margin: 0;
text-align: justify;
}
.center {
margin-left: auto;
margin-right: auto;
text-align: center;
}
#container {
background-color: white;
color: #202020;
margin-left: 1em;
margin-right: 1em;
}
#footer {
text-align: center;
}
h1 a, h2 a, h3 a, h4 a {
text-decoration: inherit;
color: inherit;
}
h1, h2, h3, h4 {
padding-left: 0.4em;
border-radius: 4px;
padding-bottom: 0.25em;
padding-top: 0.25em;
border: 1px solid #6A996A;
}
h1 {
background-color: #7BB37B;
color: #151515;
font-size: 1.2em;
padding-bottom: 0.3em;
padding-top: 0.3em;
}
h2 {
color: #313131;
font-size: 1.0em;
background-color: #ABE3AB;
}
h3 {
color: #313131;
font-size: 0.9em;
margin-bottom: -6px;
background-color: #BBF3BB;
}
h4 {
color: #313131;
font-size: 0.8em;
margin-bottom: -8px;
background-color: #D1FDD1;
}
img {
border: 0;
}
#navbar {
background-color: #738073;
border-bottom: 1px solid #5C665C;
border-top: 1px solid #5C665C;
margin-top: 12px;
padding: 0.3em;
position: relative;
text-align: center;
}
#navbar a, #navbar_secondary a {
color: white;
padding: 0.3em;
text-decoration: none;
}
#navbar a:hover, #navbar_secondary a:hover {
background-color: #313131;
color: white;
text-decoration: none;
}
#navbar_secondary {
background-color: #738073;
border-bottom: 1px solid #5C665C;
border-left: 1px solid #5C665C;
border-right: 1px solid #5C665C;
padding: 0.3em;
position: relative;
text-align: center;
}
p {
margin-left: 1em;
margin-right: 1em;
}
pre {
margin-left: 3em;
margin-right: 3em;
padding: 0.3em;
border: 1px solid #bbb;
background-color: #f7f7f7;
}
dl dt {
font-weight: bold;
}
#proj_desc {
font-size: 1.2em;
}
#repos {
margin-left: 1em;
margin-right: 1em;
border-collapse: collapse;
border: solid 1px #6A996A;
}
#repos th {
background-color: #7BB37B;
border: solid 1px #6A996A;
}
#repos td {
padding: 0.2em;
border: solid 1px #6A996A;
}

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@chapter Demuxers
@c man begin DEMUXERS
Demuxers are configured elements in FFmpeg that can read the
multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers
are enabled by default. You can list all available ones using the
configure option @code{--list-demuxers}.
You can disable all the demuxers using the configure option
@code{--disable-demuxers}, and selectively enable a single demuxer with
the option @code{--enable-demuxer=@var{DEMUXER}}, or disable it
with the option @code{--disable-demuxer=@var{DEMUXER}}.
The option @code{-demuxers} of the ff* tools will display the list of
enabled demuxers. Use @code{-formats} to view a combined list of
enabled demuxers and muxers.
The description of some of the currently available demuxers follows.
@section aa
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
@section apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files.
All headers, but the PNG signature, up to (but not including) the first
fcTL chunk are transmitted as extradata.
Frames are then split as being all the chunks between two fcTL ones, or
between the last fcTL and IEND chunks.
@table @option
@item -ignore_loop @var{bool}
Ignore the loop variable in the file if set.
@item -max_fps @var{int}
Maximum framerate in frames per second (0 for no limit).
@item -default_fps @var{int}
Default framerate in frames per second when none is specified in the file
(0 meaning as fast as possible).
@end table
@section asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
@table @option
@item -no_resync_search @var{bool}
Do not try to resynchronize by looking for a certain optional start code.
@end table
@anchor{concat}
@section concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and
demuxes them one after the other, as if all their packets had been muxed
together.
The timestamps in the files are adjusted so that the first file starts at 0
and each next file starts where the previous one finishes. Note that it is
done globally and may cause gaps if all streams do not have exactly the same
length.
All files must have the same streams (same codecs, same time base, etc.).
The duration of each file is used to adjust the timestamps of the next file:
if the duration is incorrect (because it was computed using the bit-rate or
because the file is truncated, for example), it can cause artifacts. The
@code{duration} directive can be used to override the duration stored in
each file.
@subsection Syntax
The script is a text file in extended-ASCII, with one directive per line.
Empty lines, leading spaces and lines starting with '#' are ignored. The
following directive is recognized:
@table @option
@item @code{file @var{path}}
Path to a file to read; special characters and spaces must be escaped with
backslash or single quotes.
All subsequent file-related directives apply to that file.
@item @code{ffconcat version 1.0}
Identify the script type and version. It also sets the @option{safe} option
to 1 if it was -1.
To make FFmpeg recognize the format automatically, this directive must
appear exactly as is (no extra space or byte-order-mark) on the very first
line of the script.
@item @code{duration @var{dur}}
Duration of the file. This information can be specified from the file;
specifying it here may be more efficient or help if the information from the
file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the
whole concatenated video.
@item @code{inpoint @var{timestamp}}
In point of the file. When the demuxer opens the file it instantly seeks to the
specified timestamp. Seeking is done so that all streams can be presented
successfully at In point.
This directive works best with intra frame codecs, because for non-intra frame
ones you will usually get extra packets before the actual In point and the
decoded content will most likely contain frames before In point too.
For each file, packets before the file In point will have timestamps less than
the calculated start timestamp of the file (negative in case of the first
file), and the duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified In point.
Because of potential packets before the specified In point, packet timestamps
may overlap between two concatenated files.
@item @code{outpoint @var{timestamp}}
Out point of the file. When the demuxer reaches the specified decoding
timestamp in any of the streams, it handles it as an end of file condition and
skips the current and all the remaining packets from all streams.
Out point is exclusive, which means that the demuxer will not output packets
with a decoding timestamp greater or equal to Out point.
This directive works best with intra frame codecs and formats where all streams
are tightly interleaved. For non-intra frame codecs you will usually get
additional packets with presentation timestamp after Out point therefore the
decoded content will most likely contain frames after Out point too. If your
streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest
stream until Out point.
The duration of the files (if not specified by the @code{duration}
directive) will be reduced based on their specified Out point.
@item @code{file_packet_metadata @var{key=value}}
Metadata of the packets of the file. The specified metadata will be set for
each file packet. You can specify this directive multiple times to add multiple
metadata entries.
@item @code{stream}
Introduce a stream in the virtual file.
All subsequent stream-related directives apply to the last introduced
stream.
Some streams properties must be set in order to allow identifying the
matching streams in the subfiles.
If no streams are defined in the script, the streams from the first file are
copied.
@item @code{exact_stream_id @var{id}}
Set the id of the stream.
If this directive is given, the string with the corresponding id in the
subfiles will be used.
This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
@end table
@subsection Options
This demuxer accepts the following option:
@table @option
@item safe
If set to 1, reject unsafe file paths. A file path is considered safe if it
does not contain a protocol specification and is relative and all components
only contain characters from the portable character set (letters, digits,
period, underscore and hyphen) and have no period at the beginning of a
component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically
probed and 0 otherwise.
@item auto_convert
If set to 1, try to perform automatic conversions on packet data to make the
streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream
filter to H.264 streams in MP4 format. This is necessary in particular if
there are resolution changes.
@item segment_time_metadata
If set to 1, every packet will contain the @var{lavf.concat.start_time} and the
@var{lavf.concat.duration} packet metadata values which are the start_time and
the duration of the respective file segments in the concatenated output
expressed in microseconds. The duration metadata is only set if it is known
based on the concat file.
The default is 0.
@end table
@subsection Examples
@itemize
@item
Use absolute filenames and include some comments:
@example
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
@end example
@item
Allow for input format auto-probing, use safe filenames and set the duration of
the first file:
@example
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
@end example
@end itemize
@section dash
Dynamic Adaptive Streaming over HTTP demuxer.
This demuxer presents all AVStreams found in the manifest.
By setting the discard flags on AVStreams the caller can decide
which streams to actually receive.
Each stream mirrors the @code{id} and @code{bandwidth} properties from the
@code{<Representation>} as metadata keys named "id" and "variant_bitrate" respectively.
@section flv, live_flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
@example
ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
@end example
@table @option
@item -flv_metadata @var{bool}
Allocate the streams according to the onMetaData array content.
@item -flv_ignore_prevtag @var{bool}
Ignore the size of previous tag value.
@item -flv_full_metadata @var{bool}
Output all context of the onMetadata.
@end table
@section gif
Animated GIF demuxer.
It accepts the following options:
@table @option
@item min_delay
Set the minimum valid delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 2.
@item max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds.
Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
the maximum value allowed by the specification.
@item default_delay
Set the default delay between frames in hundredths of seconds.
Range is 0 to 6000. Default value is 10.
@item ignore_loop
GIF files can contain information to loop a certain number of times (or
infinitely). If @option{ignore_loop} is set to 1, then the loop setting
from the input will be ignored and looping will not occur. If set to 0,
then looping will occur and will cycle the number of times according to
the GIF. Default value is 1.
@end table
For example, with the overlay filter, place an infinitely looping GIF
over another video:
@example
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
@end example
Note that in the above example the shortest option for overlay filter is
used to end the output video at the length of the shortest input file,
which in this case is @file{input.mp4} as the GIF in this example loops
infinitely.
@section hls
HLS demuxer
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams.
The id field is set to the bitrate variant index number. By setting
the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay),
the caller can decide which variant streams to actually receive.
The total bitrate of the variant that the stream belongs to is
available in a metadata key named "variant_bitrate".
It accepts the following options:
@table @option
@item live_start_index
segment index to start live streams at (negative values are from the end).
@item allowed_extensions
',' separated list of file extensions that hls is allowed to access.
@item max_reload
Maximum number of times a insufficient list is attempted to be reloaded.
Default value is 1000.
@item m3u8_hold_counters
The maximum number of times to load m3u8 when it refreshes without new segments.
Default value is 1000.
@item http_persistent
Use persistent HTTP connections. Applicable only for HTTP streams.
Enabled by default.
@item http_multiple
Use multiple HTTP connections for downloading HTTP segments.
Enabled by default for HTTP/1.1 servers.
@item http_seekable
Use HTTP partial requests for downloading HTTP segments.
0 = disable, 1 = enable, -1 = auto, Default is auto.
@end table
@section image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The syntax and meaning of the pattern is specified by the
option @var{pattern_type}.
The pattern may contain a suffix which is used to automatically
determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the
same for all the files in the sequence.
This demuxer accepts the following options:
@table @option
@item framerate
Set the frame rate for the video stream. It defaults to 25.
@item loop
If set to 1, loop over the input. Default value is 0.
@item pattern_type
Select the pattern type used to interpret the provided filename.
@var{pattern_type} accepts one of the following values.
@table @option
@item none
Disable pattern matching, therefore the video will only contain the specified
image. You should use this option if you do not want to create sequences from
multiple images and your filenames may contain special pattern characters.
@item sequence
Select a sequence pattern type, used to specify a sequence of files
indexed by sequential numbers.
A sequence pattern may contain the string "%d" or "%0@var{N}d", which
specifies the position of the characters representing a sequential
number in each filename matched by the pattern. If the form
"%d0@var{N}d" is used, the string representing the number in each
filename is 0-padded and @var{N} is the total number of 0-padded
digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0@var{N}d", the first filename of
the file list specified by the pattern must contain a number
inclusively contained between @var{start_number} and
@var{start_number}+@var{start_number_range}-1, and all the following
numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of
filenames of the form @file{img-001.bmp}, @file{img-002.bmp}, ...,
@file{img-010.bmp}, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form @file{i%m%g-1.jpg},
@file{i%m%g-2.jpg}, ..., @file{i%m%g-10.jpg}, etc.
Note that the pattern must not necessarily contain "%d" or
"%0@var{N}d", for example to convert a single image file
@file{img.jpeg} you can employ the command:
@example
ffmpeg -i img.jpeg img.png
@end example
@item glob
Select a glob wildcard pattern type.
The pattern is interpreted like a @code{glob()} pattern. This is only
selectable if libavformat was compiled with globbing support.
@item glob_sequence @emph{(deprecated, will be removed)}
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
@code{%*?[]@{@}} that is preceded by an unescaped "%", the pattern is
interpreted like a @code{glob()} pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters @code{%*?[]@{@}} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern @code{foo-%*.jpeg} will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
@code{foo-%?%?%?.jpeg} will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of @var{glob} and
@var{sequence}.
@end table
Default value is @var{glob_sequence}.
@item pixel_format
Set the pixel format of the images to read. If not specified the pixel
format is guessed from the first image file in the sequence.
@item start_number
Set the index of the file matched by the image file pattern to start
to read from. Default value is 0.
@item start_number_range
Set the index interval range to check when looking for the first image
file in the sequence, starting from @var{start_number}. Default value
is 5.
@item ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note
that monotonity of timestamps is not provided: images go in the same order as
without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in
nanosecond precision.
@item video_size
Set the video size of the images to read. If not specified the video
size is guessed from the first image file in the sequence.
@item export_path_metadata
If set to 1, will add two extra fields to the metadata found in input, making them
also available for other filters (see @var{drawtext} filter for examples). Default
value is 0. The extra fields are described below:
@table @option
@item lavf.image2dec.source_path
Corresponds to the full path to the input file being read.
@item lavf.image2dec.source_basename
Corresponds to the name of the file being read.
@end table
@end table
@subsection Examples
@itemize
@item
Use @command{ffmpeg} for creating a video from the images in the file
sequence @file{img-001.jpeg}, @file{img-002.jpeg}, ..., assuming an
input frame rate of 10 frames per second:
@example
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
@end example
@item
As above, but start by reading from a file with index 100 in the sequence:
@example
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
@end example
@item
Read images matching the "*.png" glob pattern , that is all the files
terminating with the ".png" suffix:
@example
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
@end example
@end itemize
@section libgme
The Game Music Emu library is a collection of video game music file emulators.
See @url{https://bitbucket.org/mpyne/game-music-emu/overview} for more information.
It accepts the following options:
@table @option
@item track_index
Set the index of which track to demux. The demuxer can only export one track.
Track indexes start at 0. Default is to pick the first track. Number of tracks
is exported as @var{tracks} metadata entry.
@item sample_rate
Set the sampling rate of the exported track. Range is 1000 to 999999. Default is 44100.
@item max_size @emph{(bytes)}
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read.
Default is 50 MiB.
@end table
@section libmodplug
ModPlug based module demuxer
See @url{https://github.com/Konstanty/libmodplug}
It will export one 2-channel 16-bit 44.1 kHz audio stream.
Optionally, a @code{pal8} 16-color video stream can be exported with or without printed metadata.
It accepts the following options:
@table @option
@item noise_reduction
Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default is 0.
@item reverb_depth
Set amount of reverb. Range 0-100. Default is 0.
@item reverb_delay
Set delay in ms, clamped to 40-250 ms. Default is 0.
@item bass_amount
Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet) to 100 (loud). Default is 0.
@item bass_range
Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100 Hz. Default is 0.
@item surround_depth
Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100 (heavy). Default is 0.
@item surround_delay
Set surround delay in ms, clamped to 5-40 ms. Default is 0.
@item max_size
The demuxer buffers the entire file into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of files that can be read. Range is 0 to 100 MiB.
0 removes buffer size limit (not recommended). Default is 5 MiB.
@item video_stream_expr
String which is evaluated using the eval API to assign colors to the generated video stream.
Variables which can be used are @code{x}, @code{y}, @code{w}, @code{h}, @code{t}, @code{speed},
@code{tempo}, @code{order}, @code{pattern} and @code{row}.
@item video_stream
Generate video stream. Can be 1 (on) or 0 (off). Default is 0.
@item video_stream_w
Set video frame width in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_h
Set video frame height in 'chars' where one char indicates 8 pixels. Range is 20-512. Default is 30.
@item video_stream_ptxt
Print metadata on video stream. Includes @code{speed}, @code{tempo}, @code{order}, @code{pattern},
@code{row} and @code{ts} (time in ms). Can be 1 (on) or 0 (off). Default is 1.
@end table
@section libopenmpt
libopenmpt based module demuxer
See @url{https://lib.openmpt.org/libopenmpt/} for more information.
Some files have multiple subsongs (tracks) this can be set with the @option{subsong}
option.
It accepts the following options:
@table @option
@item subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the
subsong. Subsong indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
@item layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts.
The default value is STEREO.
@item sample_rate
Set the sample rate for libopenmpt to output.
Range is from 1000 to INT_MAX. The value default is 48000.
@end table
@section mov/mp4/3gp
Demuxer for Quicktime File Format & ISO/IEC Base Media File Format (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part 12).
Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism, ismv, isma, f4v
@subsection Options
This demuxer accepts the following options:
@table @option
@item enable_drefs
Enable loading of external tracks, disabled by default.
Enabling this can theoretically leak information in some use cases.
@item use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default.
Enabling this poses a security risk. It should only be enabled if the source
is known to be non-malicious.
@item seek_streams_individually
When seeking, identify the closest point in each stream individually and demux packets in
that stream from identified point. This can lead to a different sequence of packets compared
to demuxing linearly from the beginning. Default is true.
@item ignore_editlist
Ignore any edit list atoms. The demuxer, by default, modifies the stream index to reflect the
timeline described by the edit list. Default is false.
@item advanced_editlist
Modify the stream index to reflect the timeline described by the edit list. @code{ignore_editlist}
must be set to false for this option to be effective.
If both @code{ignore_editlist} and this option are set to false, then only the
start of the stream index is modified to reflect initial dwell time or starting timestamp
described by the edit list. Default is true.
@item ignore_chapters
Don't parse chapters. This includes GoPro 'HiLight' tags/moments. Note that chapters are
only parsed when input is seekable. Default is false.
@item use_mfra_for
For seekable fragmented input, set fragment's starting timestamp from media fragment random access box, if present.
Following options are available:
@table @samp
@item auto
Auto-detect whether to set mfra timestamps as PTS or DTS @emph{(default)}
@item dts
Set mfra timestamps as DTS
@item pts
Set mfra timestamps as PTS
@item 0
Don't use mfra box to set timestamps
@end table
@item export_all
Export unrecognized boxes within the @var{udta} box as metadata entries. The first four
characters of the box type are set as the key. Default is false.
@item export_xmp
Export entire contents of @var{XMP_} box and @var{uuid} box as a string with key @code{xmp}. Note that
if @code{export_all} is set and this option isn't, the contents of @var{XMP_} box are still exported
but with key @code{XMP_}. Default is false.
@item activation_bytes
4-byte key required to decrypt Audible AAX and AAX+ files. See Audible AAX subsection below.
@item audible_fixed_key
Fixed key used for handling Audible AAX/AAX+ files. It has been pre-set so should not be necessary to
specify.
@item decryption_key
16-byte key, in hex, to decrypt files encrypted using ISO Common Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).
@end table
@subsection Audible AAX
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
@example
ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
@end example
@section mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
@table @option
@item resync_size
Set size limit for looking up a new synchronization. Default value is
65536.
@item skip_unknown_pmt
Skip PMTs for programs not defined in the PAT. Default value is 0.
@item fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated
from the PCR of the first program which the teletext stream is part of and is
not discarded. Default value is 1, set this option to 0 if you want your
teletext packet PTS and DTS values untouched.
@item ts_packetsize
Output option carrying the raw packet size in bytes.
Show the detected raw packet size, cannot be set by the user.
@item scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from -1
to 1 (-1 means automatic setting, 1 means enabled, 0 means
disabled). Default value is -1.
@item merge_pmt_versions
Re-use existing streams when a PMT's version is updated and elementary
streams move to different PIDs. Default value is 0.
@end table
@section mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.
@table @option
@item strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection,
to prevent regression with numerous existing endpoints not generating a proper MIME
MJPEG stream. Turning this option on by setting it to 1 will result in a stricter check
of the boundary value.
@end table
@section rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no header
specifying the assumed video parameters, the user must specify them
in order to be able to decode the data correctly.
This demuxer accepts the following options:
@table @option
@item framerate
Set input video frame rate. Default value is 25.
@item pixel_format
Set the input video pixel format. Default value is @code{yuv420p}.
@item video_size
Set the input video size. This value must be specified explicitly.
@end table
For example to read a rawvideo file @file{input.raw} with
@command{ffplay}, assuming a pixel format of @code{rgb24}, a video
size of @code{320x240}, and a frame rate of 10 images per second, use
the command:
@example
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
@end example
@section sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen
@url{http://uazu.net/sbagen/} to generate binaural beats sessions. A SBG
script looks like that:
@example
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
@end example
A SBG script can mix absolute and relative timestamps. If the script uses
either only absolute timestamps (including the script start time) or only
relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of
timestamps, then the @var{NOW} reference for relative timestamps will be
taken from the current time of day at the time the script is read, and the
script layout will be frozen according to that reference. That means that if
the script is directly played, the actual times will match the absolute
timestamps up to the sound controller's clock accuracy, but if the user
somehow pauses the playback or seeks, all times will be shifted accordingly.
@section tedcaptions
JSON captions used for @url{http://www.ted.com/, TED Talks}.
TED does not provide links to the captions, but they can be guessed from the
page. The file @file{tools/bookmarklets.html} from the FFmpeg source tree
contains a bookmarklet to expose them.
This demuxer accepts the following option:
@table @option
@item start_time
Set the start time of the TED talk, in milliseconds. The default is 15000
(15s). It is used to sync the captions with the downloadable videos, because
they include a 15s intro.
@end table
Example: convert the captions to a format most players understand:
@example
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
@end example
@section vapoursynth
Vapoursynth wrapper.
Due to security concerns, Vapoursynth scripts will not
be autodetected so the input format has to be forced. For ff* CLI tools,
add @code{-f vapoursynth} before the input @code{-i yourscript.vpy}.
This demuxer accepts the following option:
@table @option
@item max_script_size
The demuxer buffers the entire script into memory. Adjust this value to set the maximum buffer size,
which in turn, acts as a ceiling for the size of scripts that can be read.
Default is 1 MiB.
@end table
@c man end DEMUXERS

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Developer Documentation
@titlepage
@center @titlefont{Developer Documentation}
@end titlepage
@top
@contents
@chapter Notes for external developers
This document is mostly useful for internal FFmpeg developers.
External developers who need to use the API in their application should
refer to the API doxygen documentation in the public headers, and
check the examples in @file{doc/examples} and in the source code to
see how the public API is employed.
You can use the FFmpeg libraries in your commercial program, but you
are encouraged to @emph{publish any patch you make}. In this case the
best way to proceed is to send your patches to the ffmpeg-devel
mailing list following the guidelines illustrated in the remainder of
this document.
For more detailed legal information about the use of FFmpeg in
external programs read the @file{LICENSE} file in the source tree and
consult @url{https://ffmpeg.org/legal.html}.
@chapter Contributing
There are 2 ways by which code gets into FFmpeg:
@itemize @bullet
@item Submitting patches to the ffmpeg-devel mailing list.
See @ref{Submitting patches} for details.
@item Directly committing changes to the main tree.
@end itemize
Whichever way, changes should be reviewed by the maintainer of the code
before they are committed. And they should follow the @ref{Coding Rules}.
The developer making the commit and the author are responsible for their changes
and should try to fix issues their commit causes.
@anchor{Coding Rules}
@chapter Coding Rules
@section Code formatting conventions
There are the following guidelines regarding the indentation in files:
@itemize @bullet
@item
Indent size is 4.
@item
The TAB character is forbidden outside of Makefiles as is any
form of trailing whitespace. Commits containing either will be
rejected by the git repository.
@item
You should try to limit your code lines to 80 characters; however, do so if
and only if this improves readability.
@item
K&R coding style is used.
@end itemize
The presentation is one inspired by 'indent -i4 -kr -nut'.
The main priority in FFmpeg is simplicity and small code size in order to
minimize the bug count.
@section Comments
Use the JavaDoc/Doxygen format (see examples below) so that code documentation
can be generated automatically. All nontrivial functions should have a comment
above them explaining what the function does, even if it is just one sentence.
All structures and their member variables should be documented, too.
Avoid Qt-style and similar Doxygen syntax with @code{!} in it, i.e. replace
@code{//!} with @code{///} and similar. Also @@ syntax should be employed
for markup commands, i.e. use @code{@@param} and not @code{\param}.
@example
/**
* @@file
* MPEG codec.
* @@author ...
*/
/**
* Summary sentence.
* more text ...
* ...
*/
typedef struct Foobar @{
int var1; /**< var1 description */
int var2; ///< var2 description
/** var3 description */
int var3;
@} Foobar;
/**
* Summary sentence.
* more text ...
* ...
* @@param my_parameter description of my_parameter
* @@return return value description
*/
int myfunc(int my_parameter)
...
@end example
@section C language features
FFmpeg is programmed in the ISO C90 language with a few additional
features from ISO C99, namely:
@itemize @bullet
@item
the @samp{inline} keyword;
@item
@samp{//} comments;
@item
designated struct initializers (@samp{struct s x = @{ .i = 17 @};});
@item
compound literals (@samp{x = (struct s) @{ 17, 23 @};}).
@item
for loops with variable definition (@samp{for (int i = 0; i < 8; i++)});
@item
Variadic macros (@samp{#define ARRAY(nb, ...) (int[nb + 1])@{ nb, __VA_ARGS__ @}});
@item
Implementation defined behavior for signed integers is assumed to match the
expected behavior for two's complement. Non representable values in integer
casts are binary truncated. Shift right of signed values uses sign extension.
@end itemize
These features are supported by all compilers we care about, so we will not
accept patches to remove their use unless they absolutely do not impair
clarity and performance.
All code must compile with recent versions of GCC and a number of other
currently supported compilers. To ensure compatibility, please do not use
additional C99 features or GCC extensions. Especially watch out for:
@itemize @bullet
@item
mixing statements and declarations;
@item
@samp{long long} (use @samp{int64_t} instead);
@item
@samp{__attribute__} not protected by @samp{#ifdef __GNUC__} or similar;
@item
GCC statement expressions (@samp{(x = (@{ int y = 4; y; @})}).
@end itemize
@section Naming conventions
All names should be composed with underscores (_), not CamelCase. For example,
@samp{avfilter_get_video_buffer} is an acceptable function name and
@samp{AVFilterGetVideo} is not. The exception from this are type names, like
for example structs and enums; they should always be in CamelCase.
There are the following conventions for naming variables and functions:
@itemize @bullet
@item
For local variables no prefix is required.
@item
For file-scope variables and functions declared as @code{static}, no prefix
is required.
@item
For variables and functions visible outside of file scope, but only used
internally by a library, an @code{ff_} prefix should be used,
e.g. @samp{ff_w64_demuxer}.
@item
For variables and functions visible outside of file scope, used internally
across multiple libraries, use @code{avpriv_} as prefix, for example,
@samp{avpriv_report_missing_feature}.
@item
Each library has its own prefix for public symbols, in addition to the
commonly used @code{av_} (@code{avformat_} for libavformat,
@code{avcodec_} for libavcodec, @code{swr_} for libswresample, etc).
Check the existing code and choose names accordingly.
Note that some symbols without these prefixes are also exported for
retro-compatibility reasons. These exceptions are declared in the
@code{lib<name>/lib<name>.v} files.
@end itemize
Furthermore, name space reserved for the system should not be invaded.
Identifiers ending in @code{_t} are reserved by
@url{http://pubs.opengroup.org/onlinepubs/007904975/functions/xsh_chap02_02.html#tag_02_02_02, POSIX}.
Also avoid names starting with @code{__} or @code{_} followed by an uppercase
letter as they are reserved by the C standard. Names starting with @code{_}
are reserved at the file level and may not be used for externally visible
symbols. If in doubt, just avoid names starting with @code{_} altogether.
@section Miscellaneous conventions
@itemize @bullet
@item
fprintf and printf are forbidden in libavformat and libavcodec,
please use av_log() instead.
@item
Casts should be used only when necessary. Unneeded parentheses
should also be avoided if they don't make the code easier to understand.
@end itemize
@section Editor configuration
In order to configure Vim to follow FFmpeg formatting conventions, paste
the following snippet into your @file{.vimrc}:
@example
" indentation rules for FFmpeg: 4 spaces, no tabs
set expandtab
set shiftwidth=4
set softtabstop=4
set cindent
set cinoptions=(0
" Allow tabs in Makefiles.
autocmd FileType make,automake set noexpandtab shiftwidth=8 softtabstop=8
" Trailing whitespace and tabs are forbidden, so highlight them.
highlight ForbiddenWhitespace ctermbg=red guibg=red
match ForbiddenWhitespace /\s\+$\|\t/
" Do not highlight spaces at the end of line while typing on that line.
autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
@end example
For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
@lisp
(c-add-style "ffmpeg"
'("k&r"
(c-basic-offset . 4)
(indent-tabs-mode . nil)
(show-trailing-whitespace . t)
(c-offsets-alist
(statement-cont . (c-lineup-assignments +)))
)
)
(setq c-default-style "ffmpeg")
@end lisp
@chapter Development Policy
@section Patches/Committing
@subheading Licenses for patches must be compatible with FFmpeg.
Contributions should be licensed under the
@uref{http://www.gnu.org/licenses/lgpl-2.1.html, LGPL 2.1},
including an "or any later version" clause, or, if you prefer
a gift-style license, the
@uref{http://opensource.org/licenses/isc-license.txt, ISC} or
@uref{http://mit-license.org/, MIT} license.
@uref{http://www.gnu.org/licenses/gpl-2.0.html, GPL 2} including
an "or any later version" clause is also acceptable, but LGPL is
preferred.
If you add a new file, give it a proper license header. Do not copy and
paste it from a random place, use an existing file as template.
@subheading You must not commit code which breaks FFmpeg!
This means unfinished code which is enabled and breaks compilation,
or compiles but does not work/breaks the regression tests. Code which
is unfinished but disabled may be permitted under-circumstances, like
missing samples or an implementation with a small subset of features.
Always check the mailing list for any reviewers with issues and test
FATE before you push.
@subheading Keep the main commit message short with an extended description below.
The commit message should have a short first line in the form of
a @samp{topic: short description} as a header, separated by a newline
from the body consisting of an explanation of why the change is necessary.
If the commit fixes a known bug on the bug tracker, the commit message
should include its bug ID. Referring to the issue on the bug tracker does
not exempt you from writing an excerpt of the bug in the commit message.
@subheading Testing must be adequate but not excessive.
If it works for you, others, and passes FATE then it should be OK to commit
it, provided it fits the other committing criteria. You should not worry about
over-testing things. If your code has problems (portability, triggers
compiler bugs, unusual environment etc) they will be reported and eventually
fixed.
@subheading Do not commit unrelated changes together.
They should be split them into self-contained pieces. Also do not forget
that if part B depends on part A, but A does not depend on B, then A can
and should be committed first and separate from B. Keeping changes well
split into self-contained parts makes reviewing and understanding them on
the commit log mailing list easier. This also helps in case of debugging
later on.
Also if you have doubts about splitting or not splitting, do not hesitate to
ask/discuss it on the developer mailing list.
@subheading Ask before you change the build system (configure, etc).
Do not commit changes to the build system (Makefiles, configure script)
which change behavior, defaults etc, without asking first. The same
applies to compiler warning fixes, trivial looking fixes and to code
maintained by other developers. We usually have a reason for doing things
the way we do. Send your changes as patches to the ffmpeg-devel mailing
list, and if the code maintainers say OK, you may commit. This does not
apply to files you wrote and/or maintain.
@subheading Cosmetic changes should be kept in separate patches.
We refuse source indentation and other cosmetic changes if they are mixed
with functional changes, such commits will be rejected and removed. Every
developer has his own indentation style, you should not change it. Of course
if you (re)write something, you can use your own style, even though we would
prefer if the indentation throughout FFmpeg was consistent (Many projects
force a given indentation style - we do not.). If you really need to make
indentation changes (try to avoid this), separate them strictly from real
changes.
NOTE: If you had to put if()@{ .. @} over a large (> 5 lines) chunk of code,
then either do NOT change the indentation of the inner part within (do not
move it to the right)! or do so in a separate commit
@subheading Commit messages should always be filled out properly.
Always fill out the commit log message. Describe in a few lines what you
changed and why. You can refer to mailing list postings if you fix a
particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
Recommended format:
@example
area changed: Short 1 line description
details describing what and why and giving references.
@end example
@subheading Credit the author of the patch.
Make sure the author of the commit is set correctly. (see git commit --author)
If you apply a patch, send an
answer to ffmpeg-devel (or wherever you got the patch from) saying that
you applied the patch.
@subheading Complex patches should refer to discussion surrounding them.
When applying patches that have been discussed (at length) on the mailing
list, reference the thread in the log message.
@subheading Always wait long enough before pushing changes
Do NOT commit to code actively maintained by others without permission.
Send a patch to ffmpeg-devel. If no one answers within a reasonable
time-frame (12h for build failures and security fixes, 3 days small changes,
1 week for big patches) then commit your patch if you think it is OK.
Also note, the maintainer can simply ask for more time to review!
@section Code
@subheading API/ABI changes should be discussed before they are made.
Do not change behavior of the programs (renaming options etc) or public
API or ABI without first discussing it on the ffmpeg-devel mailing list.
Do not remove widely used functionality or features (redundant code can be removed).
@subheading Remember to check if you need to bump versions for libav*.
Depending on the change, you may need to change the version integer.
Incrementing the first component means no backward compatibility to
previous versions (e.g. removal of a function from the public API).
Incrementing the second component means backward compatible change
(e.g. addition of a function to the public API or extension of an
existing data structure).
Incrementing the third component means a noteworthy binary compatible
change (e.g. encoder bug fix that matters for the decoder). The third
component always starts at 100 to distinguish FFmpeg from Libav.
@subheading Warnings for correct code may be disabled if there is no other option.
Compiler warnings indicate potential bugs or code with bad style. If a type of
warning always points to correct and clean code, that warning should
be disabled, not the code changed.
Thus the remaining warnings can either be bugs or correct code.
If it is a bug, the bug has to be fixed. If it is not, the code should
be changed to not generate a warning unless that causes a slowdown
or obfuscates the code.
@subheading Check untrusted input properly.
Never write to unallocated memory, never write over the end of arrays,
always check values read from some untrusted source before using them
as array index or other risky things.
@section Documentation/Other
@subheading Subscribe to the ffmpeg-devel mailing list.
It is important to be subscribed to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Almost any non-trivial patch is to be sent there for review.
Other developers may have comments about your contribution. We expect you see
those comments, and to improve it if requested. (N.B. Experienced committers
have other channels, and may sometimes skip review for trivial fixes.) Also,
discussion here about bug fixes and FFmpeg improvements by other developers may
be helpful information for you. Finally, by being a list subscriber, your
contribution will be posted immediately to the list, without the moderation
hold which messages from non-subscribers experience.
However, it is more important to the project that we receive your patch than
that you be subscribed to the ffmpeg-devel list. If you have a patch, and don't
want to subscribe and discuss the patch, then please do send it to the list
anyway.
@subheading Subscribe to the ffmpeg-cvslog mailing list.
Diffs of all commits are sent to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-cvslog, ffmpeg-cvslog}
mailing list. Some developers read this list to review all code base changes
from all sources. Subscribing to this list is not mandatory.
@subheading Keep the documentation up to date.
Update the documentation if you change behavior or add features. If you are
unsure how best to do this, send a patch to ffmpeg-devel, the documentation
maintainer(s) will review and commit your stuff.
@subheading Important discussions should be accessible to all.
Try to keep important discussions and requests (also) on the public
developer mailing list, so that all developers can benefit from them.
@subheading Check your entries in MAINTAINERS.
Make sure that no parts of the codebase that you maintain are missing from the
@file{MAINTAINERS} file. If something that you want to maintain is missing add it with
your name after it.
If at some point you no longer want to maintain some code, then please help in
finding a new maintainer and also don't forget to update the @file{MAINTAINERS} file.
We think our rules are not too hard. If you have comments, contact us.
@chapter Code of conduct
Be friendly and respectful towards others and third parties.
Treat others the way you yourself want to be treated.
Be considerate. Not everyone shares the same viewpoint and priorities as you do.
Different opinions and interpretations help the project.
Looking at issues from a different perspective assists development.
Do not assume malice for things that can be attributed to incompetence. Even if
it is malice, it's rarely good to start with that as initial assumption.
Stay friendly even if someone acts contrarily. Everyone has a bad day
once in a while.
If you yourself have a bad day or are angry then try to take a break and reply
once you are calm and without anger if you have to.
Try to help other team members and cooperate if you can.
The goal of software development is to create technical excellence, not for any
individual to be better and "win" against the others. Large software projects
are only possible and successful through teamwork.
If someone struggles do not put them down. Give them a helping hand
instead and point them in the right direction.
Finally, keep in mind the immortal words of Bill and Ted,
"Be excellent to each other."
@anchor{Submitting patches}
@chapter Submitting patches
First, read the @ref{Coding Rules} above if you did not yet, in particular
the rules regarding patch submission.
When you submit your patch, please use @code{git format-patch} or
@code{git send-email}. We cannot read other diffs :-).
Also please do not submit a patch which contains several unrelated changes.
Split it into separate, self-contained pieces. This does not mean splitting
file by file. Instead, make the patch as small as possible while still
keeping it as a logical unit that contains an individual change, even
if it spans multiple files. This makes reviewing your patches much easier
for us and greatly increases your chances of getting your patch applied.
Use the patcheck tool of FFmpeg to check your patch.
The tool is located in the tools directory.
Run the @ref{Regression tests} before submitting a patch in order to verify
it does not cause unexpected problems.
It also helps quite a bit if you tell us what the patch does (for example
'replaces lrint by lrintf'), and why (for example '*BSD isn't C99 compliant
and has no lrint()')
Also please if you send several patches, send each patch as a separate mail,
do not attach several unrelated patches to the same mail.
Patches should be posted to the
@uref{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel, ffmpeg-devel}
mailing list. Use @code{git send-email} when possible since it will properly
send patches without requiring extra care. If you cannot, then send patches
as base64-encoded attachments, so your patch is not trashed during
transmission. Also ensure the correct mime type is used
(text/x-diff or text/x-patch or at least text/plain) and that only one
patch is inline or attached per mail.
You can check @url{https://patchwork.ffmpeg.org}, if your patch does not show up, its mime type
likely was wrong.
Your patch will be reviewed on the mailing list. You will likely be asked
to make some changes and are expected to send in an improved version that
incorporates the requests from the review. This process may go through
several iterations. Once your patch is deemed good enough, some developer
will pick it up and commit it to the official FFmpeg tree.
Give us a few days to react. But if some time passes without reaction,
send a reminder by email. Your patch should eventually be dealt with.
@chapter New codecs or formats checklist
@enumerate
@item
Did you use av_cold for codec initialization and close functions?
@item
Did you add a long_name under NULL_IF_CONFIG_SMALL to the AVCodec or
AVInputFormat/AVOutputFormat struct?
@item
Did you bump the minor version number (and reset the micro version
number) in @file{libavcodec/version.h} or @file{libavformat/version.h}?
@item
Did you register it in @file{allcodecs.c} or @file{allformats.c}?
@item
Did you add the AVCodecID to @file{avcodec.h}?
When adding new codec IDs, also add an entry to the codec descriptor
list in @file{libavcodec/codec_desc.c}.
@item
If it has a FourCC, did you add it to @file{libavformat/riff.c},
even if it is only a decoder?
@item
Did you add a rule to compile the appropriate files in the Makefile?
Remember to do this even if you're just adding a format to a file that is
already being compiled by some other rule, like a raw demuxer.
@item
Did you add an entry to the table of supported formats or codecs in
@file{doc/general.texi}?
@item
Did you add an entry in the Changelog?
@item
If it depends on a parser or a library, did you add that dependency in
configure?
@item
Did you @code{git add} the appropriate files before committing?
@item
Did you make sure it compiles standalone, i.e. with
@code{configure --disable-everything --enable-decoder=foo}
(or @code{--enable-demuxer} or whatever your component is)?
@end enumerate
@chapter Patch submission checklist
@enumerate
@item
Does @code{make fate} pass with the patch applied?
@item
Was the patch generated with git format-patch or send-email?
@item
Did you sign-off your patch? (@code{git commit -s})
See @uref{https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/plain/Documentation/process/submitting-patches.rst, Sign your work} for the meaning
of @dfn{sign-off}.
@item
Did you provide a clear git commit log message?
@item
Is the patch against latest FFmpeg git master branch?
@item
Are you subscribed to ffmpeg-devel?
(the list is subscribers only due to spam)
@item
Have you checked that the changes are minimal, so that the same cannot be
achieved with a smaller patch and/or simpler final code?
@item
If the change is to speed critical code, did you benchmark it?
@item
If you did any benchmarks, did you provide them in the mail?
@item
Have you checked that the patch does not introduce buffer overflows or
other security issues?
@item
Did you test your decoder or demuxer against damaged data? If no, see
tools/trasher, the noise bitstream filter, and
@uref{http://caca.zoy.org/wiki/zzuf, zzuf}. Your decoder or demuxer
should not crash, end in a (near) infinite loop, or allocate ridiculous
amounts of memory when fed damaged data.
@item
Did you test your decoder or demuxer against sample files?
Samples may be obtained at @url{https://samples.ffmpeg.org}.
@item
Does the patch not mix functional and cosmetic changes?
@item
Did you add tabs or trailing whitespace to the code? Both are forbidden.
@item
Is the patch attached to the email you send?
@item
Is the mime type of the patch correct? It should be text/x-diff or
text/x-patch or at least text/plain and not application/octet-stream.
@item
If the patch fixes a bug, did you provide a verbose analysis of the bug?
@item
If the patch fixes a bug, did you provide enough information, including
a sample, so the bug can be reproduced and the fix can be verified?
Note please do not attach samples >100k to mails but rather provide a
URL, you can upload to @url{https://streams.videolan.org/upload/}.
@item
Did you provide a verbose summary about what the patch does change?
@item
Did you provide a verbose explanation why it changes things like it does?
@item
Did you provide a verbose summary of the user visible advantages and
disadvantages if the patch is applied?
@item
Did you provide an example so we can verify the new feature added by the
patch easily?
@item
If you added a new file, did you insert a license header? It should be
taken from FFmpeg, not randomly copied and pasted from somewhere else.
@item
You should maintain alphabetical order in alphabetically ordered lists as
long as doing so does not break API/ABI compatibility.
@item
Lines with similar content should be aligned vertically when doing so
improves readability.
@item
Consider adding a regression test for your code.
@item
If you added YASM code please check that things still work with --disable-yasm.
@item
Make sure you check the return values of function and return appropriate
error codes. Especially memory allocation functions like @code{av_malloc()}
are notoriously left unchecked, which is a serious problem.
@item
Test your code with valgrind and or Address Sanitizer to ensure it's free
of leaks, out of array accesses, etc.
@end enumerate
@chapter Patch review process
All patches posted to ffmpeg-devel will be reviewed, unless they contain a
clear note that the patch is not for the git master branch.
Reviews and comments will be posted as replies to the patch on the
mailing list. The patch submitter then has to take care of every comment,
that can be by resubmitting a changed patch or by discussion. Resubmitted
patches will themselves be reviewed like any other patch. If at some point
a patch passes review with no comments then it is approved, that can for
simple and small patches happen immediately while large patches will generally
have to be changed and reviewed many times before they are approved.
After a patch is approved it will be committed to the repository.
We will review all submitted patches, but sometimes we are quite busy so
especially for large patches this can take several weeks.
If you feel that the review process is too slow and you are willing to try to
take over maintainership of the area of code you change then just clone
git master and maintain the area of code there. We will merge each area from
where its best maintained.
When resubmitting patches, please do not make any significant changes
not related to the comments received during review. Such patches will
be rejected. Instead, submit significant changes or new features as
separate patches.
Everyone is welcome to review patches. Also if you are waiting for your patch
to be reviewed, please consider helping to review other patches, that is a great
way to get everyone's patches reviewed sooner.
@anchor{Regression tests}
@chapter Regression tests
Before submitting a patch (or committing to the repository), you should at least
test that you did not break anything.
Running 'make fate' accomplishes this, please see @url{fate.html} for details.
[Of course, some patches may change the results of the regression tests. In
this case, the reference results of the regression tests shall be modified
accordingly].
@section Adding files to the fate-suite dataset
When there is no muxer or encoder available to generate test media for a
specific test then the media has to be included in the fate-suite.
First please make sure that the sample file is as small as possible to test the
respective decoder or demuxer sufficiently. Large files increase network
bandwidth and disk space requirements.
Once you have a working fate test and fate sample, provide in the commit
message or introductory message for the patch series that you post to
the ffmpeg-devel mailing list, a direct link to download the sample media.
@section Visualizing Test Coverage
The FFmpeg build system allows visualizing the test coverage in an easy
manner with the coverage tools @code{gcov}/@code{lcov}. This involves
the following steps:
@enumerate
@item
Configure to compile with instrumentation enabled:
@code{configure --toolchain=gcov}.
@item
Run your test case, either manually or via FATE. This can be either
the full FATE regression suite, or any arbitrary invocation of any
front-end tool provided by FFmpeg, in any combination.
@item
Run @code{make lcov} to generate coverage data in HTML format.
@item
View @code{lcov/index.html} in your preferred HTML viewer.
@end enumerate
You can use the command @code{make lcov-reset} to reset the coverage
measurements. You will need to rerun @code{make lcov} after running a
new test.
@section Using Valgrind
The configure script provides a shortcut for using valgrind to spot bugs
related to memory handling. Just add the option
@code{--toolchain=valgrind-memcheck} or @code{--toolchain=valgrind-massif}
to your configure line, and reasonable defaults will be set for running
FATE under the supervision of either the @strong{memcheck} or the
@strong{massif} tool of the valgrind suite.
In case you need finer control over how valgrind is invoked, use the
@code{--target-exec='valgrind <your_custom_valgrind_options>} option in
your configure line instead.
@anchor{Release process}
@chapter Release process
FFmpeg maintains a set of @strong{release branches}, which are the
recommended deliverable for system integrators and distributors (such as
Linux distributions, etc.). At regular times, a @strong{release
manager} prepares, tests and publishes tarballs on the
@url{https://ffmpeg.org} website.
There are two kinds of releases:
@enumerate
@item
@strong{Major releases} always include the latest and greatest
features and functionality.
@item
@strong{Point releases} are cut from @strong{release} branches,
which are named @code{release/X}, with @code{X} being the release
version number.
@end enumerate
Note that we promise to our users that shared libraries from any FFmpeg
release never break programs that have been @strong{compiled} against
previous versions of @strong{the same release series} in any case!
However, from time to time, we do make API changes that require adaptations
in applications. Such changes are only allowed in (new) major releases and
require further steps such as bumping library version numbers and/or
adjustments to the symbol versioning file. Please discuss such changes
on the @strong{ffmpeg-devel} mailing list in time to allow forward planning.
@anchor{Criteria for Point Releases}
@section Criteria for Point Releases
Changes that match the following criteria are valid candidates for
inclusion into a point release:
@enumerate
@item
Fixes a security issue, preferably identified by a @strong{CVE
number} issued by @url{http://cve.mitre.org/}.
@item
Fixes a documented bug in @url{https://trac.ffmpeg.org}.
@item
Improves the included documentation.
@item
Retains both source code and binary compatibility with previous
point releases of the same release branch.
@end enumerate
The order for checking the rules is (1 OR 2 OR 3) AND 4.
@section Release Checklist
The release process involves the following steps:
@enumerate
@item
Ensure that the @file{RELEASE} file contains the version number for
the upcoming release.
@item
Add the release at @url{https://trac.ffmpeg.org/admin/ticket/versions}.
@item
Announce the intent to do a release to the mailing list.
@item
Make sure all relevant security fixes have been backported. See
@url{https://ffmpeg.org/security.html}.
@item
Ensure that the FATE regression suite still passes in the release
branch on at least @strong{i386} and @strong{amd64}
(cf. @ref{Regression tests}).
@item
Prepare the release tarballs in @code{bz2} and @code{gz} formats, and
supplementing files that contain @code{gpg} signatures
@item
Publish the tarballs at @url{https://ffmpeg.org/releases}. Create and
push an annotated tag in the form @code{nX}, with @code{X}
containing the version number.
@item
Propose and send a patch to the @strong{ffmpeg-devel} mailing list
with a news entry for the website.
@item
Publish the news entry.
@item
Send an announcement to the mailing list.
@end enumerate
@bye

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@chapter Device Options
@c man begin DEVICE OPTIONS
The libavdevice library provides the same interface as
libavformat. Namely, an input device is considered like a demuxer, and
an output device like a muxer, and the interface and generic device
options are the same provided by libavformat (see the ffmpeg-formats
manual).
In addition each input or output device may support so-called private
options, which are specific for that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the device
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
@c man end DEVICE OPTIONS
@ifclear config-writeonly
@include indevs.texi
@end ifclear
@ifclear config-readonly
@include outdevs.texi
@end ifclear

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#!/bin/sh
OUT_DIR="${1}"
DOXYFILE="${2}"
DOXYGEN="${3}"
shift 3
if [ -e "VERSION" ]; then
VERSION=`cat "VERSION"`
else
VERSION=`git describe`
fi
$DOXYGEN - <<EOF
@INCLUDE = ${DOXYFILE}
INPUT = $@
HTML_TIMESTAMP = NO
PROJECT_NUMBER = $VERSION
OUTPUT_DIRECTORY = $OUT_DIR
EOF

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/html/

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

24
externals/ffmpeg/doc/examples/.gitignore vendored Executable file
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/avio_list_dir
/avio_reading
/decode_audio
/decode_video
/demuxing_decoding
/encode_audio
/encode_video
/extract_mvs
/filter_audio
/filtering_audio
/filtering_video
/http_multiclient
/hw_decode
/metadata
/muxing
/pc-uninstalled
/qsvdec
/remuxing
/resampling_audio
/scaling_video
/transcode_aac
/transcoding
/vaapi_encode
/vaapi_transcode

64
externals/ffmpeg/doc/examples/Makefile vendored Executable file
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EXAMPLES-$(CONFIG_AVIO_LIST_DIR_EXAMPLE) += avio_list_dir
EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE) += avio_reading
EXAMPLES-$(CONFIG_DECODE_AUDIO_EXAMPLE) += decode_audio
EXAMPLES-$(CONFIG_DECODE_VIDEO_EXAMPLE) += decode_video
EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
EXAMPLES-$(CONFIG_ENCODE_AUDIO_EXAMPLE) += encode_audio
EXAMPLES-$(CONFIG_ENCODE_VIDEO_EXAMPLE) += encode_video
EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE) += extract_mvs
EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE) += filter_audio
EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE) += filtering_audio
EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE) += filtering_video
EXAMPLES-$(CONFIG_HTTP_MULTICLIENT_EXAMPLE) += http_multiclient
EXAMPLES-$(CONFIG_HW_DECODE_EXAMPLE) += hw_decode
EXAMPLES-$(CONFIG_METADATA_EXAMPLE) += metadata
EXAMPLES-$(CONFIG_MUXING_EXAMPLE) += muxing
EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE) += qsvdec
EXAMPLES-$(CONFIG_REMUXING_EXAMPLE) += remuxing
EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE) += resampling_audio
EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE) += scaling_video
EXAMPLES-$(CONFIG_TRANSCODE_AAC_EXAMPLE) += transcode_aac
EXAMPLES-$(CONFIG_TRANSCODING_EXAMPLE) += transcoding
EXAMPLES-$(CONFIG_VAAPI_ENCODE_EXAMPLE) += vaapi_encode
EXAMPLES-$(CONFIG_VAAPI_TRANSCODE_EXAMPLE) += vaapi_transcode
EXAMPLES := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)$(EXESUF))
EXAMPLES_G := $(EXAMPLES-yes:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
ALL_EXAMPLES := $(EXAMPLES) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)$(EXESUF))
ALL_EXAMPLES_G := $(EXAMPLES_G) $(EXAMPLES-:%=doc/examples/%$(PROGSSUF)_g$(EXESUF))
PROGS += $(EXAMPLES)
EXAMPLE_MAKEFILE := $(SRC_PATH)/doc/examples/Makefile
EXAMPLES_FILES := $(wildcard $(SRC_PATH)/doc/examples/*.c) $(SRC_PATH)/doc/examples/README $(EXAMPLE_MAKEFILE)
$(foreach P,$(EXAMPLES),$(eval OBJS-$(P:%$(PROGSSUF)$(EXESUF)=%) = $(P:%$(PROGSSUF)$(EXESUF)=%).o))
$(EXAMPLES_G): %$(PROGSSUF)_g$(EXESUF): %.o
examples: $(EXAMPLES)
$(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.o): | doc/examples
OUTDIRS += doc/examples
DOXY_INPUT += $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.c)
install: install-examples
install-examples: $(EXAMPLES_FILES)
$(Q)mkdir -p "$(DATADIR)/examples"
$(INSTALL) -m 644 $(EXAMPLES_FILES) "$(DATADIR)/examples"
$(INSTALL) -m 644 $(EXAMPLE_MAKEFILE:%=%.example) "$(DATADIR)/examples/Makefile"
uninstall: uninstall-examples
uninstall-examples:
$(RM) -r "$(DATADIR)/examples"
examplesclean:
$(RM) $(ALL_EXAMPLES) $(ALL_EXAMPLES_G)
$(RM) $(CLEANSUFFIXES:%=doc/examples/%)
docclean:: examplesclean
-include $(wildcard $(EXAMPLES:%$(PROGSSUF)$(EXESUF)=%.d))
.PHONY: examples

50
externals/ffmpeg/doc/examples/Makefile.example vendored Executable file
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@@ -0,0 +1,50 @@
# use pkg-config for getting CFLAGS and LDLIBS
FFMPEG_LIBS= libavdevice \
libavformat \
libavfilter \
libavcodec \
libswresample \
libswscale \
libavutil \
CFLAGS += -Wall -g
CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
EXAMPLES= avio_list_dir \
avio_reading \
decode_audio \
decode_video \
demuxing_decoding \
encode_audio \
encode_video \
extract_mvs \
filtering_video \
filtering_audio \
http_multiclient \
hw_decode \
metadata \
muxing \
remuxing \
resampling_audio \
scaling_video \
transcode_aac \
transcoding \
OBJS=$(addsuffix .o,$(EXAMPLES))
# the following examples make explicit use of the math library
avcodec: LDLIBS += -lm
encode_audio: LDLIBS += -lm
muxing: LDLIBS += -lm
resampling_audio: LDLIBS += -lm
.phony: all clean-test clean
all: $(OBJS) $(EXAMPLES)
clean-test:
$(RM) test*.pgm test.h264 test.mp2 test.sw test.mpg
clean: clean-test
$(RM) $(EXAMPLES) $(OBJS)

23
externals/ffmpeg/doc/examples/README vendored Executable file
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@@ -0,0 +1,23 @@
FFmpeg examples README
----------------------
Both following use cases rely on pkg-config and make, thus make sure
that you have them installed and working on your system.
Method 1: build the installed examples in a generic read/write user directory
Copy to a read/write user directory and just use "make", it will link
to the libraries on your system, assuming the PKG_CONFIG_PATH is
correctly configured.
Method 2: build the examples in-tree
Assuming you are in the source FFmpeg checkout directory, you need to build
FFmpeg (no need to make install in any prefix). Then just run "make examples".
This will build the examples using the FFmpeg build system. You can clean those
examples using "make examplesclean"
If you want to try the dedicated Makefile examples (to emulate the first
method), go into doc/examples and run a command such as
PKG_CONFIG_PATH=pc-uninstalled make.

130
externals/ffmpeg/doc/examples/avio_list_dir.c vendored Executable file
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/*
* Copyright (c) 2014 Lukasz Marek
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
static const char *type_string(int type)
{
switch (type) {
case AVIO_ENTRY_DIRECTORY:
return "<DIR>";
case AVIO_ENTRY_FILE:
return "<FILE>";
case AVIO_ENTRY_BLOCK_DEVICE:
return "<BLOCK DEVICE>";
case AVIO_ENTRY_CHARACTER_DEVICE:
return "<CHARACTER DEVICE>";
case AVIO_ENTRY_NAMED_PIPE:
return "<PIPE>";
case AVIO_ENTRY_SYMBOLIC_LINK:
return "<LINK>";
case AVIO_ENTRY_SOCKET:
return "<SOCKET>";
case AVIO_ENTRY_SERVER:
return "<SERVER>";
case AVIO_ENTRY_SHARE:
return "<SHARE>";
case AVIO_ENTRY_WORKGROUP:
return "<WORKGROUP>";
case AVIO_ENTRY_UNKNOWN:
default:
break;
}
return "<UNKNOWN>";
}
static int list_op(const char *input_dir)
{
AVIODirEntry *entry = NULL;
AVIODirContext *ctx = NULL;
int cnt, ret;
char filemode[4], uid_and_gid[20];
if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
goto fail;
}
cnt = 0;
for (;;) {
if ((ret = avio_read_dir(ctx, &entry)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
goto fail;
}
if (!entry)
break;
if (entry->filemode == -1) {
snprintf(filemode, 4, "???");
} else {
snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
}
snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
if (cnt == 0)
av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
"TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
"ACCESSED", "STATUS_CHANGED");
av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
type_string(entry->type),
entry->size,
entry->name,
uid_and_gid,
filemode,
entry->modification_timestamp,
entry->access_timestamp,
entry->status_change_timestamp);
avio_free_directory_entry(&entry);
cnt++;
};
fail:
avio_close_dir(&ctx);
return ret;
}
static void usage(const char *program_name)
{
fprintf(stderr, "usage: %s input_dir\n"
"API example program to show how to list files in directory "
"accessed through AVIOContext.\n", program_name);
}
int main(int argc, char *argv[])
{
int ret;
av_log_set_level(AV_LOG_DEBUG);
if (argc < 2) {
usage(argv[0]);
return 1;
}
avformat_network_init();
ret = list_op(argv[1]);
avformat_network_deinit();
return ret < 0 ? 1 : 0;
}

134
externals/ffmpeg/doc/examples/avio_reading.c vendored Executable file
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@@ -0,0 +1,134 @@
/*
* Copyright (c) 2014 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat AVIOContext API example.
*
* Make libavformat demuxer access media content through a custom
* AVIOContext read callback.
* @example avio_reading.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavformat/avio.h>
#include <libavutil/file.h>
struct buffer_data {
uint8_t *ptr;
size_t size; ///< size left in the buffer
};
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
{
struct buffer_data *bd = (struct buffer_data *)opaque;
buf_size = FFMIN(buf_size, bd->size);
if (!buf_size)
return AVERROR_EOF;
printf("ptr:%p size:%zu\n", bd->ptr, bd->size);
/* copy internal buffer data to buf */
memcpy(buf, bd->ptr, buf_size);
bd->ptr += buf_size;
bd->size -= buf_size;
return buf_size;
}
int main(int argc, char *argv[])
{
AVFormatContext *fmt_ctx = NULL;
AVIOContext *avio_ctx = NULL;
uint8_t *buffer = NULL, *avio_ctx_buffer = NULL;
size_t buffer_size, avio_ctx_buffer_size = 4096;
char *input_filename = NULL;
int ret = 0;
struct buffer_data bd = { 0 };
if (argc != 2) {
fprintf(stderr, "usage: %s input_file\n"
"API example program to show how to read from a custom buffer "
"accessed through AVIOContext.\n", argv[0]);
return 1;
}
input_filename = argv[1];
/* slurp file content into buffer */
ret = av_file_map(input_filename, &buffer, &buffer_size, 0, NULL);
if (ret < 0)
goto end;
/* fill opaque structure used by the AVIOContext read callback */
bd.ptr = buffer;
bd.size = buffer_size;
if (!(fmt_ctx = avformat_alloc_context())) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx_buffer = av_malloc(avio_ctx_buffer_size);
if (!avio_ctx_buffer) {
ret = AVERROR(ENOMEM);
goto end;
}
avio_ctx = avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size,
0, &bd, &read_packet, NULL, NULL);
if (!avio_ctx) {
ret = AVERROR(ENOMEM);
goto end;
}
fmt_ctx->pb = avio_ctx;
ret = avformat_open_input(&fmt_ctx, NULL, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open input\n");
goto end;
}
ret = avformat_find_stream_info(fmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Could not find stream information\n");
goto end;
}
av_dump_format(fmt_ctx, 0, input_filename, 0);
end:
avformat_close_input(&fmt_ctx);
/* note: the internal buffer could have changed, and be != avio_ctx_buffer */
if (avio_ctx)
av_freep(&avio_ctx->buffer);
avio_context_free(&avio_ctx);
av_file_unmap(buffer, buffer_size);
if (ret < 0) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

236
externals/ffmpeg/doc/examples/decode_audio.c vendored Executable file
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@@ -0,0 +1,236 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio decoding with libavcodec API example
*
* @example decode_audio.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavutil/frame.h>
#include <libavutil/mem.h>
#include <libavcodec/avcodec.h>
#define AUDIO_INBUF_SIZE 20480
#define AUDIO_REFILL_THRESH 4096
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame,
FILE *outfile)
{
int i, ch;
int ret, data_size;
/* send the packet with the compressed data to the decoder */
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting the packet to the decoder\n");
exit(1);
}
/* read all the output frames (in general there may be any number of them */
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
data_size = av_get_bytes_per_sample(dec_ctx->sample_fmt);
if (data_size < 0) {
/* This should not occur, checking just for paranoia */
fprintf(stderr, "Failed to calculate data size\n");
exit(1);
}
for (i = 0; i < frame->nb_samples; i++)
for (ch = 0; ch < dec_ctx->channels; ch++)
fwrite(frame->data[ch] + data_size*i, 1, data_size, outfile);
}
}
int main(int argc, char **argv)
{
const char *outfilename, *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVCodecParserContext *parser = NULL;
int len, ret;
FILE *f, *outfile;
uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
AVPacket *pkt;
AVFrame *decoded_frame = NULL;
enum AVSampleFormat sfmt;
int n_channels = 0;
const char *fmt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "Parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
outfile = fopen(outfilename, "wb");
if (!outfile) {
av_free(c);
exit(1);
}
/* decode until eof */
data = inbuf;
data_size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
while (data_size > 0) {
if (!decoded_frame) {
if (!(decoded_frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
}
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size,
AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, pkt, decoded_frame, outfile);
if (data_size < AUDIO_REFILL_THRESH) {
memmove(inbuf, data, data_size);
data = inbuf;
len = fread(data + data_size, 1,
AUDIO_INBUF_SIZE - data_size, f);
if (len > 0)
data_size += len;
}
}
/* flush the decoder */
pkt->data = NULL;
pkt->size = 0;
decode(c, pkt, decoded_frame, outfile);
/* print output pcm infomations, because there have no metadata of pcm */
sfmt = c->sample_fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
}
n_channels = c->channels;
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, c->sample_rate,
outfilename);
end:
fclose(outfile);
fclose(f);
avcodec_free_context(&c);
av_parser_close(parser);
av_frame_free(&decoded_frame);
av_packet_free(&pkt);
return 0;
}

187
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@@ -0,0 +1,187 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video decoding with libavcodec API example
*
* @example decode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#define INBUF_SIZE 4096
static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
char *filename)
{
FILE *f;
int i;
f = fopen(filename,"w");
fprintf(f, "P5\n%d %d\n%d\n", xsize, ysize, 255);
for (i = 0; i < ysize; i++)
fwrite(buf + i * wrap, 1, xsize, f);
fclose(f);
}
static void decode(AVCodecContext *dec_ctx, AVFrame *frame, AVPacket *pkt,
const char *filename)
{
char buf[1024];
int ret;
ret = avcodec_send_packet(dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error sending a packet for decoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
exit(1);
}
printf("saving frame %3d\n", dec_ctx->frame_number);
fflush(stdout);
/* the picture is allocated by the decoder. no need to
free it */
snprintf(buf, sizeof(buf), "%s-%d", filename, dec_ctx->frame_number);
pgm_save(frame->data[0], frame->linesize[0],
frame->width, frame->height, buf);
}
}
int main(int argc, char **argv)
{
const char *filename, *outfilename;
const AVCodec *codec;
AVCodecParserContext *parser;
AVCodecContext *c= NULL;
FILE *f;
AVFrame *frame;
uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
uint8_t *data;
size_t data_size;
int ret;
AVPacket *pkt;
if (argc <= 2) {
fprintf(stderr, "Usage: %s <input file> <output file>\n"
"And check your input file is encoded by mpeg1video please.\n", argv[0]);
exit(0);
}
filename = argv[1];
outfilename = argv[2];
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
/* find the MPEG-1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
parser = av_parser_init(codec->id);
if (!parser) {
fprintf(stderr, "parser not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
/* For some codecs, such as msmpeg4 and mpeg4, width and height
MUST be initialized there because this information is not
available in the bitstream. */
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "rb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
while (!feof(f)) {
/* read raw data from the input file */
data_size = fread(inbuf, 1, INBUF_SIZE, f);
if (!data_size)
break;
/* use the parser to split the data into frames */
data = inbuf;
while (data_size > 0) {
ret = av_parser_parse2(parser, c, &pkt->data, &pkt->size,
data, data_size, AV_NOPTS_VALUE, AV_NOPTS_VALUE, 0);
if (ret < 0) {
fprintf(stderr, "Error while parsing\n");
exit(1);
}
data += ret;
data_size -= ret;
if (pkt->size)
decode(c, frame, pkt, outfilename);
}
}
/* flush the decoder */
decode(c, frame, NULL, outfilename);
fclose(f);
av_parser_close(parser);
avcodec_free_context(&c);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

379
externals/ffmpeg/doc/examples/demuxing_decoding.c vendored Executable file
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@@ -0,0 +1,379 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Demuxing and decoding example.
*
* Show how to use the libavformat and libavcodec API to demux and
* decode audio and video data.
* @example demuxing_decoding.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/samplefmt.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
static int width, height;
static enum AVPixelFormat pix_fmt;
static AVStream *video_stream = NULL, *audio_stream = NULL;
static const char *src_filename = NULL;
static const char *video_dst_filename = NULL;
static const char *audio_dst_filename = NULL;
static FILE *video_dst_file = NULL;
static FILE *audio_dst_file = NULL;
static uint8_t *video_dst_data[4] = {NULL};
static int video_dst_linesize[4];
static int video_dst_bufsize;
static int video_stream_idx = -1, audio_stream_idx = -1;
static AVFrame *frame = NULL;
static AVPacket pkt;
static int video_frame_count = 0;
static int audio_frame_count = 0;
static int output_video_frame(AVFrame *frame)
{
if (frame->width != width || frame->height != height ||
frame->format != pix_fmt) {
/* To handle this change, one could call av_image_alloc again and
* decode the following frames into another rawvideo file. */
fprintf(stderr, "Error: Width, height and pixel format have to be "
"constant in a rawvideo file, but the width, height or "
"pixel format of the input video changed:\n"
"old: width = %d, height = %d, format = %s\n"
"new: width = %d, height = %d, format = %s\n",
width, height, av_get_pix_fmt_name(pix_fmt),
frame->width, frame->height,
av_get_pix_fmt_name(frame->format));
return -1;
}
printf("video_frame n:%d coded_n:%d\n",
video_frame_count++, frame->coded_picture_number);
/* copy decoded frame to destination buffer:
* this is required since rawvideo expects non aligned data */
av_image_copy(video_dst_data, video_dst_linesize,
(const uint8_t **)(frame->data), frame->linesize,
pix_fmt, width, height);
/* write to rawvideo file */
fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
return 0;
}
static int output_audio_frame(AVFrame *frame)
{
size_t unpadded_linesize = frame->nb_samples * av_get_bytes_per_sample(frame->format);
printf("audio_frame n:%d nb_samples:%d pts:%s\n",
audio_frame_count++, frame->nb_samples,
av_ts2timestr(frame->pts, &audio_dec_ctx->time_base));
/* Write the raw audio data samples of the first plane. This works
* fine for packed formats (e.g. AV_SAMPLE_FMT_S16). However,
* most audio decoders output planar audio, which uses a separate
* plane of audio samples for each channel (e.g. AV_SAMPLE_FMT_S16P).
* In other words, this code will write only the first audio channel
* in these cases.
* You should use libswresample or libavfilter to convert the frame
* to packed data. */
fwrite(frame->extended_data[0], 1, unpadded_linesize, audio_dst_file);
return 0;
}
static int decode_packet(AVCodecContext *dec, const AVPacket *pkt)
{
int ret = 0;
// submit the packet to the decoder
ret = avcodec_send_packet(dec, pkt);
if (ret < 0) {
fprintf(stderr, "Error submitting a packet for decoding (%s)\n", av_err2str(ret));
return ret;
}
// get all the available frames from the decoder
while (ret >= 0) {
ret = avcodec_receive_frame(dec, frame);
if (ret < 0) {
// those two return values are special and mean there is no output
// frame available, but there were no errors during decoding
if (ret == AVERROR_EOF || ret == AVERROR(EAGAIN))
return 0;
fprintf(stderr, "Error during decoding (%s)\n", av_err2str(ret));
return ret;
}
// write the frame data to output file
if (dec->codec->type == AVMEDIA_TYPE_VIDEO)
ret = output_video_frame(frame);
else
ret = output_audio_frame(frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
static int open_codec_context(int *stream_idx,
AVCodecContext **dec_ctx, AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret, stream_index;
AVStream *st;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
stream_index = ret;
st = fmt_ctx->streams[stream_index];
/* find decoder for the stream */
dec = avcodec_find_decoder(st->codecpar->codec_id);
if (!dec) {
fprintf(stderr, "Failed to find %s codec\n",
av_get_media_type_string(type));
return AVERROR(EINVAL);
}
/* Allocate a codec context for the decoder */
*dec_ctx = avcodec_alloc_context3(dec);
if (!*dec_ctx) {
fprintf(stderr, "Failed to allocate the %s codec context\n",
av_get_media_type_string(type));
return AVERROR(ENOMEM);
}
/* Copy codec parameters from input stream to output codec context */
if ((ret = avcodec_parameters_to_context(*dec_ctx, st->codecpar)) < 0) {
fprintf(stderr, "Failed to copy %s codec parameters to decoder context\n",
av_get_media_type_string(type));
return ret;
}
/* Init the decoders */
if ((ret = avcodec_open2(*dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
*stream_idx = stream_index;
}
return 0;
}
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"sample format %s is not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return -1;
}
int main (int argc, char **argv)
{
int ret = 0;
if (argc != 4) {
fprintf(stderr, "usage: %s input_file video_output_file audio_output_file\n"
"API example program to show how to read frames from an input file.\n"
"This program reads frames from a file, decodes them, and writes decoded\n"
"video frames to a rawvideo file named video_output_file, and decoded\n"
"audio frames to a rawaudio file named audio_output_file.\n",
argv[0]);
exit(1);
}
src_filename = argv[1];
video_dst_filename = argv[2];
audio_dst_filename = argv[3];
/* open input file, and allocate format context */
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
/* retrieve stream information */
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
if (open_codec_context(&video_stream_idx, &video_dec_ctx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
video_stream = fmt_ctx->streams[video_stream_idx];
video_dst_file = fopen(video_dst_filename, "wb");
if (!video_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", video_dst_filename);
ret = 1;
goto end;
}
/* allocate image where the decoded image will be put */
width = video_dec_ctx->width;
height = video_dec_ctx->height;
pix_fmt = video_dec_ctx->pix_fmt;
ret = av_image_alloc(video_dst_data, video_dst_linesize,
width, height, pix_fmt, 1);
if (ret < 0) {
fprintf(stderr, "Could not allocate raw video buffer\n");
goto end;
}
video_dst_bufsize = ret;
}
if (open_codec_context(&audio_stream_idx, &audio_dec_ctx, fmt_ctx, AVMEDIA_TYPE_AUDIO) >= 0) {
audio_stream = fmt_ctx->streams[audio_stream_idx];
audio_dst_file = fopen(audio_dst_filename, "wb");
if (!audio_dst_file) {
fprintf(stderr, "Could not open destination file %s\n", audio_dst_filename);
ret = 1;
goto end;
}
}
/* dump input information to stderr */
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!audio_stream && !video_stream) {
fprintf(stderr, "Could not find audio or video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* initialize packet, set data to NULL, let the demuxer fill it */
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (video_stream)
printf("Demuxing video from file '%s' into '%s'\n", src_filename, video_dst_filename);
if (audio_stream)
printf("Demuxing audio from file '%s' into '%s'\n", src_filename, audio_dst_filename);
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
// check if the packet belongs to a stream we are interested in, otherwise
// skip it
if (pkt.stream_index == video_stream_idx)
ret = decode_packet(video_dec_ctx, &pkt);
else if (pkt.stream_index == audio_stream_idx)
ret = decode_packet(audio_dec_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0)
break;
}
/* flush the decoders */
if (video_dec_ctx)
decode_packet(video_dec_ctx, NULL);
if (audio_dec_ctx)
decode_packet(audio_dec_ctx, NULL);
printf("Demuxing succeeded.\n");
if (video_stream) {
printf("Play the output video file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(pix_fmt), width, height,
video_dst_filename);
}
if (audio_stream) {
enum AVSampleFormat sfmt = audio_dec_ctx->sample_fmt;
int n_channels = audio_dec_ctx->channels;
const char *fmt;
if (av_sample_fmt_is_planar(sfmt)) {
const char *packed = av_get_sample_fmt_name(sfmt);
printf("Warning: the sample format the decoder produced is planar "
"(%s). This example will output the first channel only.\n",
packed ? packed : "?");
sfmt = av_get_packed_sample_fmt(sfmt);
n_channels = 1;
}
if ((ret = get_format_from_sample_fmt(&fmt, sfmt)) < 0)
goto end;
printf("Play the output audio file with the command:\n"
"ffplay -f %s -ac %d -ar %d %s\n",
fmt, n_channels, audio_dec_ctx->sample_rate,
audio_dst_filename);
}
end:
avcodec_free_context(&video_dec_ctx);
avcodec_free_context(&audio_dec_ctx);
avformat_close_input(&fmt_ctx);
if (video_dst_file)
fclose(video_dst_file);
if (audio_dst_file)
fclose(audio_dst_file);
av_frame_free(&frame);
av_free(video_dst_data[0]);
return ret < 0;
}

238
externals/ffmpeg/doc/examples/encode_audio.c vendored Executable file
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@@ -0,0 +1,238 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* audio encoding with libavcodec API example.
*
* @example encode_audio.c
*/
#include <stdint.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavutil/channel_layout.h>
#include <libavutil/common.h>
#include <libavutil/frame.h>
#include <libavutil/samplefmt.h>
/* check that a given sample format is supported by the encoder */
static int check_sample_fmt(const AVCodec *codec, enum AVSampleFormat sample_fmt)
{
const enum AVSampleFormat *p = codec->sample_fmts;
while (*p != AV_SAMPLE_FMT_NONE) {
if (*p == sample_fmt)
return 1;
p++;
}
return 0;
}
/* just pick the highest supported samplerate */
static int select_sample_rate(const AVCodec *codec)
{
const int *p;
int best_samplerate = 0;
if (!codec->supported_samplerates)
return 44100;
p = codec->supported_samplerates;
while (*p) {
if (!best_samplerate || abs(44100 - *p) < abs(44100 - best_samplerate))
best_samplerate = *p;
p++;
}
return best_samplerate;
}
/* select layout with the highest channel count */
static int select_channel_layout(const AVCodec *codec)
{
const uint64_t *p;
uint64_t best_ch_layout = 0;
int best_nb_channels = 0;
if (!codec->channel_layouts)
return AV_CH_LAYOUT_STEREO;
p = codec->channel_layouts;
while (*p) {
int nb_channels = av_get_channel_layout_nb_channels(*p);
if (nb_channels > best_nb_channels) {
best_ch_layout = *p;
best_nb_channels = nb_channels;
}
p++;
}
return best_ch_layout;
}
static void encode(AVCodecContext *ctx, AVFrame *frame, AVPacket *pkt,
FILE *output)
{
int ret;
/* send the frame for encoding */
ret = avcodec_send_frame(ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending the frame to the encoder\n");
exit(1);
}
/* read all the available output packets (in general there may be any
* number of them */
while (ret >= 0) {
ret = avcodec_receive_packet(ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error encoding audio frame\n");
exit(1);
}
fwrite(pkt->data, 1, pkt->size, output);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename;
const AVCodec *codec;
AVCodecContext *c= NULL;
AVFrame *frame;
AVPacket *pkt;
int i, j, k, ret;
FILE *f;
uint16_t *samples;
float t, tincr;
if (argc <= 1) {
fprintf(stderr, "Usage: %s <output file>\n", argv[0]);
return 0;
}
filename = argv[1];
/* find the MP2 encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "Codec not found\n");
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate audio codec context\n");
exit(1);
}
/* put sample parameters */
c->bit_rate = 64000;
/* check that the encoder supports s16 pcm input */
c->sample_fmt = AV_SAMPLE_FMT_S16;
if (!check_sample_fmt(codec, c->sample_fmt)) {
fprintf(stderr, "Encoder does not support sample format %s",
av_get_sample_fmt_name(c->sample_fmt));
exit(1);
}
/* select other audio parameters supported by the encoder */
c->sample_rate = select_sample_rate(codec);
c->channel_layout = select_channel_layout(codec);
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
/* open it */
if (avcodec_open2(c, codec, NULL) < 0) {
fprintf(stderr, "Could not open codec\n");
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
/* packet for holding encoded output */
pkt = av_packet_alloc();
if (!pkt) {
fprintf(stderr, "could not allocate the packet\n");
exit(1);
}
/* frame containing input raw audio */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate audio frame\n");
exit(1);
}
frame->nb_samples = c->frame_size;
frame->format = c->sample_fmt;
frame->channel_layout = c->channel_layout;
/* allocate the data buffers */
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate audio data buffers\n");
exit(1);
}
/* encode a single tone sound */
t = 0;
tincr = 2 * M_PI * 440.0 / c->sample_rate;
for (i = 0; i < 200; i++) {
/* make sure the frame is writable -- makes a copy if the encoder
* kept a reference internally */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
samples = (uint16_t*)frame->data[0];
for (j = 0; j < c->frame_size; j++) {
samples[2*j] = (int)(sin(t) * 10000);
for (k = 1; k < c->channels; k++)
samples[2*j + k] = samples[2*j];
t += tincr;
}
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
fclose(f);
av_frame_free(&frame);
av_packet_free(&pkt);
avcodec_free_context(&c);
return 0;
}

198
externals/ffmpeg/doc/examples/encode_video.c vendored Executable file
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@@ -0,0 +1,198 @@
/*
* Copyright (c) 2001 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* video encoding with libavcodec API example
*
* @example encode_video.c
*/
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <libavcodec/avcodec.h>
#include <libavutil/opt.h>
#include <libavutil/imgutils.h>
static void encode(AVCodecContext *enc_ctx, AVFrame *frame, AVPacket *pkt,
FILE *outfile)
{
int ret;
/* send the frame to the encoder */
if (frame)
printf("Send frame %3"PRId64"\n", frame->pts);
ret = avcodec_send_frame(enc_ctx, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame for encoding\n");
exit(1);
}
while (ret >= 0) {
ret = avcodec_receive_packet(enc_ctx, pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return;
else if (ret < 0) {
fprintf(stderr, "Error during encoding\n");
exit(1);
}
printf("Write packet %3"PRId64" (size=%5d)\n", pkt->pts, pkt->size);
fwrite(pkt->data, 1, pkt->size, outfile);
av_packet_unref(pkt);
}
}
int main(int argc, char **argv)
{
const char *filename, *codec_name;
const AVCodec *codec;
AVCodecContext *c= NULL;
int i, ret, x, y;
FILE *f;
AVFrame *frame;
AVPacket *pkt;
uint8_t endcode[] = { 0, 0, 1, 0xb7 };
if (argc <= 2) {
fprintf(stderr, "Usage: %s <output file> <codec name>\n", argv[0]);
exit(0);
}
filename = argv[1];
codec_name = argv[2];
/* find the mpeg1video encoder */
codec = avcodec_find_encoder_by_name(codec_name);
if (!codec) {
fprintf(stderr, "Codec '%s' not found\n", codec_name);
exit(1);
}
c = avcodec_alloc_context3(codec);
if (!c) {
fprintf(stderr, "Could not allocate video codec context\n");
exit(1);
}
pkt = av_packet_alloc();
if (!pkt)
exit(1);
/* put sample parameters */
c->bit_rate = 400000;
/* resolution must be a multiple of two */
c->width = 352;
c->height = 288;
/* frames per second */
c->time_base = (AVRational){1, 25};
c->framerate = (AVRational){25, 1};
/* emit one intra frame every ten frames
* check frame pict_type before passing frame
* to encoder, if frame->pict_type is AV_PICTURE_TYPE_I
* then gop_size is ignored and the output of encoder
* will always be I frame irrespective to gop_size
*/
c->gop_size = 10;
c->max_b_frames = 1;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (codec->id == AV_CODEC_ID_H264)
av_opt_set(c->priv_data, "preset", "slow", 0);
/* open it */
ret = avcodec_open2(c, codec, NULL);
if (ret < 0) {
fprintf(stderr, "Could not open codec: %s\n", av_err2str(ret));
exit(1);
}
f = fopen(filename, "wb");
if (!f) {
fprintf(stderr, "Could not open %s\n", filename);
exit(1);
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
frame->format = c->pix_fmt;
frame->width = c->width;
frame->height = c->height;
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate the video frame data\n");
exit(1);
}
/* encode 1 second of video */
for (i = 0; i < 25; i++) {
fflush(stdout);
/* make sure the frame data is writable */
ret = av_frame_make_writable(frame);
if (ret < 0)
exit(1);
/* prepare a dummy image */
/* Y */
for (y = 0; y < c->height; y++) {
for (x = 0; x < c->width; x++) {
frame->data[0][y * frame->linesize[0] + x] = x + y + i * 3;
}
}
/* Cb and Cr */
for (y = 0; y < c->height/2; y++) {
for (x = 0; x < c->width/2; x++) {
frame->data[1][y * frame->linesize[1] + x] = 128 + y + i * 2;
frame->data[2][y * frame->linesize[2] + x] = 64 + x + i * 5;
}
}
frame->pts = i;
/* encode the image */
encode(c, frame, pkt, f);
}
/* flush the encoder */
encode(c, NULL, pkt, f);
/* add sequence end code to have a real MPEG file */
if (codec->id == AV_CODEC_ID_MPEG1VIDEO || codec->id == AV_CODEC_ID_MPEG2VIDEO)
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
avcodec_free_context(&c);
av_frame_free(&frame);
av_packet_free(&pkt);
return 0;
}

178
externals/ffmpeg/doc/examples/extract_mvs.c vendored Executable file
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@@ -0,0 +1,178 @@
/*
* Copyright (c) 2012 Stefano Sabatini
* Copyright (c) 2014 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#include <libavutil/motion_vector.h>
#include <libavformat/avformat.h>
static AVFormatContext *fmt_ctx = NULL;
static AVCodecContext *video_dec_ctx = NULL;
static AVStream *video_stream = NULL;
static const char *src_filename = NULL;
static int video_stream_idx = -1;
static AVFrame *frame = NULL;
static int video_frame_count = 0;
static int decode_packet(const AVPacket *pkt)
{
int ret = avcodec_send_packet(video_dec_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error while sending a packet to the decoder: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
ret = avcodec_receive_frame(video_dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
fprintf(stderr, "Error while receiving a frame from the decoder: %s\n", av_err2str(ret));
return ret;
}
if (ret >= 0) {
int i;
AVFrameSideData *sd;
video_frame_count++;
sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
if (sd) {
const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
for (i = 0; i < sd->size / sizeof(*mvs); i++) {
const AVMotionVector *mv = &mvs[i];
printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
video_frame_count, mv->source,
mv->w, mv->h, mv->src_x, mv->src_y,
mv->dst_x, mv->dst_y, mv->flags);
}
}
av_frame_unref(frame);
}
}
return 0;
}
static int open_codec_context(AVFormatContext *fmt_ctx, enum AVMediaType type)
{
int ret;
AVStream *st;
AVCodecContext *dec_ctx = NULL;
AVCodec *dec = NULL;
AVDictionary *opts = NULL;
ret = av_find_best_stream(fmt_ctx, type, -1, -1, &dec, 0);
if (ret < 0) {
fprintf(stderr, "Could not find %s stream in input file '%s'\n",
av_get_media_type_string(type), src_filename);
return ret;
} else {
int stream_idx = ret;
st = fmt_ctx->streams[stream_idx];
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx) {
fprintf(stderr, "Failed to allocate codec\n");
return AVERROR(EINVAL);
}
ret = avcodec_parameters_to_context(dec_ctx, st->codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters to codec context\n");
return ret;
}
/* Init the video decoder */
av_dict_set(&opts, "flags2", "+export_mvs", 0);
if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
fprintf(stderr, "Failed to open %s codec\n",
av_get_media_type_string(type));
return ret;
}
video_stream_idx = stream_idx;
video_stream = fmt_ctx->streams[video_stream_idx];
video_dec_ctx = dec_ctx;
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0;
AVPacket pkt = { 0 };
if (argc != 2) {
fprintf(stderr, "Usage: %s <video>\n", argv[0]);
exit(1);
}
src_filename = argv[1];
if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
fprintf(stderr, "Could not open source file %s\n", src_filename);
exit(1);
}
if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
fprintf(stderr, "Could not find stream information\n");
exit(1);
}
open_codec_context(fmt_ctx, AVMEDIA_TYPE_VIDEO);
av_dump_format(fmt_ctx, 0, src_filename, 0);
if (!video_stream) {
fprintf(stderr, "Could not find video stream in the input, aborting\n");
ret = 1;
goto end;
}
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Could not allocate frame\n");
ret = AVERROR(ENOMEM);
goto end;
}
printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
/* read frames from the file */
while (av_read_frame(fmt_ctx, &pkt) >= 0) {
if (pkt.stream_index == video_stream_idx)
ret = decode_packet(&pkt);
av_packet_unref(&pkt);
if (ret < 0)
break;
}
/* flush cached frames */
decode_packet(NULL);
end:
avcodec_free_context(&video_dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
return ret < 0;
}

363
externals/ffmpeg/doc/examples/filter_audio.c vendored Executable file
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@@ -0,0 +1,363 @@
/*
* copyright (c) 2013 Andrew Kelley
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* libavfilter API usage example.
*
* @example filter_audio.c
* This example will generate a sine wave audio,
* pass it through a simple filter chain, and then compute the MD5 checksum of
* the output data.
*
* The filter chain it uses is:
* (input) -> abuffer -> volume -> aformat -> abuffersink -> (output)
*
* abuffer: This provides the endpoint where you can feed the decoded samples.
* volume: In this example we hardcode it to 0.90.
* aformat: This converts the samples to the samplefreq, channel layout,
* and sample format required by the audio device.
* abuffersink: This provides the endpoint where you can read the samples after
* they have passed through the filter chain.
*/
#include <inttypes.h>
#include <math.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavutil/channel_layout.h"
#include "libavutil/md5.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavfilter/avfilter.h"
#include "libavfilter/buffersink.h"
#include "libavfilter/buffersrc.h"
#define INPUT_SAMPLERATE 48000
#define INPUT_FORMAT AV_SAMPLE_FMT_FLTP
#define INPUT_CHANNEL_LAYOUT AV_CH_LAYOUT_5POINT0
#define VOLUME_VAL 0.90
static int init_filter_graph(AVFilterGraph **graph, AVFilterContext **src,
AVFilterContext **sink)
{
AVFilterGraph *filter_graph;
AVFilterContext *abuffer_ctx;
const AVFilter *abuffer;
AVFilterContext *volume_ctx;
const AVFilter *volume;
AVFilterContext *aformat_ctx;
const AVFilter *aformat;
AVFilterContext *abuffersink_ctx;
const AVFilter *abuffersink;
AVDictionary *options_dict = NULL;
uint8_t options_str[1024];
uint8_t ch_layout[64];
int err;
/* Create a new filtergraph, which will contain all the filters. */
filter_graph = avfilter_graph_alloc();
if (!filter_graph) {
fprintf(stderr, "Unable to create filter graph.\n");
return AVERROR(ENOMEM);
}
/* Create the abuffer filter;
* it will be used for feeding the data into the graph. */
abuffer = avfilter_get_by_name("abuffer");
if (!abuffer) {
fprintf(stderr, "Could not find the abuffer filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffer_ctx = avfilter_graph_alloc_filter(filter_graph, abuffer, "src");
if (!abuffer_ctx) {
fprintf(stderr, "Could not allocate the abuffer instance.\n");
return AVERROR(ENOMEM);
}
/* Set the filter options through the AVOptions API. */
av_get_channel_layout_string(ch_layout, sizeof(ch_layout), 0, INPUT_CHANNEL_LAYOUT);
av_opt_set (abuffer_ctx, "channel_layout", ch_layout, AV_OPT_SEARCH_CHILDREN);
av_opt_set (abuffer_ctx, "sample_fmt", av_get_sample_fmt_name(INPUT_FORMAT), AV_OPT_SEARCH_CHILDREN);
av_opt_set_q (abuffer_ctx, "time_base", (AVRational){ 1, INPUT_SAMPLERATE }, AV_OPT_SEARCH_CHILDREN);
av_opt_set_int(abuffer_ctx, "sample_rate", INPUT_SAMPLERATE, AV_OPT_SEARCH_CHILDREN);
/* Now initialize the filter; we pass NULL options, since we have already
* set all the options above. */
err = avfilter_init_str(abuffer_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffer filter.\n");
return err;
}
/* Create volume filter. */
volume = avfilter_get_by_name("volume");
if (!volume) {
fprintf(stderr, "Could not find the volume filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
volume_ctx = avfilter_graph_alloc_filter(filter_graph, volume, "volume");
if (!volume_ctx) {
fprintf(stderr, "Could not allocate the volume instance.\n");
return AVERROR(ENOMEM);
}
/* A different way of passing the options is as key/value pairs in a
* dictionary. */
av_dict_set(&options_dict, "volume", AV_STRINGIFY(VOLUME_VAL), 0);
err = avfilter_init_dict(volume_ctx, &options_dict);
av_dict_free(&options_dict);
if (err < 0) {
fprintf(stderr, "Could not initialize the volume filter.\n");
return err;
}
/* Create the aformat filter;
* it ensures that the output is of the format we want. */
aformat = avfilter_get_by_name("aformat");
if (!aformat) {
fprintf(stderr, "Could not find the aformat filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
aformat_ctx = avfilter_graph_alloc_filter(filter_graph, aformat, "aformat");
if (!aformat_ctx) {
fprintf(stderr, "Could not allocate the aformat instance.\n");
return AVERROR(ENOMEM);
}
/* A third way of passing the options is in a string of the form
* key1=value1:key2=value2.... */
snprintf(options_str, sizeof(options_str),
"sample_fmts=%s:sample_rates=%d:channel_layouts=0x%"PRIx64,
av_get_sample_fmt_name(AV_SAMPLE_FMT_S16), 44100,
(uint64_t)AV_CH_LAYOUT_STEREO);
err = avfilter_init_str(aformat_ctx, options_str);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not initialize the aformat filter.\n");
return err;
}
/* Finally create the abuffersink filter;
* it will be used to get the filtered data out of the graph. */
abuffersink = avfilter_get_by_name("abuffersink");
if (!abuffersink) {
fprintf(stderr, "Could not find the abuffersink filter.\n");
return AVERROR_FILTER_NOT_FOUND;
}
abuffersink_ctx = avfilter_graph_alloc_filter(filter_graph, abuffersink, "sink");
if (!abuffersink_ctx) {
fprintf(stderr, "Could not allocate the abuffersink instance.\n");
return AVERROR(ENOMEM);
}
/* This filter takes no options. */
err = avfilter_init_str(abuffersink_ctx, NULL);
if (err < 0) {
fprintf(stderr, "Could not initialize the abuffersink instance.\n");
return err;
}
/* Connect the filters;
* in this simple case the filters just form a linear chain. */
err = avfilter_link(abuffer_ctx, 0, volume_ctx, 0);
if (err >= 0)
err = avfilter_link(volume_ctx, 0, aformat_ctx, 0);
if (err >= 0)
err = avfilter_link(aformat_ctx, 0, abuffersink_ctx, 0);
if (err < 0) {
fprintf(stderr, "Error connecting filters\n");
return err;
}
/* Configure the graph. */
err = avfilter_graph_config(filter_graph, NULL);
if (err < 0) {
av_log(NULL, AV_LOG_ERROR, "Error configuring the filter graph\n");
return err;
}
*graph = filter_graph;
*src = abuffer_ctx;
*sink = abuffersink_ctx;
return 0;
}
/* Do something useful with the filtered data: this simple
* example just prints the MD5 checksum of each plane to stdout. */
static int process_output(struct AVMD5 *md5, AVFrame *frame)
{
int planar = av_sample_fmt_is_planar(frame->format);
int channels = av_get_channel_layout_nb_channels(frame->channel_layout);
int planes = planar ? channels : 1;
int bps = av_get_bytes_per_sample(frame->format);
int plane_size = bps * frame->nb_samples * (planar ? 1 : channels);
int i, j;
for (i = 0; i < planes; i++) {
uint8_t checksum[16];
av_md5_init(md5);
av_md5_sum(checksum, frame->extended_data[i], plane_size);
fprintf(stdout, "plane %d: 0x", i);
for (j = 0; j < sizeof(checksum); j++)
fprintf(stdout, "%02X", checksum[j]);
fprintf(stdout, "\n");
}
fprintf(stdout, "\n");
return 0;
}
/* Construct a frame of audio data to be filtered;
* this simple example just synthesizes a sine wave. */
static int get_input(AVFrame *frame, int frame_num)
{
int err, i, j;
#define FRAME_SIZE 1024
/* Set up the frame properties and allocate the buffer for the data. */
frame->sample_rate = INPUT_SAMPLERATE;
frame->format = INPUT_FORMAT;
frame->channel_layout = INPUT_CHANNEL_LAYOUT;
frame->nb_samples = FRAME_SIZE;
frame->pts = frame_num * FRAME_SIZE;
err = av_frame_get_buffer(frame, 0);
if (err < 0)
return err;
/* Fill the data for each channel. */
for (i = 0; i < 5; i++) {
float *data = (float*)frame->extended_data[i];
for (j = 0; j < frame->nb_samples; j++)
data[j] = sin(2 * M_PI * (frame_num + j) * (i + 1) / FRAME_SIZE);
}
return 0;
}
int main(int argc, char *argv[])
{
struct AVMD5 *md5;
AVFilterGraph *graph;
AVFilterContext *src, *sink;
AVFrame *frame;
uint8_t errstr[1024];
float duration;
int err, nb_frames, i;
if (argc < 2) {
fprintf(stderr, "Usage: %s <duration>\n", argv[0]);
return 1;
}
duration = atof(argv[1]);
nb_frames = duration * INPUT_SAMPLERATE / FRAME_SIZE;
if (nb_frames <= 0) {
fprintf(stderr, "Invalid duration: %s\n", argv[1]);
return 1;
}
/* Allocate the frame we will be using to store the data. */
frame = av_frame_alloc();
if (!frame) {
fprintf(stderr, "Error allocating the frame\n");
return 1;
}
md5 = av_md5_alloc();
if (!md5) {
fprintf(stderr, "Error allocating the MD5 context\n");
return 1;
}
/* Set up the filtergraph. */
err = init_filter_graph(&graph, &src, &sink);
if (err < 0) {
fprintf(stderr, "Unable to init filter graph:");
goto fail;
}
/* the main filtering loop */
for (i = 0; i < nb_frames; i++) {
/* get an input frame to be filtered */
err = get_input(frame, i);
if (err < 0) {
fprintf(stderr, "Error generating input frame:");
goto fail;
}
/* Send the frame to the input of the filtergraph. */
err = av_buffersrc_add_frame(src, frame);
if (err < 0) {
av_frame_unref(frame);
fprintf(stderr, "Error submitting the frame to the filtergraph:");
goto fail;
}
/* Get all the filtered output that is available. */
while ((err = av_buffersink_get_frame(sink, frame)) >= 0) {
/* now do something with our filtered frame */
err = process_output(md5, frame);
if (err < 0) {
fprintf(stderr, "Error processing the filtered frame:");
goto fail;
}
av_frame_unref(frame);
}
if (err == AVERROR(EAGAIN)) {
/* Need to feed more frames in. */
continue;
} else if (err == AVERROR_EOF) {
/* Nothing more to do, finish. */
break;
} else if (err < 0) {
/* An error occurred. */
fprintf(stderr, "Error filtering the data:");
goto fail;
}
}
avfilter_graph_free(&graph);
av_frame_free(&frame);
av_freep(&md5);
return 0;
fail:
av_strerror(err, errstr, sizeof(errstr));
fprintf(stderr, "%s\n", errstr);
return 1;
}

292
externals/ffmpeg/doc/examples/filtering_audio.c vendored Executable file
View File

@@ -0,0 +1,292 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2012 Clément Bœsch
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for audio decoding and filtering
* @example filtering_audio.c
*/
#include <unistd.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
static const char *filter_descr = "aresample=8000,aformat=sample_fmts=s16:channel_layouts=mono";
static const char *player = "ffplay -f s16le -ar 8000 -ac 1 -";
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int audio_stream_index = -1;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the audio stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_AUDIO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find an audio stream in the input file\n");
return ret;
}
audio_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
/* init the audio decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open audio decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *abuffersrc = avfilter_get_by_name("abuffer");
const AVFilter *abuffersink = avfilter_get_by_name("abuffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
static const enum AVSampleFormat out_sample_fmts[] = { AV_SAMPLE_FMT_S16, -1 };
static const int64_t out_channel_layouts[] = { AV_CH_LAYOUT_MONO, -1 };
static const int out_sample_rates[] = { 8000, -1 };
const AVFilterLink *outlink;
AVRational time_base = fmt_ctx->streams[audio_stream_index]->time_base;
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer audio source: the decoded frames from the decoder will be inserted here. */
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout = av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
time_base.num, time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt), dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, abuffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
/* buffer audio sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, abuffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_fmts", out_sample_fmts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "channel_layouts", out_channel_layouts, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "sample_rates", out_sample_rates, -1,
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Print summary of the sink buffer
* Note: args buffer is reused to store channel layout string */
outlink = buffersink_ctx->inputs[0];
av_get_channel_layout_string(args, sizeof(args), -1, outlink->channel_layout);
av_log(NULL, AV_LOG_INFO, "Output: srate:%dHz fmt:%s chlayout:%s\n",
(int)outlink->sample_rate,
(char *)av_x_if_null(av_get_sample_fmt_name(outlink->format), "?"),
args);
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void print_frame(const AVFrame *frame)
{
const int n = frame->nb_samples * av_get_channel_layout_nb_channels(frame->channel_layout);
const uint16_t *p = (uint16_t*)frame->data[0];
const uint16_t *p_end = p + n;
while (p < p_end) {
fputc(*p & 0xff, stdout);
fputc(*p>>8 & 0xff, stdout);
p++;
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame = av_frame_alloc();
AVFrame *filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if (argc != 2) {
fprintf(stderr, "Usage: %s file | %s\n", argv[0], player);
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == audio_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
if (ret >= 0) {
/* push the audio data from decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the audio filtergraph\n");
break;
}
/* pull filtered audio from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
print_frame(filt_frame);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

291
externals/ffmpeg/doc/examples/filtering_video.c vendored Executable file
View File

@@ -0,0 +1,291 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for decoding and filtering
* @example filtering_video.c
*/
#define _XOPEN_SOURCE 600 /* for usleep */
#include <unistd.h>
#include <stdio.h>
#include <stdlib.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
const char *filter_descr = "scale=78:24,transpose=cclock";
/* other way:
scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
*/
static AVFormatContext *fmt_ctx;
static AVCodecContext *dec_ctx;
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
static int video_stream_index = -1;
static int64_t last_pts = AV_NOPTS_VALUE;
static int open_input_file(const char *filename)
{
int ret;
AVCodec *dec;
if ((ret = avformat_open_input(&fmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
/* select the video stream */
ret = av_find_best_stream(fmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &dec, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find a video stream in the input file\n");
return ret;
}
video_stream_index = ret;
/* create decoding context */
dec_ctx = avcodec_alloc_context3(dec);
if (!dec_ctx)
return AVERROR(ENOMEM);
avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
/* init the video decoder */
if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video decoder\n");
return ret;
}
return 0;
}
static int init_filters(const char *filters_descr)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = avfilter_get_by_name("buffer");
const AVFilter *buffersink = avfilter_get_by_name("buffersink");
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
/* buffer video source: the decoded frames from the decoder will be inserted here. */
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
time_base.num, time_base.den,
dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
/* buffer video sink: to terminate the filter chain. */
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_int_list(buffersink_ctx, "pix_fmts", pix_fmts,
AV_PIX_FMT_NONE, AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
/*
* Set the endpoints for the filter graph. The filter_graph will
* be linked to the graph described by filters_descr.
*/
/*
* The buffer source output must be connected to the input pad of
* the first filter described by filters_descr; since the first
* filter input label is not specified, it is set to "in" by
* default.
*/
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
/*
* The buffer sink input must be connected to the output pad of
* the last filter described by filters_descr; since the last
* filter output label is not specified, it is set to "out" by
* default.
*/
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse_ptr(filter_graph, filters_descr,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static void display_frame(const AVFrame *frame, AVRational time_base)
{
int x, y;
uint8_t *p0, *p;
int64_t delay;
if (frame->pts != AV_NOPTS_VALUE) {
if (last_pts != AV_NOPTS_VALUE) {
/* sleep roughly the right amount of time;
* usleep is in microseconds, just like AV_TIME_BASE. */
delay = av_rescale_q(frame->pts - last_pts,
time_base, AV_TIME_BASE_Q);
if (delay > 0 && delay < 1000000)
usleep(delay);
}
last_pts = frame->pts;
}
/* Trivial ASCII grayscale display. */
p0 = frame->data[0];
puts("\033c");
for (y = 0; y < frame->height; y++) {
p = p0;
for (x = 0; x < frame->width; x++)
putchar(" .-+#"[*(p++) / 52]);
putchar('\n');
p0 += frame->linesize[0];
}
fflush(stdout);
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet;
AVFrame *frame;
AVFrame *filt_frame;
if (argc != 2) {
fprintf(stderr, "Usage: %s file\n", argv[0]);
exit(1);
}
frame = av_frame_alloc();
filt_frame = av_frame_alloc();
if (!frame || !filt_frame) {
perror("Could not allocate frame");
exit(1);
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = init_filters(filter_descr)) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(fmt_ctx, &packet)) < 0)
break;
if (packet.stream_index == video_stream_index) {
ret = avcodec_send_packet(dec_ctx, &packet);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while sending a packet to the decoder\n");
break;
}
while (ret >= 0) {
ret = avcodec_receive_frame(dec_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
break;
} else if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while receiving a frame from the decoder\n");
goto end;
}
frame->pts = frame->best_effort_timestamp;
/* push the decoded frame into the filtergraph */
if (av_buffersrc_add_frame_flags(buffersrc_ctx, frame, AV_BUFFERSRC_FLAG_KEEP_REF) < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
break;
}
/* pull filtered frames from the filtergraph */
while (1) {
ret = av_buffersink_get_frame(buffersink_ctx, filt_frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
if (ret < 0)
goto end;
display_frame(filt_frame, buffersink_ctx->inputs[0]->time_base);
av_frame_unref(filt_frame);
}
av_frame_unref(frame);
}
}
av_packet_unref(&packet);
}
end:
avfilter_graph_free(&filter_graph);
avcodec_free_context(&dec_ctx);
avformat_close_input(&fmt_ctx);
av_frame_free(&frame);
av_frame_free(&filt_frame);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
exit(1);
}
exit(0);
}

156
externals/ffmpeg/doc/examples/http_multiclient.c vendored Executable file
View File

@@ -0,0 +1,156 @@
/*
* Copyright (c) 2015 Stephan Holljes
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat multi-client network API usage example.
*
* @example http_multiclient.c
* This example will serve a file without decoding or demuxing it over http.
* Multiple clients can connect and will receive the same file.
*/
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <unistd.h>
static void process_client(AVIOContext *client, const char *in_uri)
{
AVIOContext *input = NULL;
uint8_t buf[1024];
int ret, n, reply_code;
uint8_t *resource = NULL;
while ((ret = avio_handshake(client)) > 0) {
av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
// check for strlen(resource) is necessary, because av_opt_get()
// may return empty string.
if (resource && strlen(resource))
break;
av_freep(&resource);
}
if (ret < 0)
goto end;
av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
reply_code = 200;
} else {
reply_code = AVERROR_HTTP_NOT_FOUND;
}
if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
goto end;
}
av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
while ((ret = avio_handshake(client)) > 0);
if (ret < 0)
goto end;
fprintf(stderr, "Handshake performed.\n");
if (reply_code != 200)
goto end;
fprintf(stderr, "Opening input file.\n");
if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
av_err2str(ret));
goto end;
}
for(;;) {
n = avio_read(input, buf, sizeof(buf));
if (n < 0) {
if (n == AVERROR_EOF)
break;
av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
av_err2str(n));
break;
}
avio_write(client, buf, n);
avio_flush(client);
}
end:
fprintf(stderr, "Flushing client\n");
avio_flush(client);
fprintf(stderr, "Closing client\n");
avio_close(client);
fprintf(stderr, "Closing input\n");
avio_close(input);
av_freep(&resource);
}
int main(int argc, char **argv)
{
AVDictionary *options = NULL;
AVIOContext *client = NULL, *server = NULL;
const char *in_uri, *out_uri;
int ret, pid;
av_log_set_level(AV_LOG_TRACE);
if (argc < 3) {
printf("usage: %s input http://hostname[:port]\n"
"API example program to serve http to multiple clients.\n"
"\n", argv[0]);
return 1;
}
in_uri = argv[1];
out_uri = argv[2];
avformat_network_init();
if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
return ret;
}
if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
return ret;
}
fprintf(stderr, "Entering main loop.\n");
for(;;) {
if ((ret = avio_accept(server, &client)) < 0)
goto end;
fprintf(stderr, "Accepted client, forking process.\n");
// XXX: Since we don't reap our children and don't ignore signals
// this produces zombie processes.
pid = fork();
if (pid < 0) {
perror("Fork failed");
ret = AVERROR(errno);
goto end;
}
if (pid == 0) {
fprintf(stderr, "In child.\n");
process_client(client, in_uri);
avio_close(server);
exit(0);
}
if (pid > 0)
avio_close(client);
}
end:
avio_close(server);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

252
externals/ffmpeg/doc/examples/hw_decode.c vendored Executable file
View File

@@ -0,0 +1,252 @@
/*
* Copyright (c) 2017 Jun Zhao
* Copyright (c) 2017 Kaixuan Liu
*
* HW Acceleration API (video decoding) decode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* HW-Accelerated decoding example.
*
* @example hw_decode.c
* This example shows how to do HW-accelerated decoding with output
* frames from the HW video surfaces.
*/
#include <stdio.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
#include <libavutil/opt.h>
#include <libavutil/avassert.h>
#include <libavutil/imgutils.h>
static AVBufferRef *hw_device_ctx = NULL;
static enum AVPixelFormat hw_pix_fmt;
static FILE *output_file = NULL;
static int hw_decoder_init(AVCodecContext *ctx, const enum AVHWDeviceType type)
{
int err = 0;
if ((err = av_hwdevice_ctx_create(&hw_device_ctx, type,
NULL, NULL, 0)) < 0) {
fprintf(stderr, "Failed to create specified HW device.\n");
return err;
}
ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
return err;
}
static enum AVPixelFormat get_hw_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != -1; p++) {
if (*p == hw_pix_fmt)
return *p;
}
fprintf(stderr, "Failed to get HW surface format.\n");
return AV_PIX_FMT_NONE;
}
static int decode_write(AVCodecContext *avctx, AVPacket *packet)
{
AVFrame *frame = NULL, *sw_frame = NULL;
AVFrame *tmp_frame = NULL;
uint8_t *buffer = NULL;
int size;
int ret = 0;
ret = avcodec_send_packet(avctx, packet);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (1) {
if (!(frame = av_frame_alloc()) || !(sw_frame = av_frame_alloc())) {
fprintf(stderr, "Can not alloc frame\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = avcodec_receive_frame(avctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
av_frame_free(&sw_frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding\n");
goto fail;
}
if (frame->format == hw_pix_fmt) {
/* retrieve data from GPU to CPU */
if ((ret = av_hwframe_transfer_data(sw_frame, frame, 0)) < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
tmp_frame = sw_frame;
} else
tmp_frame = frame;
size = av_image_get_buffer_size(tmp_frame->format, tmp_frame->width,
tmp_frame->height, 1);
buffer = av_malloc(size);
if (!buffer) {
fprintf(stderr, "Can not alloc buffer\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ret = av_image_copy_to_buffer(buffer, size,
(const uint8_t * const *)tmp_frame->data,
(const int *)tmp_frame->linesize, tmp_frame->format,
tmp_frame->width, tmp_frame->height, 1);
if (ret < 0) {
fprintf(stderr, "Can not copy image to buffer\n");
goto fail;
}
if ((ret = fwrite(buffer, 1, size, output_file)) < 0) {
fprintf(stderr, "Failed to dump raw data.\n");
goto fail;
}
fail:
av_frame_free(&frame);
av_frame_free(&sw_frame);
av_freep(&buffer);
if (ret < 0)
return ret;
}
}
int main(int argc, char *argv[])
{
AVFormatContext *input_ctx = NULL;
int video_stream, ret;
AVStream *video = NULL;
AVCodecContext *decoder_ctx = NULL;
AVCodec *decoder = NULL;
AVPacket packet;
enum AVHWDeviceType type;
int i;
if (argc < 4) {
fprintf(stderr, "Usage: %s <device type> <input file> <output file>\n", argv[0]);
return -1;
}
type = av_hwdevice_find_type_by_name(argv[1]);
if (type == AV_HWDEVICE_TYPE_NONE) {
fprintf(stderr, "Device type %s is not supported.\n", argv[1]);
fprintf(stderr, "Available device types:");
while((type = av_hwdevice_iterate_types(type)) != AV_HWDEVICE_TYPE_NONE)
fprintf(stderr, " %s", av_hwdevice_get_type_name(type));
fprintf(stderr, "\n");
return -1;
}
/* open the input file */
if (avformat_open_input(&input_ctx, argv[2], NULL, NULL) != 0) {
fprintf(stderr, "Cannot open input file '%s'\n", argv[2]);
return -1;
}
if (avformat_find_stream_info(input_ctx, NULL) < 0) {
fprintf(stderr, "Cannot find input stream information.\n");
return -1;
}
/* find the video stream information */
ret = av_find_best_stream(input_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file\n");
return -1;
}
video_stream = ret;
for (i = 0;; i++) {
const AVCodecHWConfig *config = avcodec_get_hw_config(decoder, i);
if (!config) {
fprintf(stderr, "Decoder %s does not support device type %s.\n",
decoder->name, av_hwdevice_get_type_name(type));
return -1;
}
if (config->methods & AV_CODEC_HW_CONFIG_METHOD_HW_DEVICE_CTX &&
config->device_type == type) {
hw_pix_fmt = config->pix_fmt;
break;
}
}
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = input_ctx->streams[video_stream];
if (avcodec_parameters_to_context(decoder_ctx, video->codecpar) < 0)
return -1;
decoder_ctx->get_format = get_hw_format;
if (hw_decoder_init(decoder_ctx, type) < 0)
return -1;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0) {
fprintf(stderr, "Failed to open codec for stream #%u\n", video_stream);
return -1;
}
/* open the file to dump raw data */
output_file = fopen(argv[3], "w+");
/* actual decoding and dump the raw data */
while (ret >= 0) {
if ((ret = av_read_frame(input_ctx, &packet)) < 0)
break;
if (video_stream == packet.stream_index)
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
}
/* flush the decoder */
packet.data = NULL;
packet.size = 0;
ret = decode_write(decoder_ctx, &packet);
av_packet_unref(&packet);
if (output_file)
fclose(output_file);
avcodec_free_context(&decoder_ctx);
avformat_close_input(&input_ctx);
av_buffer_unref(&hw_device_ctx);
return 0;
}

60
externals/ffmpeg/doc/examples/metadata.c vendored Executable file
View File

@@ -0,0 +1,60 @@
/*
* Copyright (c) 2011 Reinhard Tartler
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Shows how the metadata API can be used in application programs.
* @example metadata.c
*/
#include <stdio.h>
#include <libavformat/avformat.h>
#include <libavutil/dict.h>
int main (int argc, char **argv)
{
AVFormatContext *fmt_ctx = NULL;
AVDictionaryEntry *tag = NULL;
int ret;
if (argc != 2) {
printf("usage: %s <input_file>\n"
"example program to demonstrate the use of the libavformat metadata API.\n"
"\n", argv[0]);
return 1;
}
if ((ret = avformat_open_input(&fmt_ctx, argv[1], NULL, NULL)))
return ret;
if ((ret = avformat_find_stream_info(fmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
while ((tag = av_dict_get(fmt_ctx->metadata, "", tag, AV_DICT_IGNORE_SUFFIX)))
printf("%s=%s\n", tag->key, tag->value);
avformat_close_input(&fmt_ctx);
return 0;
}

650
externals/ffmpeg/doc/examples/muxing.c vendored Executable file
View File

@@ -0,0 +1,650 @@
/*
* Copyright (c) 2003 Fabrice Bellard
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat API example.
*
* Output a media file in any supported libavformat format. The default
* codecs are used.
* @example muxing.c
*/
#include <stdlib.h>
#include <stdio.h>
#include <string.h>
#include <math.h>
#include <libavutil/avassert.h>
#include <libavutil/channel_layout.h>
#include <libavutil/opt.h>
#include <libavutil/mathematics.h>
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
#include <libswscale/swscale.h>
#include <libswresample/swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, AVCodecContext *c,
AVStream *st, AVFrame *frame)
{
int ret;
// send the frame to the encoder
ret = avcodec_send_frame(c, frame);
if (ret < 0) {
fprintf(stderr, "Error sending a frame to the encoder: %s\n",
av_err2str(ret));
exit(1);
}
while (ret >= 0) {
AVPacket pkt = { 0 };
ret = avcodec_receive_packet(c, &pkt);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error encoding a frame: %s\n", av_err2str(ret));
exit(1);
}
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(&pkt, c->time_base, st->time_base);
pkt.stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, &pkt);
ret = av_interleaved_write_frame(fmt_ctx, &pkt);
av_packet_unref(&pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing output packet: %s\n", av_err2str(ret));
exit(1);
}
}
return ret == AVERROR_EOF ? 1 : 0;
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
for (j = 0; j <frame->nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVFrame *frame;
int ret;
int dst_nb_samples;
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
return write_frame(oc, c, ost->st, frame);
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) > 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx, (const uint8_t * const *) ost->tmp_frame->data,
ost->tmp_frame->linesize, 0, c->height, ost->frame->data,
ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
return write_frame(oc, ost->enc, ost->st, get_video_frame(ost));
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, NULL, filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}

271
externals/ffmpeg/doc/examples/qsvdec.c vendored Executable file
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@@ -0,0 +1,271 @@
/*
* Copyright (c) 2015 Anton Khirnov
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel QSV-accelerated H.264 decoding example.
*
* @example qsvdec.c
* This example shows how to do QSV-accelerated H.264 decoding with output
* frames in the GPU video surfaces.
*/
#include "config.h"
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/buffer.h"
#include "libavutil/error.h"
#include "libavutil/hwcontext.h"
#include "libavutil/hwcontext_qsv.h"
#include "libavutil/mem.h"
typedef struct DecodeContext {
AVBufferRef *hw_device_ref;
} DecodeContext;
static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
{
while (*pix_fmts != AV_PIX_FMT_NONE) {
if (*pix_fmts == AV_PIX_FMT_QSV) {
DecodeContext *decode = avctx->opaque;
AVHWFramesContext *frames_ctx;
AVQSVFramesContext *frames_hwctx;
int ret;
/* create a pool of surfaces to be used by the decoder */
avctx->hw_frames_ctx = av_hwframe_ctx_alloc(decode->hw_device_ref);
if (!avctx->hw_frames_ctx)
return AV_PIX_FMT_NONE;
frames_ctx = (AVHWFramesContext*)avctx->hw_frames_ctx->data;
frames_hwctx = frames_ctx->hwctx;
frames_ctx->format = AV_PIX_FMT_QSV;
frames_ctx->sw_format = avctx->sw_pix_fmt;
frames_ctx->width = FFALIGN(avctx->coded_width, 32);
frames_ctx->height = FFALIGN(avctx->coded_height, 32);
frames_ctx->initial_pool_size = 32;
frames_hwctx->frame_type = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
ret = av_hwframe_ctx_init(avctx->hw_frames_ctx);
if (ret < 0)
return AV_PIX_FMT_NONE;
return AV_PIX_FMT_QSV;
}
pix_fmts++;
}
fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
return AV_PIX_FMT_NONE;
}
static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
AVFrame *frame, AVFrame *sw_frame,
AVPacket *pkt, AVIOContext *output_ctx)
{
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
while (ret >= 0) {
int i, j;
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
break;
else if (ret < 0) {
fprintf(stderr, "Error during decoding\n");
return ret;
}
/* A real program would do something useful with the decoded frame here.
* We just retrieve the raw data and write it to a file, which is rather
* useless but pedagogic. */
ret = av_hwframe_transfer_data(sw_frame, frame, 0);
if (ret < 0) {
fprintf(stderr, "Error transferring the data to system memory\n");
goto fail;
}
for (i = 0; i < FF_ARRAY_ELEMS(sw_frame->data) && sw_frame->data[i]; i++)
for (j = 0; j < (sw_frame->height >> (i > 0)); j++)
avio_write(output_ctx, sw_frame->data[i] + j * sw_frame->linesize[i], sw_frame->width);
fail:
av_frame_unref(sw_frame);
av_frame_unref(frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_ctx = NULL;
AVStream *video_st = NULL;
AVCodecContext *decoder_ctx = NULL;
const AVCodec *decoder;
AVPacket pkt = { 0 };
AVFrame *frame = NULL, *sw_frame = NULL;
DecodeContext decode = { NULL };
AVIOContext *output_ctx = NULL;
int ret, i;
if (argc < 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
/* open the input file */
ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
goto finish;
}
/* find the first H.264 video stream */
for (i = 0; i < input_ctx->nb_streams; i++) {
AVStream *st = input_ctx->streams[i];
if (st->codecpar->codec_id == AV_CODEC_ID_H264 && !video_st)
video_st = st;
else
st->discard = AVDISCARD_ALL;
}
if (!video_st) {
fprintf(stderr, "No H.264 video stream in the input file\n");
goto finish;
}
/* open the hardware device */
ret = av_hwdevice_ctx_create(&decode.hw_device_ref, AV_HWDEVICE_TYPE_QSV,
"auto", NULL, 0);
if (ret < 0) {
fprintf(stderr, "Cannot open the hardware device\n");
goto finish;
}
/* initialize the decoder */
decoder = avcodec_find_decoder_by_name("h264_qsv");
if (!decoder) {
fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
goto finish;
}
decoder_ctx = avcodec_alloc_context3(decoder);
if (!decoder_ctx) {
ret = AVERROR(ENOMEM);
goto finish;
}
decoder_ctx->codec_id = AV_CODEC_ID_H264;
if (video_st->codecpar->extradata_size) {
decoder_ctx->extradata = av_mallocz(video_st->codecpar->extradata_size +
AV_INPUT_BUFFER_PADDING_SIZE);
if (!decoder_ctx->extradata) {
ret = AVERROR(ENOMEM);
goto finish;
}
memcpy(decoder_ctx->extradata, video_st->codecpar->extradata,
video_st->codecpar->extradata_size);
decoder_ctx->extradata_size = video_st->codecpar->extradata_size;
}
decoder_ctx->opaque = &decode;
decoder_ctx->get_format = get_format;
ret = avcodec_open2(decoder_ctx, NULL, NULL);
if (ret < 0) {
fprintf(stderr, "Error opening the decoder: ");
goto finish;
}
/* open the output stream */
ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Error opening the output context: ");
goto finish;
}
frame = av_frame_alloc();
sw_frame = av_frame_alloc();
if (!frame || !sw_frame) {
ret = AVERROR(ENOMEM);
goto finish;
}
/* actual decoding */
while (ret >= 0) {
ret = av_read_frame(input_ctx, &pkt);
if (ret < 0)
break;
if (pkt.stream_index == video_st->index)
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
av_packet_unref(&pkt);
}
/* flush the decoder */
pkt.data = NULL;
pkt.size = 0;
ret = decode_packet(&decode, decoder_ctx, frame, sw_frame, &pkt, output_ctx);
finish:
if (ret < 0) {
char buf[1024];
av_strerror(ret, buf, sizeof(buf));
fprintf(stderr, "%s\n", buf);
}
avformat_close_input(&input_ctx);
av_frame_free(&frame);
av_frame_free(&sw_frame);
avcodec_free_context(&decoder_ctx);
av_buffer_unref(&decode.hw_device_ref);
avio_close(output_ctx);
return ret;
}

191
externals/ffmpeg/doc/examples/remuxing.c vendored Executable file
View File

@@ -0,0 +1,191 @@
/*
* Copyright (c) 2013 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libavformat/libavcodec demuxing and muxing API example.
*
* Remux streams from one container format to another.
* @example remuxing.c
*/
#include <libavutil/timestamp.h>
#include <libavformat/avformat.h>
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt, const char *tag)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("%s: pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
tag,
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
int main(int argc, char **argv)
{
AVOutputFormat *ofmt = NULL;
AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
AVPacket pkt;
const char *in_filename, *out_filename;
int ret, i;
int stream_index = 0;
int *stream_mapping = NULL;
int stream_mapping_size = 0;
if (argc < 3) {
printf("usage: %s input output\n"
"API example program to remux a media file with libavformat and libavcodec.\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return 1;
}
in_filename = argv[1];
out_filename = argv[2];
if ((ret = avformat_open_input(&ifmt_ctx, in_filename, 0, 0)) < 0) {
fprintf(stderr, "Could not open input file '%s'", in_filename);
goto end;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, 0)) < 0) {
fprintf(stderr, "Failed to retrieve input stream information");
goto end;
}
av_dump_format(ifmt_ctx, 0, in_filename, 0);
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, out_filename);
if (!ofmt_ctx) {
fprintf(stderr, "Could not create output context\n");
ret = AVERROR_UNKNOWN;
goto end;
}
stream_mapping_size = ifmt_ctx->nb_streams;
stream_mapping = av_mallocz_array(stream_mapping_size, sizeof(*stream_mapping));
if (!stream_mapping) {
ret = AVERROR(ENOMEM);
goto end;
}
ofmt = ofmt_ctx->oformat;
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *out_stream;
AVStream *in_stream = ifmt_ctx->streams[i];
AVCodecParameters *in_codecpar = in_stream->codecpar;
if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO &&
in_codecpar->codec_type != AVMEDIA_TYPE_VIDEO &&
in_codecpar->codec_type != AVMEDIA_TYPE_SUBTITLE) {
stream_mapping[i] = -1;
continue;
}
stream_mapping[i] = stream_index++;
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
fprintf(stderr, "Failed allocating output stream\n");
ret = AVERROR_UNKNOWN;
goto end;
}
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
if (ret < 0) {
fprintf(stderr, "Failed to copy codec parameters\n");
goto end;
}
out_stream->codecpar->codec_tag = 0;
}
av_dump_format(ofmt_ctx, 0, out_filename, 1);
if (!(ofmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, out_filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open output file '%s'", out_filename);
goto end;
}
}
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file\n");
goto end;
}
while (1) {
AVStream *in_stream, *out_stream;
ret = av_read_frame(ifmt_ctx, &pkt);
if (ret < 0)
break;
in_stream = ifmt_ctx->streams[pkt.stream_index];
if (pkt.stream_index >= stream_mapping_size ||
stream_mapping[pkt.stream_index] < 0) {
av_packet_unref(&pkt);
continue;
}
pkt.stream_index = stream_mapping[pkt.stream_index];
out_stream = ofmt_ctx->streams[pkt.stream_index];
log_packet(ifmt_ctx, &pkt, "in");
/* copy packet */
pkt.pts = av_rescale_q_rnd(pkt.pts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.dts = av_rescale_q_rnd(pkt.dts, in_stream->time_base, out_stream->time_base, AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
pkt.duration = av_rescale_q(pkt.duration, in_stream->time_base, out_stream->time_base);
pkt.pos = -1;
log_packet(ofmt_ctx, &pkt, "out");
ret = av_interleaved_write_frame(ofmt_ctx, &pkt);
if (ret < 0) {
fprintf(stderr, "Error muxing packet\n");
break;
}
av_packet_unref(&pkt);
}
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
/* close output */
if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
av_freep(&stream_mapping);
if (ret < 0 && ret != AVERROR_EOF) {
fprintf(stderr, "Error occurred: %s\n", av_err2str(ret));
return 1;
}
return 0;
}

214
externals/ffmpeg/doc/examples/resampling_audio.c vendored Executable file
View File

@@ -0,0 +1,214 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @example resampling_audio.c
* libswresample API use example.
*/
#include <libavutil/opt.h>
#include <libavutil/channel_layout.h>
#include <libavutil/samplefmt.h>
#include <libswresample/swresample.h>
static int get_format_from_sample_fmt(const char **fmt,
enum AVSampleFormat sample_fmt)
{
int i;
struct sample_fmt_entry {
enum AVSampleFormat sample_fmt; const char *fmt_be, *fmt_le;
} sample_fmt_entries[] = {
{ AV_SAMPLE_FMT_U8, "u8", "u8" },
{ AV_SAMPLE_FMT_S16, "s16be", "s16le" },
{ AV_SAMPLE_FMT_S32, "s32be", "s32le" },
{ AV_SAMPLE_FMT_FLT, "f32be", "f32le" },
{ AV_SAMPLE_FMT_DBL, "f64be", "f64le" },
};
*fmt = NULL;
for (i = 0; i < FF_ARRAY_ELEMS(sample_fmt_entries); i++) {
struct sample_fmt_entry *entry = &sample_fmt_entries[i];
if (sample_fmt == entry->sample_fmt) {
*fmt = AV_NE(entry->fmt_be, entry->fmt_le);
return 0;
}
}
fprintf(stderr,
"Sample format %s not supported as output format\n",
av_get_sample_fmt_name(sample_fmt));
return AVERROR(EINVAL);
}
/**
* Fill dst buffer with nb_samples, generated starting from t.
*/
static void fill_samples(double *dst, int nb_samples, int nb_channels, int sample_rate, double *t)
{
int i, j;
double tincr = 1.0 / sample_rate, *dstp = dst;
const double c = 2 * M_PI * 440.0;
/* generate sin tone with 440Hz frequency and duplicated channels */
for (i = 0; i < nb_samples; i++) {
*dstp = sin(c * *t);
for (j = 1; j < nb_channels; j++)
dstp[j] = dstp[0];
dstp += nb_channels;
*t += tincr;
}
}
int main(int argc, char **argv)
{
int64_t src_ch_layout = AV_CH_LAYOUT_STEREO, dst_ch_layout = AV_CH_LAYOUT_SURROUND;
int src_rate = 48000, dst_rate = 44100;
uint8_t **src_data = NULL, **dst_data = NULL;
int src_nb_channels = 0, dst_nb_channels = 0;
int src_linesize, dst_linesize;
int src_nb_samples = 1024, dst_nb_samples, max_dst_nb_samples;
enum AVSampleFormat src_sample_fmt = AV_SAMPLE_FMT_DBL, dst_sample_fmt = AV_SAMPLE_FMT_S16;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
const char *fmt;
struct SwrContext *swr_ctx;
double t;
int ret;
if (argc != 2) {
fprintf(stderr, "Usage: %s output_file\n"
"API example program to show how to resample an audio stream with libswresample.\n"
"This program generates a series of audio frames, resamples them to a specified "
"output format and rate and saves them to an output file named output_file.\n",
argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create resampler context */
swr_ctx = swr_alloc();
if (!swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
ret = AVERROR(ENOMEM);
goto end;
}
/* set options */
av_opt_set_int(swr_ctx, "in_channel_layout", src_ch_layout, 0);
av_opt_set_int(swr_ctx, "in_sample_rate", src_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", src_sample_fmt, 0);
av_opt_set_int(swr_ctx, "out_channel_layout", dst_ch_layout, 0);
av_opt_set_int(swr_ctx, "out_sample_rate", dst_rate, 0);
av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", dst_sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
goto end;
}
/* allocate source and destination samples buffers */
src_nb_channels = av_get_channel_layout_nb_channels(src_ch_layout);
ret = av_samples_alloc_array_and_samples(&src_data, &src_linesize, src_nb_channels,
src_nb_samples, src_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate source samples\n");
goto end;
}
/* compute the number of converted samples: buffering is avoided
* ensuring that the output buffer will contain at least all the
* converted input samples */
max_dst_nb_samples = dst_nb_samples =
av_rescale_rnd(src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
/* buffer is going to be directly written to a rawaudio file, no alignment */
dst_nb_channels = av_get_channel_layout_nb_channels(dst_ch_layout);
ret = av_samples_alloc_array_and_samples(&dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 0);
if (ret < 0) {
fprintf(stderr, "Could not allocate destination samples\n");
goto end;
}
t = 0;
do {
/* generate synthetic audio */
fill_samples((double *)src_data[0], src_nb_samples, src_nb_channels, src_rate, &t);
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, src_rate) +
src_nb_samples, dst_rate, src_rate, AV_ROUND_UP);
if (dst_nb_samples > max_dst_nb_samples) {
av_freep(&dst_data[0]);
ret = av_samples_alloc(dst_data, &dst_linesize, dst_nb_channels,
dst_nb_samples, dst_sample_fmt, 1);
if (ret < 0)
break;
max_dst_nb_samples = dst_nb_samples;
}
/* convert to destination format */
ret = swr_convert(swr_ctx, dst_data, dst_nb_samples, (const uint8_t **)src_data, src_nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
goto end;
}
dst_bufsize = av_samples_get_buffer_size(&dst_linesize, dst_nb_channels,
ret, dst_sample_fmt, 1);
if (dst_bufsize < 0) {
fprintf(stderr, "Could not get sample buffer size\n");
goto end;
}
printf("t:%f in:%d out:%d\n", t, src_nb_samples, ret);
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
} while (t < 10);
if ((ret = get_format_from_sample_fmt(&fmt, dst_sample_fmt)) < 0)
goto end;
fprintf(stderr, "Resampling succeeded. Play the output file with the command:\n"
"ffplay -f %s -channel_layout %"PRId64" -channels %d -ar %d %s\n",
fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
end:
fclose(dst_file);
if (src_data)
av_freep(&src_data[0]);
av_freep(&src_data);
if (dst_data)
av_freep(&dst_data[0]);
av_freep(&dst_data);
swr_free(&swr_ctx);
return ret < 0;
}

140
externals/ffmpeg/doc/examples/scaling_video.c vendored Executable file
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@@ -0,0 +1,140 @@
/*
* Copyright (c) 2012 Stefano Sabatini
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* libswscale API use example.
* @example scaling_video.c
*/
#include <libavutil/imgutils.h>
#include <libavutil/parseutils.h>
#include <libswscale/swscale.h>
static void fill_yuv_image(uint8_t *data[4], int linesize[4],
int width, int height, int frame_index)
{
int x, y;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
data[0][y * linesize[0] + x] = x + y + frame_index * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
data[1][y * linesize[1] + x] = 128 + y + frame_index * 2;
data[2][y * linesize[2] + x] = 64 + x + frame_index * 5;
}
}
}
int main(int argc, char **argv)
{
uint8_t *src_data[4], *dst_data[4];
int src_linesize[4], dst_linesize[4];
int src_w = 320, src_h = 240, dst_w, dst_h;
enum AVPixelFormat src_pix_fmt = AV_PIX_FMT_YUV420P, dst_pix_fmt = AV_PIX_FMT_RGB24;
const char *dst_size = NULL;
const char *dst_filename = NULL;
FILE *dst_file;
int dst_bufsize;
struct SwsContext *sws_ctx;
int i, ret;
if (argc != 3) {
fprintf(stderr, "Usage: %s output_file output_size\n"
"API example program to show how to scale an image with libswscale.\n"
"This program generates a series of pictures, rescales them to the given "
"output_size and saves them to an output file named output_file\n."
"\n", argv[0]);
exit(1);
}
dst_filename = argv[1];
dst_size = argv[2];
if (av_parse_video_size(&dst_w, &dst_h, dst_size) < 0) {
fprintf(stderr,
"Invalid size '%s', must be in the form WxH or a valid size abbreviation\n",
dst_size);
exit(1);
}
dst_file = fopen(dst_filename, "wb");
if (!dst_file) {
fprintf(stderr, "Could not open destination file %s\n", dst_filename);
exit(1);
}
/* create scaling context */
sws_ctx = sws_getContext(src_w, src_h, src_pix_fmt,
dst_w, dst_h, dst_pix_fmt,
SWS_BILINEAR, NULL, NULL, NULL);
if (!sws_ctx) {
fprintf(stderr,
"Impossible to create scale context for the conversion "
"fmt:%s s:%dx%d -> fmt:%s s:%dx%d\n",
av_get_pix_fmt_name(src_pix_fmt), src_w, src_h,
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h);
ret = AVERROR(EINVAL);
goto end;
}
/* allocate source and destination image buffers */
if ((ret = av_image_alloc(src_data, src_linesize,
src_w, src_h, src_pix_fmt, 16)) < 0) {
fprintf(stderr, "Could not allocate source image\n");
goto end;
}
/* buffer is going to be written to rawvideo file, no alignment */
if ((ret = av_image_alloc(dst_data, dst_linesize,
dst_w, dst_h, dst_pix_fmt, 1)) < 0) {
fprintf(stderr, "Could not allocate destination image\n");
goto end;
}
dst_bufsize = ret;
for (i = 0; i < 100; i++) {
/* generate synthetic video */
fill_yuv_image(src_data, src_linesize, src_w, src_h, i);
/* convert to destination format */
sws_scale(sws_ctx, (const uint8_t * const*)src_data,
src_linesize, 0, src_h, dst_data, dst_linesize);
/* write scaled image to file */
fwrite(dst_data[0], 1, dst_bufsize, dst_file);
}
fprintf(stderr, "Scaling succeeded. Play the output file with the command:\n"
"ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
end:
fclose(dst_file);
av_freep(&src_data[0]);
av_freep(&dst_data[0]);
sws_freeContext(sws_ctx);
return ret < 0;
}

885
externals/ffmpeg/doc/examples/transcode_aac.c vendored Executable file
View File

@@ -0,0 +1,885 @@
/*
* Copyright (c) 2013-2018 Andreas Unterweger
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Simple audio converter
*
* @example transcode_aac.c
* Convert an input audio file to AAC in an MP4 container using FFmpeg.
* Formats other than MP4 are supported based on the output file extension.
* @author Andreas Unterweger (dustsigns@gmail.com)
*/
#include <stdio.h>
#include "libavformat/avformat.h"
#include "libavformat/avio.h"
#include "libavcodec/avcodec.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/frame.h"
#include "libavutil/opt.h"
#include "libswresample/swresample.h"
/* The output bit rate in bit/s */
#define OUTPUT_BIT_RATE 96000
/* The number of output channels */
#define OUTPUT_CHANNELS 2
/**
* Open an input file and the required decoder.
* @param filename File to be opened
* @param[out] input_format_context Format context of opened file
* @param[out] input_codec_context Codec context of opened file
* @return Error code (0 if successful)
*/
static int open_input_file(const char *filename,
AVFormatContext **input_format_context,
AVCodecContext **input_codec_context)
{
AVCodecContext *avctx;
AVCodec *input_codec;
int error;
/* Open the input file to read from it. */
if ((error = avformat_open_input(input_format_context, filename, NULL,
NULL)) < 0) {
fprintf(stderr, "Could not open input file '%s' (error '%s')\n",
filename, av_err2str(error));
*input_format_context = NULL;
return error;
}
/* Get information on the input file (number of streams etc.). */
if ((error = avformat_find_stream_info(*input_format_context, NULL)) < 0) {
fprintf(stderr, "Could not open find stream info (error '%s')\n",
av_err2str(error));
avformat_close_input(input_format_context);
return error;
}
/* Make sure that there is only one stream in the input file. */
if ((*input_format_context)->nb_streams != 1) {
fprintf(stderr, "Expected one audio input stream, but found %d\n",
(*input_format_context)->nb_streams);
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/* Find a decoder for the audio stream. */
if (!(input_codec = avcodec_find_decoder((*input_format_context)->streams[0]->codecpar->codec_id))) {
fprintf(stderr, "Could not find input codec\n");
avformat_close_input(input_format_context);
return AVERROR_EXIT;
}
/* Allocate a new decoding context. */
avctx = avcodec_alloc_context3(input_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate a decoding context\n");
avformat_close_input(input_format_context);
return AVERROR(ENOMEM);
}
/* Initialize the stream parameters with demuxer information. */
error = avcodec_parameters_to_context(avctx, (*input_format_context)->streams[0]->codecpar);
if (error < 0) {
avformat_close_input(input_format_context);
avcodec_free_context(&avctx);
return error;
}
/* Open the decoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, input_codec, NULL)) < 0) {
fprintf(stderr, "Could not open input codec (error '%s')\n",
av_err2str(error));
avcodec_free_context(&avctx);
avformat_close_input(input_format_context);
return error;
}
/* Save the decoder context for easier access later. */
*input_codec_context = avctx;
return 0;
}
/**
* Open an output file and the required encoder.
* Also set some basic encoder parameters.
* Some of these parameters are based on the input file's parameters.
* @param filename File to be opened
* @param input_codec_context Codec context of input file
* @param[out] output_format_context Format context of output file
* @param[out] output_codec_context Codec context of output file
* @return Error code (0 if successful)
*/
static int open_output_file(const char *filename,
AVCodecContext *input_codec_context,
AVFormatContext **output_format_context,
AVCodecContext **output_codec_context)
{
AVCodecContext *avctx = NULL;
AVIOContext *output_io_context = NULL;
AVStream *stream = NULL;
AVCodec *output_codec = NULL;
int error;
/* Open the output file to write to it. */
if ((error = avio_open(&output_io_context, filename,
AVIO_FLAG_WRITE)) < 0) {
fprintf(stderr, "Could not open output file '%s' (error '%s')\n",
filename, av_err2str(error));
return error;
}
/* Create a new format context for the output container format. */
if (!(*output_format_context = avformat_alloc_context())) {
fprintf(stderr, "Could not allocate output format context\n");
return AVERROR(ENOMEM);
}
/* Associate the output file (pointer) with the container format context. */
(*output_format_context)->pb = output_io_context;
/* Guess the desired container format based on the file extension. */
if (!((*output_format_context)->oformat = av_guess_format(NULL, filename,
NULL))) {
fprintf(stderr, "Could not find output file format\n");
goto cleanup;
}
if (!((*output_format_context)->url = av_strdup(filename))) {
fprintf(stderr, "Could not allocate url.\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/* Find the encoder to be used by its name. */
if (!(output_codec = avcodec_find_encoder(AV_CODEC_ID_AAC))) {
fprintf(stderr, "Could not find an AAC encoder.\n");
goto cleanup;
}
/* Create a new audio stream in the output file container. */
if (!(stream = avformat_new_stream(*output_format_context, NULL))) {
fprintf(stderr, "Could not create new stream\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
avctx = avcodec_alloc_context3(output_codec);
if (!avctx) {
fprintf(stderr, "Could not allocate an encoding context\n");
error = AVERROR(ENOMEM);
goto cleanup;
}
/* Set the basic encoder parameters.
* The input file's sample rate is used to avoid a sample rate conversion. */
avctx->channels = OUTPUT_CHANNELS;
avctx->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
avctx->sample_rate = input_codec_context->sample_rate;
avctx->sample_fmt = output_codec->sample_fmts[0];
avctx->bit_rate = OUTPUT_BIT_RATE;
/* Allow the use of the experimental AAC encoder. */
avctx->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
/* Set the sample rate for the container. */
stream->time_base.den = input_codec_context->sample_rate;
stream->time_base.num = 1;
/* Some container formats (like MP4) require global headers to be present.
* Mark the encoder so that it behaves accordingly. */
if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
avctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Open the encoder for the audio stream to use it later. */
if ((error = avcodec_open2(avctx, output_codec, NULL)) < 0) {
fprintf(stderr, "Could not open output codec (error '%s')\n",
av_err2str(error));
goto cleanup;
}
error = avcodec_parameters_from_context(stream->codecpar, avctx);
if (error < 0) {
fprintf(stderr, "Could not initialize stream parameters\n");
goto cleanup;
}
/* Save the encoder context for easier access later. */
*output_codec_context = avctx;
return 0;
cleanup:
avcodec_free_context(&avctx);
avio_closep(&(*output_format_context)->pb);
avformat_free_context(*output_format_context);
*output_format_context = NULL;
return error < 0 ? error : AVERROR_EXIT;
}
/**
* Initialize one data packet for reading or writing.
* @param packet Packet to be initialized
*/
static void init_packet(AVPacket *packet)
{
av_init_packet(packet);
/* Set the packet data and size so that it is recognized as being empty. */
packet->data = NULL;
packet->size = 0;
}
/**
* Initialize one audio frame for reading from the input file.
* @param[out] frame Frame to be initialized
* @return Error code (0 if successful)
*/
static int init_input_frame(AVFrame **frame)
{
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate input frame\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Initialize the audio resampler based on the input and output codec settings.
* If the input and output sample formats differ, a conversion is required
* libswresample takes care of this, but requires initialization.
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param[out] resample_context Resample context for the required conversion
* @return Error code (0 if successful)
*/
static int init_resampler(AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext **resample_context)
{
int error;
/*
* Create a resampler context for the conversion.
* Set the conversion parameters.
* Default channel layouts based on the number of channels
* are assumed for simplicity (they are sometimes not detected
* properly by the demuxer and/or decoder).
*/
*resample_context = swr_alloc_set_opts(NULL,
av_get_default_channel_layout(output_codec_context->channels),
output_codec_context->sample_fmt,
output_codec_context->sample_rate,
av_get_default_channel_layout(input_codec_context->channels),
input_codec_context->sample_fmt,
input_codec_context->sample_rate,
0, NULL);
if (!*resample_context) {
fprintf(stderr, "Could not allocate resample context\n");
return AVERROR(ENOMEM);
}
/*
* Perform a sanity check so that the number of converted samples is
* not greater than the number of samples to be converted.
* If the sample rates differ, this case has to be handled differently
*/
av_assert0(output_codec_context->sample_rate == input_codec_context->sample_rate);
/* Open the resampler with the specified parameters. */
if ((error = swr_init(*resample_context)) < 0) {
fprintf(stderr, "Could not open resample context\n");
swr_free(resample_context);
return error;
}
return 0;
}
/**
* Initialize a FIFO buffer for the audio samples to be encoded.
* @param[out] fifo Sample buffer
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
{
/* Create the FIFO buffer based on the specified output sample format. */
if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
output_codec_context->channels, 1))) {
fprintf(stderr, "Could not allocate FIFO\n");
return AVERROR(ENOMEM);
}
return 0;
}
/**
* Write the header of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_header(AVFormatContext *output_format_context)
{
int error;
if ((error = avformat_write_header(output_format_context, NULL)) < 0) {
fprintf(stderr, "Could not write output file header (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* Decode one audio frame from the input file.
* @param frame Audio frame to be decoded
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param[out] data_present Indicates whether data has been decoded
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false, there
* is more data to be decoded, i.e., this
* function has to be called again.
* @return Error code (0 if successful)
*/
static int decode_audio_frame(AVFrame *frame,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
int *data_present, int *finished)
{
/* Packet used for temporary storage. */
AVPacket input_packet;
int error;
init_packet(&input_packet);
/* Read one audio frame from the input file into a temporary packet. */
if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
/* If we are at the end of the file, flush the decoder below. */
if (error == AVERROR_EOF)
*finished = 1;
else {
fprintf(stderr, "Could not read frame (error '%s')\n",
av_err2str(error));
return error;
}
}
/* Send the audio frame stored in the temporary packet to the decoder.
* The input audio stream decoder is used to do this. */
if ((error = avcodec_send_packet(input_codec_context, &input_packet)) < 0) {
fprintf(stderr, "Could not send packet for decoding (error '%s')\n",
av_err2str(error));
return error;
}
/* Receive one frame from the decoder. */
error = avcodec_receive_frame(input_codec_context, frame);
/* If the decoder asks for more data to be able to decode a frame,
* return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the end of the input file is reached, stop decoding. */
} else if (error == AVERROR_EOF) {
*finished = 1;
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not decode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return decoded data. */
} else {
*data_present = 1;
goto cleanup;
}
cleanup:
av_packet_unref(&input_packet);
return error;
}
/**
* Initialize a temporary storage for the specified number of audio samples.
* The conversion requires temporary storage due to the different format.
* The number of audio samples to be allocated is specified in frame_size.
* @param[out] converted_input_samples Array of converted samples. The
* dimensions are reference, channel
* (for multi-channel audio), sample.
* @param output_codec_context Codec context of the output file
* @param frame_size Number of samples to be converted in
* each round
* @return Error code (0 if successful)
*/
static int init_converted_samples(uint8_t ***converted_input_samples,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/* Allocate as many pointers as there are audio channels.
* Each pointer will later point to the audio samples of the corresponding
* channels (although it may be NULL for interleaved formats).
*/
if (!(*converted_input_samples = calloc(output_codec_context->channels,
sizeof(**converted_input_samples)))) {
fprintf(stderr, "Could not allocate converted input sample pointers\n");
return AVERROR(ENOMEM);
}
/* Allocate memory for the samples of all channels in one consecutive
* block for convenience. */
if ((error = av_samples_alloc(*converted_input_samples, NULL,
output_codec_context->channels,
frame_size,
output_codec_context->sample_fmt, 0)) < 0) {
fprintf(stderr,
"Could not allocate converted input samples (error '%s')\n",
av_err2str(error));
av_freep(&(*converted_input_samples)[0]);
free(*converted_input_samples);
return error;
}
return 0;
}
/**
* Convert the input audio samples into the output sample format.
* The conversion happens on a per-frame basis, the size of which is
* specified by frame_size.
* @param input_data Samples to be decoded. The dimensions are
* channel (for multi-channel audio), sample.
* @param[out] converted_data Converted samples. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @param resample_context Resample context for the conversion
* @return Error code (0 if successful)
*/
static int convert_samples(const uint8_t **input_data,
uint8_t **converted_data, const int frame_size,
SwrContext *resample_context)
{
int error;
/* Convert the samples using the resampler. */
if ((error = swr_convert(resample_context,
converted_data, frame_size,
input_data , frame_size)) < 0) {
fprintf(stderr, "Could not convert input samples (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
/**
* Add converted input audio samples to the FIFO buffer for later processing.
* @param fifo Buffer to add the samples to
* @param converted_input_samples Samples to be added. The dimensions are channel
* (for multi-channel audio), sample.
* @param frame_size Number of samples to be converted
* @return Error code (0 if successful)
*/
static int add_samples_to_fifo(AVAudioFifo *fifo,
uint8_t **converted_input_samples,
const int frame_size)
{
int error;
/* Make the FIFO as large as it needs to be to hold both,
* the old and the new samples. */
if ((error = av_audio_fifo_realloc(fifo, av_audio_fifo_size(fifo) + frame_size)) < 0) {
fprintf(stderr, "Could not reallocate FIFO\n");
return error;
}
/* Store the new samples in the FIFO buffer. */
if (av_audio_fifo_write(fifo, (void **)converted_input_samples,
frame_size) < frame_size) {
fprintf(stderr, "Could not write data to FIFO\n");
return AVERROR_EXIT;
}
return 0;
}
/**
* Read one audio frame from the input file, decode, convert and store
* it in the FIFO buffer.
* @param fifo Buffer used for temporary storage
* @param input_format_context Format context of the input file
* @param input_codec_context Codec context of the input file
* @param output_codec_context Codec context of the output file
* @param resampler_context Resample context for the conversion
* @param[out] finished Indicates whether the end of file has
* been reached and all data has been
* decoded. If this flag is false,
* there is more data to be decoded,
* i.e., this function has to be called
* again.
* @return Error code (0 if successful)
*/
static int read_decode_convert_and_store(AVAudioFifo *fifo,
AVFormatContext *input_format_context,
AVCodecContext *input_codec_context,
AVCodecContext *output_codec_context,
SwrContext *resampler_context,
int *finished)
{
/* Temporary storage of the input samples of the frame read from the file. */
AVFrame *input_frame = NULL;
/* Temporary storage for the converted input samples. */
uint8_t **converted_input_samples = NULL;
int data_present = 0;
int ret = AVERROR_EXIT;
/* Initialize temporary storage for one input frame. */
if (init_input_frame(&input_frame))
goto cleanup;
/* Decode one frame worth of audio samples. */
if (decode_audio_frame(input_frame, input_format_context,
input_codec_context, &data_present, finished))
goto cleanup;
/* If we are at the end of the file and there are no more samples
* in the decoder which are delayed, we are actually finished.
* This must not be treated as an error. */
if (*finished) {
ret = 0;
goto cleanup;
}
/* If there is decoded data, convert and store it. */
if (data_present) {
/* Initialize the temporary storage for the converted input samples. */
if (init_converted_samples(&converted_input_samples, output_codec_context,
input_frame->nb_samples))
goto cleanup;
/* Convert the input samples to the desired output sample format.
* This requires a temporary storage provided by converted_input_samples. */
if (convert_samples((const uint8_t**)input_frame->extended_data, converted_input_samples,
input_frame->nb_samples, resampler_context))
goto cleanup;
/* Add the converted input samples to the FIFO buffer for later processing. */
if (add_samples_to_fifo(fifo, converted_input_samples,
input_frame->nb_samples))
goto cleanup;
ret = 0;
}
ret = 0;
cleanup:
if (converted_input_samples) {
av_freep(&converted_input_samples[0]);
free(converted_input_samples);
}
av_frame_free(&input_frame);
return ret;
}
/**
* Initialize one input frame for writing to the output file.
* The frame will be exactly frame_size samples large.
* @param[out] frame Frame to be initialized
* @param output_codec_context Codec context of the output file
* @param frame_size Size of the frame
* @return Error code (0 if successful)
*/
static int init_output_frame(AVFrame **frame,
AVCodecContext *output_codec_context,
int frame_size)
{
int error;
/* Create a new frame to store the audio samples. */
if (!(*frame = av_frame_alloc())) {
fprintf(stderr, "Could not allocate output frame\n");
return AVERROR_EXIT;
}
/* Set the frame's parameters, especially its size and format.
* av_frame_get_buffer needs this to allocate memory for the
* audio samples of the frame.
* Default channel layouts based on the number of channels
* are assumed for simplicity. */
(*frame)->nb_samples = frame_size;
(*frame)->channel_layout = output_codec_context->channel_layout;
(*frame)->format = output_codec_context->sample_fmt;
(*frame)->sample_rate = output_codec_context->sample_rate;
/* Allocate the samples of the created frame. This call will make
* sure that the audio frame can hold as many samples as specified. */
if ((error = av_frame_get_buffer(*frame, 0)) < 0) {
fprintf(stderr, "Could not allocate output frame samples (error '%s')\n",
av_err2str(error));
av_frame_free(frame);
return error;
}
return 0;
}
/* Global timestamp for the audio frames. */
static int64_t pts = 0;
/**
* Encode one frame worth of audio to the output file.
* @param frame Samples to be encoded
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @param[out] data_present Indicates whether data has been
* encoded
* @return Error code (0 if successful)
*/
static int encode_audio_frame(AVFrame *frame,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context,
int *data_present)
{
/* Packet used for temporary storage. */
AVPacket output_packet;
int error;
init_packet(&output_packet);
/* Set a timestamp based on the sample rate for the container. */
if (frame) {
frame->pts = pts;
pts += frame->nb_samples;
}
/* Send the audio frame stored in the temporary packet to the encoder.
* The output audio stream encoder is used to do this. */
error = avcodec_send_frame(output_codec_context, frame);
/* The encoder signals that it has nothing more to encode. */
if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not send packet for encoding (error '%s')\n",
av_err2str(error));
return error;
}
/* Receive one encoded frame from the encoder. */
error = avcodec_receive_packet(output_codec_context, &output_packet);
/* If the encoder asks for more data to be able to provide an
* encoded frame, return indicating that no data is present. */
if (error == AVERROR(EAGAIN)) {
error = 0;
goto cleanup;
/* If the last frame has been encoded, stop encoding. */
} else if (error == AVERROR_EOF) {
error = 0;
goto cleanup;
} else if (error < 0) {
fprintf(stderr, "Could not encode frame (error '%s')\n",
av_err2str(error));
goto cleanup;
/* Default case: Return encoded data. */
} else {
*data_present = 1;
}
/* Write one audio frame from the temporary packet to the output file. */
if (*data_present &&
(error = av_write_frame(output_format_context, &output_packet)) < 0) {
fprintf(stderr, "Could not write frame (error '%s')\n",
av_err2str(error));
goto cleanup;
}
cleanup:
av_packet_unref(&output_packet);
return error;
}
/**
* Load one audio frame from the FIFO buffer, encode and write it to the
* output file.
* @param fifo Buffer used for temporary storage
* @param output_format_context Format context of the output file
* @param output_codec_context Codec context of the output file
* @return Error code (0 if successful)
*/
static int load_encode_and_write(AVAudioFifo *fifo,
AVFormatContext *output_format_context,
AVCodecContext *output_codec_context)
{
/* Temporary storage of the output samples of the frame written to the file. */
AVFrame *output_frame;
/* Use the maximum number of possible samples per frame.
* If there is less than the maximum possible frame size in the FIFO
* buffer use this number. Otherwise, use the maximum possible frame size. */
const int frame_size = FFMIN(av_audio_fifo_size(fifo),
output_codec_context->frame_size);
int data_written;
/* Initialize temporary storage for one output frame. */
if (init_output_frame(&output_frame, output_codec_context, frame_size))
return AVERROR_EXIT;
/* Read as many samples from the FIFO buffer as required to fill the frame.
* The samples are stored in the frame temporarily. */
if (av_audio_fifo_read(fifo, (void **)output_frame->data, frame_size) < frame_size) {
fprintf(stderr, "Could not read data from FIFO\n");
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
/* Encode one frame worth of audio samples. */
if (encode_audio_frame(output_frame, output_format_context,
output_codec_context, &data_written)) {
av_frame_free(&output_frame);
return AVERROR_EXIT;
}
av_frame_free(&output_frame);
return 0;
}
/**
* Write the trailer of the output file container.
* @param output_format_context Format context of the output file
* @return Error code (0 if successful)
*/
static int write_output_file_trailer(AVFormatContext *output_format_context)
{
int error;
if ((error = av_write_trailer(output_format_context)) < 0) {
fprintf(stderr, "Could not write output file trailer (error '%s')\n",
av_err2str(error));
return error;
}
return 0;
}
int main(int argc, char **argv)
{
AVFormatContext *input_format_context = NULL, *output_format_context = NULL;
AVCodecContext *input_codec_context = NULL, *output_codec_context = NULL;
SwrContext *resample_context = NULL;
AVAudioFifo *fifo = NULL;
int ret = AVERROR_EXIT;
if (argc != 3) {
fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
exit(1);
}
/* Open the input file for reading. */
if (open_input_file(argv[1], &input_format_context,
&input_codec_context))
goto cleanup;
/* Open the output file for writing. */
if (open_output_file(argv[2], input_codec_context,
&output_format_context, &output_codec_context))
goto cleanup;
/* Initialize the resampler to be able to convert audio sample formats. */
if (init_resampler(input_codec_context, output_codec_context,
&resample_context))
goto cleanup;
/* Initialize the FIFO buffer to store audio samples to be encoded. */
if (init_fifo(&fifo, output_codec_context))
goto cleanup;
/* Write the header of the output file container. */
if (write_output_file_header(output_format_context))
goto cleanup;
/* Loop as long as we have input samples to read or output samples
* to write; abort as soon as we have neither. */
while (1) {
/* Use the encoder's desired frame size for processing. */
const int output_frame_size = output_codec_context->frame_size;
int finished = 0;
/* Make sure that there is one frame worth of samples in the FIFO
* buffer so that the encoder can do its work.
* Since the decoder's and the encoder's frame size may differ, we
* need to FIFO buffer to store as many frames worth of input samples
* that they make up at least one frame worth of output samples. */
while (av_audio_fifo_size(fifo) < output_frame_size) {
/* Decode one frame worth of audio samples, convert it to the
* output sample format and put it into the FIFO buffer. */
if (read_decode_convert_and_store(fifo, input_format_context,
input_codec_context,
output_codec_context,
resample_context, &finished))
goto cleanup;
/* If we are at the end of the input file, we continue
* encoding the remaining audio samples to the output file. */
if (finished)
break;
}
/* If we have enough samples for the encoder, we encode them.
* At the end of the file, we pass the remaining samples to
* the encoder. */
while (av_audio_fifo_size(fifo) >= output_frame_size ||
(finished && av_audio_fifo_size(fifo) > 0))
/* Take one frame worth of audio samples from the FIFO buffer,
* encode it and write it to the output file. */
if (load_encode_and_write(fifo, output_format_context,
output_codec_context))
goto cleanup;
/* If we are at the end of the input file and have encoded
* all remaining samples, we can exit this loop and finish. */
if (finished) {
int data_written;
/* Flush the encoder as it may have delayed frames. */
do {
data_written = 0;
if (encode_audio_frame(NULL, output_format_context,
output_codec_context, &data_written))
goto cleanup;
} while (data_written);
break;
}
}
/* Write the trailer of the output file container. */
if (write_output_file_trailer(output_format_context))
goto cleanup;
ret = 0;
cleanup:
if (fifo)
av_audio_fifo_free(fifo);
swr_free(&resample_context);
if (output_codec_context)
avcodec_free_context(&output_codec_context);
if (output_format_context) {
avio_closep(&output_format_context->pb);
avformat_free_context(output_format_context);
}
if (input_codec_context)
avcodec_free_context(&input_codec_context);
if (input_format_context)
avformat_close_input(&input_format_context);
return ret;
}

620
externals/ffmpeg/doc/examples/transcoding.c vendored Executable file
View File

@@ -0,0 +1,620 @@
/*
* Copyright (c) 2010 Nicolas George
* Copyright (c) 2011 Stefano Sabatini
* Copyright (c) 2014 Andrey Utkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* API example for demuxing, decoding, filtering, encoding and muxing
* @example transcoding.c
*/
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavfilter/buffersink.h>
#include <libavfilter/buffersrc.h>
#include <libavutil/opt.h>
#include <libavutil/pixdesc.h>
static AVFormatContext *ifmt_ctx;
static AVFormatContext *ofmt_ctx;
typedef struct FilteringContext {
AVFilterContext *buffersink_ctx;
AVFilterContext *buffersrc_ctx;
AVFilterGraph *filter_graph;
} FilteringContext;
static FilteringContext *filter_ctx;
typedef struct StreamContext {
AVCodecContext *dec_ctx;
AVCodecContext *enc_ctx;
} StreamContext;
static StreamContext *stream_ctx;
static int open_input_file(const char *filename)
{
int ret;
unsigned int i;
ifmt_ctx = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open input file\n");
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot find stream information\n");
return ret;
}
stream_ctx = av_mallocz_array(ifmt_ctx->nb_streams, sizeof(*stream_ctx));
if (!stream_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
AVStream *stream = ifmt_ctx->streams[i];
AVCodec *dec = avcodec_find_decoder(stream->codecpar->codec_id);
AVCodecContext *codec_ctx;
if (!dec) {
av_log(NULL, AV_LOG_ERROR, "Failed to find decoder for stream #%u\n", i);
return AVERROR_DECODER_NOT_FOUND;
}
codec_ctx = avcodec_alloc_context3(dec);
if (!codec_ctx) {
av_log(NULL, AV_LOG_ERROR, "Failed to allocate the decoder context for stream #%u\n", i);
return AVERROR(ENOMEM);
}
ret = avcodec_parameters_to_context(codec_ctx, stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy decoder parameters to input decoder context "
"for stream #%u\n", i);
return ret;
}
/* Reencode video & audio and remux subtitles etc. */
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| codec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
if (codec_ctx->codec_type == AVMEDIA_TYPE_VIDEO)
codec_ctx->framerate = av_guess_frame_rate(ifmt_ctx, stream, NULL);
/* Open decoder */
ret = avcodec_open2(codec_ctx, dec, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to open decoder for stream #%u\n", i);
return ret;
}
}
stream_ctx[i].dec_ctx = codec_ctx;
}
av_dump_format(ifmt_ctx, 0, filename, 0);
return 0;
}
static int open_output_file(const char *filename)
{
AVStream *out_stream;
AVStream *in_stream;
AVCodecContext *dec_ctx, *enc_ctx;
AVCodec *encoder;
int ret;
unsigned int i;
ofmt_ctx = NULL;
avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, filename);
if (!ofmt_ctx) {
av_log(NULL, AV_LOG_ERROR, "Could not create output context\n");
return AVERROR_UNKNOWN;
}
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
out_stream = avformat_new_stream(ofmt_ctx, NULL);
if (!out_stream) {
av_log(NULL, AV_LOG_ERROR, "Failed allocating output stream\n");
return AVERROR_UNKNOWN;
}
in_stream = ifmt_ctx->streams[i];
dec_ctx = stream_ctx[i].dec_ctx;
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO
|| dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
/* in this example, we choose transcoding to same codec */
encoder = avcodec_find_encoder(dec_ctx->codec_id);
if (!encoder) {
av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
return AVERROR_INVALIDDATA;
}
enc_ctx = avcodec_alloc_context3(encoder);
if (!enc_ctx) {
av_log(NULL, AV_LOG_FATAL, "Failed to allocate the encoder context\n");
return AVERROR(ENOMEM);
}
/* In this example, we transcode to same properties (picture size,
* sample rate etc.). These properties can be changed for output
* streams easily using filters */
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
enc_ctx->height = dec_ctx->height;
enc_ctx->width = dec_ctx->width;
enc_ctx->sample_aspect_ratio = dec_ctx->sample_aspect_ratio;
/* take first format from list of supported formats */
if (encoder->pix_fmts)
enc_ctx->pix_fmt = encoder->pix_fmts[0];
else
enc_ctx->pix_fmt = dec_ctx->pix_fmt;
/* video time_base can be set to whatever is handy and supported by encoder */
enc_ctx->time_base = av_inv_q(dec_ctx->framerate);
} else {
enc_ctx->sample_rate = dec_ctx->sample_rate;
enc_ctx->channel_layout = dec_ctx->channel_layout;
enc_ctx->channels = av_get_channel_layout_nb_channels(enc_ctx->channel_layout);
/* take first format from list of supported formats */
enc_ctx->sample_fmt = encoder->sample_fmts[0];
enc_ctx->time_base = (AVRational){1, enc_ctx->sample_rate};
}
if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/* Third parameter can be used to pass settings to encoder */
ret = avcodec_open2(enc_ctx, encoder, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot open video encoder for stream #%u\n", i);
return ret;
}
ret = avcodec_parameters_from_context(out_stream->codecpar, enc_ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
return ret;
}
out_stream->time_base = enc_ctx->time_base;
stream_ctx[i].enc_ctx = enc_ctx;
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_UNKNOWN) {
av_log(NULL, AV_LOG_FATAL, "Elementary stream #%d is of unknown type, cannot proceed\n", i);
return AVERROR_INVALIDDATA;
} else {
/* if this stream must be remuxed */
ret = avcodec_parameters_copy(out_stream->codecpar, in_stream->codecpar);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Copying parameters for stream #%u failed\n", i);
return ret;
}
out_stream->time_base = in_stream->time_base;
}
}
av_dump_format(ofmt_ctx, 0, filename, 1);
if (!(ofmt_ctx->oformat->flags & AVFMT_NOFILE)) {
ret = avio_open(&ofmt_ctx->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Could not open output file '%s'", filename);
return ret;
}
}
/* init muxer, write output file header */
ret = avformat_write_header(ofmt_ctx, NULL);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error occurred when opening output file\n");
return ret;
}
return 0;
}
static int init_filter(FilteringContext* fctx, AVCodecContext *dec_ctx,
AVCodecContext *enc_ctx, const char *filter_spec)
{
char args[512];
int ret = 0;
const AVFilter *buffersrc = NULL;
const AVFilter *buffersink = NULL;
AVFilterContext *buffersrc_ctx = NULL;
AVFilterContext *buffersink_ctx = NULL;
AVFilterInOut *outputs = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
AVFilterGraph *filter_graph = avfilter_graph_alloc();
if (!outputs || !inputs || !filter_graph) {
ret = AVERROR(ENOMEM);
goto end;
}
if (dec_ctx->codec_type == AVMEDIA_TYPE_VIDEO) {
buffersrc = avfilter_get_by_name("buffer");
buffersink = avfilter_get_by_name("buffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
snprintf(args, sizeof(args),
"video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
dec_ctx->time_base.num, dec_ctx->time_base.den,
dec_ctx->sample_aspect_ratio.num,
dec_ctx->sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "pix_fmts",
(uint8_t*)&enc_ctx->pix_fmt, sizeof(enc_ctx->pix_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output pixel format\n");
goto end;
}
} else if (dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
buffersrc = avfilter_get_by_name("abuffer");
buffersink = avfilter_get_by_name("abuffersink");
if (!buffersrc || !buffersink) {
av_log(NULL, AV_LOG_ERROR, "filtering source or sink element not found\n");
ret = AVERROR_UNKNOWN;
goto end;
}
if (!dec_ctx->channel_layout)
dec_ctx->channel_layout =
av_get_default_channel_layout(dec_ctx->channels);
snprintf(args, sizeof(args),
"time_base=%d/%d:sample_rate=%d:sample_fmt=%s:channel_layout=0x%"PRIx64,
dec_ctx->time_base.num, dec_ctx->time_base.den, dec_ctx->sample_rate,
av_get_sample_fmt_name(dec_ctx->sample_fmt),
dec_ctx->channel_layout);
ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
args, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer source\n");
goto end;
}
ret = avfilter_graph_create_filter(&buffersink_ctx, buffersink, "out",
NULL, NULL, filter_graph);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot create audio buffer sink\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_fmts",
(uint8_t*)&enc_ctx->sample_fmt, sizeof(enc_ctx->sample_fmt),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample format\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "channel_layouts",
(uint8_t*)&enc_ctx->channel_layout,
sizeof(enc_ctx->channel_layout), AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output channel layout\n");
goto end;
}
ret = av_opt_set_bin(buffersink_ctx, "sample_rates",
(uint8_t*)&enc_ctx->sample_rate, sizeof(enc_ctx->sample_rate),
AV_OPT_SEARCH_CHILDREN);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Cannot set output sample rate\n");
goto end;
}
} else {
ret = AVERROR_UNKNOWN;
goto end;
}
/* Endpoints for the filter graph. */
outputs->name = av_strdup("in");
outputs->filter_ctx = buffersrc_ctx;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = buffersink_ctx;
inputs->pad_idx = 0;
inputs->next = NULL;
if (!outputs->name || !inputs->name) {
ret = AVERROR(ENOMEM);
goto end;
}
if ((ret = avfilter_graph_parse_ptr(filter_graph, filter_spec,
&inputs, &outputs, NULL)) < 0)
goto end;
if ((ret = avfilter_graph_config(filter_graph, NULL)) < 0)
goto end;
/* Fill FilteringContext */
fctx->buffersrc_ctx = buffersrc_ctx;
fctx->buffersink_ctx = buffersink_ctx;
fctx->filter_graph = filter_graph;
end:
avfilter_inout_free(&inputs);
avfilter_inout_free(&outputs);
return ret;
}
static int init_filters(void)
{
const char *filter_spec;
unsigned int i;
int ret;
filter_ctx = av_malloc_array(ifmt_ctx->nb_streams, sizeof(*filter_ctx));
if (!filter_ctx)
return AVERROR(ENOMEM);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
filter_ctx[i].buffersrc_ctx = NULL;
filter_ctx[i].buffersink_ctx = NULL;
filter_ctx[i].filter_graph = NULL;
if (!(ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO
|| ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO))
continue;
if (ifmt_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_VIDEO)
filter_spec = "null"; /* passthrough (dummy) filter for video */
else
filter_spec = "anull"; /* passthrough (dummy) filter for audio */
ret = init_filter(&filter_ctx[i], stream_ctx[i].dec_ctx,
stream_ctx[i].enc_ctx, filter_spec);
if (ret)
return ret;
}
return 0;
}
static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, int *got_frame) {
int ret;
int got_frame_local;
AVPacket enc_pkt;
int (*enc_func)(AVCodecContext *, AVPacket *, const AVFrame *, int *) =
(ifmt_ctx->streams[stream_index]->codecpar->codec_type ==
AVMEDIA_TYPE_VIDEO) ? avcodec_encode_video2 : avcodec_encode_audio2;
if (!got_frame)
got_frame = &got_frame_local;
av_log(NULL, AV_LOG_INFO, "Encoding frame\n");
/* encode filtered frame */
enc_pkt.data = NULL;
enc_pkt.size = 0;
av_init_packet(&enc_pkt);
ret = enc_func(stream_ctx[stream_index].enc_ctx, &enc_pkt,
filt_frame, got_frame);
av_frame_free(&filt_frame);
if (ret < 0)
return ret;
if (!(*got_frame))
return 0;
/* prepare packet for muxing */
enc_pkt.stream_index = stream_index;
av_packet_rescale_ts(&enc_pkt,
stream_ctx[stream_index].enc_ctx->time_base,
ofmt_ctx->streams[stream_index]->time_base);
av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
/* mux encoded frame */
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
return ret;
}
static int filter_encode_write_frame(AVFrame *frame, unsigned int stream_index)
{
int ret;
AVFrame *filt_frame;
av_log(NULL, AV_LOG_INFO, "Pushing decoded frame to filters\n");
/* push the decoded frame into the filtergraph */
ret = av_buffersrc_add_frame_flags(filter_ctx[stream_index].buffersrc_ctx,
frame, 0);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error while feeding the filtergraph\n");
return ret;
}
/* pull filtered frames from the filtergraph */
while (1) {
filt_frame = av_frame_alloc();
if (!filt_frame) {
ret = AVERROR(ENOMEM);
break;
}
av_log(NULL, AV_LOG_INFO, "Pulling filtered frame from filters\n");
ret = av_buffersink_get_frame(filter_ctx[stream_index].buffersink_ctx,
filt_frame);
if (ret < 0) {
/* if no more frames for output - returns AVERROR(EAGAIN)
* if flushed and no more frames for output - returns AVERROR_EOF
* rewrite retcode to 0 to show it as normal procedure completion
*/
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
ret = 0;
av_frame_free(&filt_frame);
break;
}
filt_frame->pict_type = AV_PICTURE_TYPE_NONE;
ret = encode_write_frame(filt_frame, stream_index, NULL);
if (ret < 0)
break;
}
return ret;
}
static int flush_encoder(unsigned int stream_index)
{
int ret;
int got_frame;
if (!(stream_ctx[stream_index].enc_ctx->codec->capabilities &
AV_CODEC_CAP_DELAY))
return 0;
while (1) {
av_log(NULL, AV_LOG_INFO, "Flushing stream #%u encoder\n", stream_index);
ret = encode_write_frame(NULL, stream_index, &got_frame);
if (ret < 0)
break;
if (!got_frame)
return 0;
}
return ret;
}
int main(int argc, char **argv)
{
int ret;
AVPacket packet = { .data = NULL, .size = 0 };
AVFrame *frame = NULL;
enum AVMediaType type;
unsigned int stream_index;
unsigned int i;
int got_frame;
int (*dec_func)(AVCodecContext *, AVFrame *, int *, const AVPacket *);
if (argc != 3) {
av_log(NULL, AV_LOG_ERROR, "Usage: %s <input file> <output file>\n", argv[0]);
return 1;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if ((ret = open_output_file(argv[2])) < 0)
goto end;
if ((ret = init_filters()) < 0)
goto end;
/* read all packets */
while (1) {
if ((ret = av_read_frame(ifmt_ctx, &packet)) < 0)
break;
stream_index = packet.stream_index;
type = ifmt_ctx->streams[packet.stream_index]->codecpar->codec_type;
av_log(NULL, AV_LOG_DEBUG, "Demuxer gave frame of stream_index %u\n",
stream_index);
if (filter_ctx[stream_index].filter_graph) {
av_log(NULL, AV_LOG_DEBUG, "Going to reencode&filter the frame\n");
frame = av_frame_alloc();
if (!frame) {
ret = AVERROR(ENOMEM);
break;
}
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
stream_ctx[stream_index].dec_ctx->time_base);
dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
avcodec_decode_audio4;
ret = dec_func(stream_ctx[stream_index].dec_ctx, frame,
&got_frame, &packet);
if (ret < 0) {
av_frame_free(&frame);
av_log(NULL, AV_LOG_ERROR, "Decoding failed\n");
break;
}
if (got_frame) {
frame->pts = frame->best_effort_timestamp;
ret = filter_encode_write_frame(frame, stream_index);
av_frame_free(&frame);
if (ret < 0)
goto end;
} else {
av_frame_free(&frame);
}
} else {
/* remux this frame without reencoding */
av_packet_rescale_ts(&packet,
ifmt_ctx->streams[stream_index]->time_base,
ofmt_ctx->streams[stream_index]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &packet);
if (ret < 0)
goto end;
}
av_packet_unref(&packet);
}
/* flush filters and encoders */
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
/* flush filter */
if (!filter_ctx[i].filter_graph)
continue;
ret = filter_encode_write_frame(NULL, i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing filter failed\n");
goto end;
}
/* flush encoder */
ret = flush_encoder(i);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Flushing encoder failed\n");
goto end;
}
}
av_write_trailer(ofmt_ctx);
end:
av_packet_unref(&packet);
av_frame_free(&frame);
for (i = 0; i < ifmt_ctx->nb_streams; i++) {
avcodec_free_context(&stream_ctx[i].dec_ctx);
if (ofmt_ctx && ofmt_ctx->nb_streams > i && ofmt_ctx->streams[i] && stream_ctx[i].enc_ctx)
avcodec_free_context(&stream_ctx[i].enc_ctx);
if (filter_ctx && filter_ctx[i].filter_graph)
avfilter_graph_free(&filter_ctx[i].filter_graph);
}
av_free(filter_ctx);
av_free(stream_ctx);
avformat_close_input(&ifmt_ctx);
if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
avio_closep(&ofmt_ctx->pb);
avformat_free_context(ofmt_ctx);
if (ret < 0)
av_log(NULL, AV_LOG_ERROR, "Error occurred: %s\n", av_err2str(ret));
return ret ? 1 : 0;
}

224
externals/ffmpeg/doc/examples/vaapi_encode.c vendored Executable file
View File

@@ -0,0 +1,224 @@
/*
* Video Acceleration API (video encoding) encode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel VAAPI-accelerated encoding example.
*
* @example vaapi_encode.c
* This example shows how to do VAAPI-accelerated encoding. now only support NV12
* raw file, usage like: vaapi_encode 1920 1080 input.yuv output.h264
*
*/
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <libavcodec/avcodec.h>
#include <libavutil/pixdesc.h>
#include <libavutil/hwcontext.h>
static int width, height;
static AVBufferRef *hw_device_ctx = NULL;
static int set_hwframe_ctx(AVCodecContext *ctx, AVBufferRef *hw_device_ctx)
{
AVBufferRef *hw_frames_ref;
AVHWFramesContext *frames_ctx = NULL;
int err = 0;
if (!(hw_frames_ref = av_hwframe_ctx_alloc(hw_device_ctx))) {
fprintf(stderr, "Failed to create VAAPI frame context.\n");
return -1;
}
frames_ctx = (AVHWFramesContext *)(hw_frames_ref->data);
frames_ctx->format = AV_PIX_FMT_VAAPI;
frames_ctx->sw_format = AV_PIX_FMT_NV12;
frames_ctx->width = width;
frames_ctx->height = height;
frames_ctx->initial_pool_size = 20;
if ((err = av_hwframe_ctx_init(hw_frames_ref)) < 0) {
fprintf(stderr, "Failed to initialize VAAPI frame context."
"Error code: %s\n",av_err2str(err));
av_buffer_unref(&hw_frames_ref);
return err;
}
ctx->hw_frames_ctx = av_buffer_ref(hw_frames_ref);
if (!ctx->hw_frames_ctx)
err = AVERROR(ENOMEM);
av_buffer_unref(&hw_frames_ref);
return err;
}
static int encode_write(AVCodecContext *avctx, AVFrame *frame, FILE *fout)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(avctx, frame)) < 0) {
fprintf(stderr, "Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(avctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
ret = fwrite(enc_pkt.data, enc_pkt.size, 1, fout);
av_packet_unref(&enc_pkt);
}
end:
ret = ((ret == AVERROR(EAGAIN)) ? 0 : -1);
return ret;
}
int main(int argc, char *argv[])
{
int size, err;
FILE *fin = NULL, *fout = NULL;
AVFrame *sw_frame = NULL, *hw_frame = NULL;
AVCodecContext *avctx = NULL;
AVCodec *codec = NULL;
const char *enc_name = "h264_vaapi";
if (argc < 5) {
fprintf(stderr, "Usage: %s <width> <height> <input file> <output file>\n", argv[0]);
return -1;
}
width = atoi(argv[1]);
height = atoi(argv[2]);
size = width * height;
if (!(fin = fopen(argv[3], "r"))) {
fprintf(stderr, "Fail to open input file : %s\n", strerror(errno));
return -1;
}
if (!(fout = fopen(argv[4], "w+b"))) {
fprintf(stderr, "Fail to open output file : %s\n", strerror(errno));
err = -1;
goto close;
}
err = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI,
NULL, NULL, 0);
if (err < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(err));
goto close;
}
if (!(codec = avcodec_find_encoder_by_name(enc_name))) {
fprintf(stderr, "Could not find encoder.\n");
err = -1;
goto close;
}
if (!(avctx = avcodec_alloc_context3(codec))) {
err = AVERROR(ENOMEM);
goto close;
}
avctx->width = width;
avctx->height = height;
avctx->time_base = (AVRational){1, 25};
avctx->framerate = (AVRational){25, 1};
avctx->sample_aspect_ratio = (AVRational){1, 1};
avctx->pix_fmt = AV_PIX_FMT_VAAPI;
/* set hw_frames_ctx for encoder's AVCodecContext */
if ((err = set_hwframe_ctx(avctx, hw_device_ctx)) < 0) {
fprintf(stderr, "Failed to set hwframe context.\n");
goto close;
}
if ((err = avcodec_open2(avctx, codec, NULL)) < 0) {
fprintf(stderr, "Cannot open video encoder codec. Error code: %s\n", av_err2str(err));
goto close;
}
while (1) {
if (!(sw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
/* read data into software frame, and transfer them into hw frame */
sw_frame->width = width;
sw_frame->height = height;
sw_frame->format = AV_PIX_FMT_NV12;
if ((err = av_frame_get_buffer(sw_frame, 0)) < 0)
goto close;
if ((err = fread((uint8_t*)(sw_frame->data[0]), size, 1, fin)) <= 0)
break;
if ((err = fread((uint8_t*)(sw_frame->data[1]), size/2, 1, fin)) <= 0)
break;
if (!(hw_frame = av_frame_alloc())) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_get_buffer(avctx->hw_frames_ctx, hw_frame, 0)) < 0) {
fprintf(stderr, "Error code: %s.\n", av_err2str(err));
goto close;
}
if (!hw_frame->hw_frames_ctx) {
err = AVERROR(ENOMEM);
goto close;
}
if ((err = av_hwframe_transfer_data(hw_frame, sw_frame, 0)) < 0) {
fprintf(stderr, "Error while transferring frame data to surface."
"Error code: %s.\n", av_err2str(err));
goto close;
}
if ((err = (encode_write(avctx, hw_frame, fout))) < 0) {
fprintf(stderr, "Failed to encode.\n");
goto close;
}
av_frame_free(&hw_frame);
av_frame_free(&sw_frame);
}
/* flush encoder */
err = encode_write(avctx, NULL, fout);
if (err == AVERROR_EOF)
err = 0;
close:
if (fin)
fclose(fin);
if (fout)
fclose(fout);
av_frame_free(&sw_frame);
av_frame_free(&hw_frame);
avcodec_free_context(&avctx);
av_buffer_unref(&hw_device_ctx);
return err;
}

306
externals/ffmpeg/doc/examples/vaapi_transcode.c vendored Executable file
View File

@@ -0,0 +1,306 @@
/*
* Video Acceleration API (video transcoding) transcode sample
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/**
* @file
* Intel VAAPI-accelerated transcoding example.
*
* @example vaapi_transcode.c
* This example shows how to do VAAPI-accelerated transcoding.
* Usage: vaapi_transcode input_stream codec output_stream
* e.g: - vaapi_transcode input.mp4 h264_vaapi output_h264.mp4
* - vaapi_transcode input.mp4 vp9_vaapi output_vp9.ivf
*/
#include <stdio.h>
#include <errno.h>
#include <libavutil/hwcontext.h>
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
static AVFormatContext *ifmt_ctx = NULL, *ofmt_ctx = NULL;
static AVBufferRef *hw_device_ctx = NULL;
static AVCodecContext *decoder_ctx = NULL, *encoder_ctx = NULL;
static int video_stream = -1;
static AVStream *ost;
static int initialized = 0;
static enum AVPixelFormat get_vaapi_format(AVCodecContext *ctx,
const enum AVPixelFormat *pix_fmts)
{
const enum AVPixelFormat *p;
for (p = pix_fmts; *p != AV_PIX_FMT_NONE; p++) {
if (*p == AV_PIX_FMT_VAAPI)
return *p;
}
fprintf(stderr, "Unable to decode this file using VA-API.\n");
return AV_PIX_FMT_NONE;
}
static int open_input_file(const char *filename)
{
int ret;
AVCodec *decoder = NULL;
AVStream *video = NULL;
if ((ret = avformat_open_input(&ifmt_ctx, filename, NULL, NULL)) < 0) {
fprintf(stderr, "Cannot open input file '%s', Error code: %s\n",
filename, av_err2str(ret));
return ret;
}
if ((ret = avformat_find_stream_info(ifmt_ctx, NULL)) < 0) {
fprintf(stderr, "Cannot find input stream information. Error code: %s\n",
av_err2str(ret));
return ret;
}
ret = av_find_best_stream(ifmt_ctx, AVMEDIA_TYPE_VIDEO, -1, -1, &decoder, 0);
if (ret < 0) {
fprintf(stderr, "Cannot find a video stream in the input file. "
"Error code: %s\n", av_err2str(ret));
return ret;
}
video_stream = ret;
if (!(decoder_ctx = avcodec_alloc_context3(decoder)))
return AVERROR(ENOMEM);
video = ifmt_ctx->streams[video_stream];
if ((ret = avcodec_parameters_to_context(decoder_ctx, video->codecpar)) < 0) {
fprintf(stderr, "avcodec_parameters_to_context error. Error code: %s\n",
av_err2str(ret));
return ret;
}
decoder_ctx->hw_device_ctx = av_buffer_ref(hw_device_ctx);
if (!decoder_ctx->hw_device_ctx) {
fprintf(stderr, "A hardware device reference create failed.\n");
return AVERROR(ENOMEM);
}
decoder_ctx->get_format = get_vaapi_format;
if ((ret = avcodec_open2(decoder_ctx, decoder, NULL)) < 0)
fprintf(stderr, "Failed to open codec for decoding. Error code: %s\n",
av_err2str(ret));
return ret;
}
static int encode_write(AVFrame *frame)
{
int ret = 0;
AVPacket enc_pkt;
av_init_packet(&enc_pkt);
enc_pkt.data = NULL;
enc_pkt.size = 0;
if ((ret = avcodec_send_frame(encoder_ctx, frame)) < 0) {
fprintf(stderr, "Error during encoding. Error code: %s\n", av_err2str(ret));
goto end;
}
while (1) {
ret = avcodec_receive_packet(encoder_ctx, &enc_pkt);
if (ret)
break;
enc_pkt.stream_index = 0;
av_packet_rescale_ts(&enc_pkt, ifmt_ctx->streams[video_stream]->time_base,
ofmt_ctx->streams[0]->time_base);
ret = av_interleaved_write_frame(ofmt_ctx, &enc_pkt);
if (ret < 0) {
fprintf(stderr, "Error during writing data to output file. "
"Error code: %s\n", av_err2str(ret));
return -1;
}
}
end:
if (ret == AVERROR_EOF)
return 0;
ret = ((ret == AVERROR(EAGAIN)) ? 0:-1);
return ret;
}
static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
{
AVFrame *frame;
int ret = 0;
ret = avcodec_send_packet(decoder_ctx, pkt);
if (ret < 0) {
fprintf(stderr, "Error during decoding. Error code: %s\n", av_err2str(ret));
return ret;
}
while (ret >= 0) {
if (!(frame = av_frame_alloc()))
return AVERROR(ENOMEM);
ret = avcodec_receive_frame(decoder_ctx, frame);
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
av_frame_free(&frame);
return 0;
} else if (ret < 0) {
fprintf(stderr, "Error while decoding. Error code: %s\n", av_err2str(ret));
goto fail;
}
if (!initialized) {
/* we need to ref hw_frames_ctx of decoder to initialize encoder's codec.
Only after we get a decoded frame, can we obtain its hw_frames_ctx */
encoder_ctx->hw_frames_ctx = av_buffer_ref(decoder_ctx->hw_frames_ctx);
if (!encoder_ctx->hw_frames_ctx) {
ret = AVERROR(ENOMEM);
goto fail;
}
/* set AVCodecContext Parameters for encoder, here we keep them stay
* the same as decoder.
* xxx: now the sample can't handle resolution change case.
*/
encoder_ctx->time_base = av_inv_q(decoder_ctx->framerate);
encoder_ctx->pix_fmt = AV_PIX_FMT_VAAPI;
encoder_ctx->width = decoder_ctx->width;
encoder_ctx->height = decoder_ctx->height;
if ((ret = avcodec_open2(encoder_ctx, enc_codec, NULL)) < 0) {
fprintf(stderr, "Failed to open encode codec. Error code: %s\n",
av_err2str(ret));
goto fail;
}
if (!(ost = avformat_new_stream(ofmt_ctx, enc_codec))) {
fprintf(stderr, "Failed to allocate stream for output format.\n");
ret = AVERROR(ENOMEM);
goto fail;
}
ost->time_base = encoder_ctx->time_base;
ret = avcodec_parameters_from_context(ost->codecpar, encoder_ctx);
if (ret < 0) {
fprintf(stderr, "Failed to copy the stream parameters. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
/* write the stream header */
if ((ret = avformat_write_header(ofmt_ctx, NULL)) < 0) {
fprintf(stderr, "Error while writing stream header. "
"Error code: %s\n", av_err2str(ret));
goto fail;
}
initialized = 1;
}
if ((ret = encode_write(frame)) < 0)
fprintf(stderr, "Error during encoding and writing.\n");
fail:
av_frame_free(&frame);
if (ret < 0)
return ret;
}
return 0;
}
int main(int argc, char **argv)
{
int ret = 0;
AVPacket dec_pkt;
AVCodec *enc_codec;
if (argc != 4) {
fprintf(stderr, "Usage: %s <input file> <encode codec> <output file>\n"
"The output format is guessed according to the file extension.\n"
"\n", argv[0]);
return -1;
}
ret = av_hwdevice_ctx_create(&hw_device_ctx, AV_HWDEVICE_TYPE_VAAPI, NULL, NULL, 0);
if (ret < 0) {
fprintf(stderr, "Failed to create a VAAPI device. Error code: %s\n", av_err2str(ret));
return -1;
}
if ((ret = open_input_file(argv[1])) < 0)
goto end;
if (!(enc_codec = avcodec_find_encoder_by_name(argv[2]))) {
fprintf(stderr, "Could not find encoder '%s'\n", argv[2]);
ret = -1;
goto end;
}
if ((ret = (avformat_alloc_output_context2(&ofmt_ctx, NULL, NULL, argv[3]))) < 0) {
fprintf(stderr, "Failed to deduce output format from file extension. Error code: "
"%s\n", av_err2str(ret));
goto end;
}
if (!(encoder_ctx = avcodec_alloc_context3(enc_codec))) {
ret = AVERROR(ENOMEM);
goto end;
}
ret = avio_open(&ofmt_ctx->pb, argv[3], AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Cannot open output file. "
"Error code: %s\n", av_err2str(ret));
goto end;
}
/* read all packets and only transcoding video */
while (ret >= 0) {
if ((ret = av_read_frame(ifmt_ctx, &dec_pkt)) < 0)
break;
if (video_stream == dec_pkt.stream_index)
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
}
/* flush decoder */
dec_pkt.data = NULL;
dec_pkt.size = 0;
ret = dec_enc(&dec_pkt, enc_codec);
av_packet_unref(&dec_pkt);
/* flush encoder */
ret = encode_write(NULL);
/* write the trailer for output stream */
av_write_trailer(ofmt_ctx);
end:
avformat_close_input(&ifmt_ctx);
avformat_close_input(&ofmt_ctx);
avcodec_free_context(&decoder_ctx);
avcodec_free_context(&encoder_ctx);
av_buffer_unref(&hw_device_ctx);
return ret;
}

686
externals/ffmpeg/doc/faq.texi vendored Executable file
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@@ -0,0 +1,686 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg FAQ
@titlepage
@center @titlefont{FFmpeg FAQ}
@end titlepage
@top
@contents
@chapter General Questions
@section Why doesn't FFmpeg support feature [xyz]?
Because no one has taken on that task yet. FFmpeg development is
driven by the tasks that are important to the individual developers.
If there is a feature that is important to you, the best way to get
it implemented is to undertake the task yourself or sponsor a developer.
@section FFmpeg does not support codec XXX. Can you include a Windows DLL loader to support it?
No. Windows DLLs are not portable, bloated and often slow.
Moreover FFmpeg strives to support all codecs natively.
A DLL loader is not conducive to that goal.
@section I cannot read this file although this format seems to be supported by ffmpeg.
Even if ffmpeg can read the container format, it may not support all its
codecs. Please consult the supported codec list in the ffmpeg
documentation.
@section Which codecs are supported by Windows?
Windows does not support standard formats like MPEG very well, unless you
install some additional codecs.
The following list of video codecs should work on most Windows systems:
@table @option
@item msmpeg4v2
.avi/.asf
@item msmpeg4
.asf only
@item wmv1
.asf only
@item wmv2
.asf only
@item mpeg4
Only if you have some MPEG-4 codec like ffdshow or Xvid installed.
@item mpeg1video
.mpg only
@end table
Note, ASF files often have .wmv or .wma extensions in Windows. It should also
be mentioned that Microsoft claims a patent on the ASF format, and may sue
or threaten users who create ASF files with non-Microsoft software. It is
strongly advised to avoid ASF where possible.
The following list of audio codecs should work on most Windows systems:
@table @option
@item adpcm_ima_wav
@item adpcm_ms
@item pcm_s16le
always
@item libmp3lame
If some MP3 codec like LAME is installed.
@end table
@chapter Compilation
@section @code{error: can't find a register in class 'GENERAL_REGS' while reloading 'asm'}
This is a bug in gcc. Do not report it to us. Instead, please report it to
the gcc developers. Note that we will not add workarounds for gcc bugs.
Also note that (some of) the gcc developers believe this is not a bug or
not a bug they should fix:
@url{https://gcc.gnu.org/bugzilla/show_bug.cgi?id=11203}.
Then again, some of them do not know the difference between an undecidable
problem and an NP-hard problem...
@section I have installed this library with my distro's package manager. Why does @command{configure} not see it?
Distributions usually split libraries in several packages. The main package
contains the files necessary to run programs using the library. The
development package contains the files necessary to build programs using the
library. Sometimes, docs and/or data are in a separate package too.
To build FFmpeg, you need to install the development package. It is usually
called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
build is finished, but be sure to keep the main package.
@section How do I make @command{pkg-config} find my libraries?
Somewhere along with your libraries, there is a @file{.pc} file (or several)
in a @file{pkgconfig} directory. You need to set environment variables to
point @command{pkg-config} to these files.
If you need to @emph{add} directories to @command{pkg-config}'s search list
(typical use case: library installed separately), add it to
@code{$PKG_CONFIG_PATH}:
@example
export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
@end example
If you need to @emph{replace} @command{pkg-config}'s search list
(typical use case: cross-compiling), set it in
@code{$PKG_CONFIG_LIBDIR}:
@example
export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
@end example
If you need to know the library's internal dependencies (typical use: static
linking), add the @code{--static} option to @command{pkg-config}:
@example
./configure --pkg-config-flags=--static
@end example
@section How do I use @command{pkg-config} when cross-compiling?
The best way is to install @command{pkg-config} in your cross-compilation
environment. It will automatically use the cross-compilation libraries.
You can also use @command{pkg-config} from the host environment by
specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
In that case, you must point @command{pkg-config} to the correct directories
using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
As an intermediate solution, you can place in your cross-compilation
environment a script that calls the host @command{pkg-config} with
@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
@example
#!/bin/sh
PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
export PKG_CONFIG_LIBDIR
exec /usr/bin/pkg-config "$@@"
@end example
@chapter Usage
@section ffmpeg does not work; what is wrong?
Try a @code{make distclean} in the ffmpeg source directory before the build.
If this does not help see
(@url{https://ffmpeg.org/bugreports.html}).
@section How do I encode single pictures into movies?
First, rename your pictures to follow a numerical sequence.
For example, img1.jpg, img2.jpg, img3.jpg,...
Then you may run:
@example
ffmpeg -f image2 -i img%d.jpg /tmp/a.mpg
@end example
Notice that @samp{%d} is replaced by the image number.
@file{img%03d.jpg} means the sequence @file{img001.jpg}, @file{img002.jpg}, etc.
Use the @option{-start_number} option to declare a starting number for
the sequence. This is useful if your sequence does not start with
@file{img001.jpg} but is still in a numerical order. The following
example will start with @file{img100.jpg}:
@example
ffmpeg -f image2 -start_number 100 -i img%d.jpg /tmp/a.mpg
@end example
If you have large number of pictures to rename, you can use the
following command to ease the burden. The command, using the bourne
shell syntax, symbolically links all files in the current directory
that match @code{*jpg} to the @file{/tmp} directory in the sequence of
@file{img001.jpg}, @file{img002.jpg} and so on.
@example
x=1; for i in *jpg; do counter=$(printf %03d $x); ln -s "$i" /tmp/img"$counter".jpg; x=$(($x+1)); done
@end example
If you want to sequence them by oldest modified first, substitute
@code{$(ls -r -t *jpg)} in place of @code{*jpg}.
Then run:
@example
ffmpeg -f image2 -i /tmp/img%03d.jpg /tmp/a.mpg
@end example
The same logic is used for any image format that ffmpeg reads.
You can also use @command{cat} to pipe images to ffmpeg:
@example
cat *.jpg | ffmpeg -f image2pipe -c:v mjpeg -i - output.mpg
@end example
@section How do I encode movie to single pictures?
Use:
@example
ffmpeg -i movie.mpg movie%d.jpg
@end example
The @file{movie.mpg} used as input will be converted to
@file{movie1.jpg}, @file{movie2.jpg}, etc...
Instead of relying on file format self-recognition, you may also use
@table @option
@item -c:v ppm
@item -c:v png
@item -c:v mjpeg
@end table
to force the encoding.
Applying that to the previous example:
@example
ffmpeg -i movie.mpg -f image2 -c:v mjpeg menu%d.jpg
@end example
Beware that there is no "jpeg" codec. Use "mjpeg" instead.
@section Why do I see a slight quality degradation with multithreaded MPEG* encoding?
For multithreaded MPEG* encoding, the encoded slices must be independent,
otherwise thread n would practically have to wait for n-1 to finish, so it's
quite logical that there is a small reduction of quality. This is not a bug.
@section How can I read from the standard input or write to the standard output?
Use @file{-} as file name.
@section -f jpeg doesn't work.
Try '-f image2 test%d.jpg'.
@section Why can I not change the frame rate?
Some codecs, like MPEG-1/2, only allow a small number of fixed frame rates.
Choose a different codec with the -c:v command line option.
@section How do I encode Xvid or DivX video with ffmpeg?
Both Xvid and DivX (version 4+) are implementations of the ISO MPEG-4
standard (note that there are many other coding formats that use this
same standard). Thus, use '-c:v mpeg4' to encode in these formats. The
default fourcc stored in an MPEG-4-coded file will be 'FMP4'. If you want
a different fourcc, use the '-vtag' option. E.g., '-vtag xvid' will
force the fourcc 'xvid' to be stored as the video fourcc rather than the
default.
@section Which are good parameters for encoding high quality MPEG-4?
'-mbd rd -flags +mv4+aic -trellis 2 -cmp 2 -subcmp 2 -g 300 -pass 1/2',
things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
@section Which are good parameters for encoding high quality MPEG-1/MPEG-2?
'-mbd rd -trellis 2 -cmp 2 -subcmp 2 -g 100 -pass 1/2'
but beware the '-g 100' might cause problems with some decoders.
Things to try: '-bf 2', '-mpv_flags qp_rd', '-mpv_flags mv0', '-mpv_flags skip_rd'.
@section Interlaced video looks very bad when encoded with ffmpeg, what is wrong?
You should use '-flags +ilme+ildct' and maybe '-flags +alt' for interlaced
material, and try '-top 0/1' if the result looks really messed-up.
@section How can I read DirectShow files?
If you have built FFmpeg with @code{./configure --enable-avisynth}
(only possible on MinGW/Cygwin platforms),
then you may use any file that DirectShow can read as input.
Just create an "input.avs" text file with this single line ...
@example
DirectShowSource("C:\path to your file\yourfile.asf")
@end example
... and then feed that text file to ffmpeg:
@example
ffmpeg -i input.avs
@end example
For ANY other help on AviSynth, please visit the
@uref{http://www.avisynth.org/, AviSynth homepage}.
@section How can I join video files?
To "join" video files is quite ambiguous. The following list explains the
different kinds of "joining" and points out how those are addressed in
FFmpeg. To join video files may mean:
@itemize
@item
To put them one after the other: this is called to @emph{concatenate} them
(in short: concat) and is addressed
@ref{How can I concatenate video files, in this very faq}.
@item
To put them together in the same file, to let the user choose between the
different versions (example: different audio languages): this is called to
@emph{multiplex} them together (in short: mux), and is done by simply
invoking ffmpeg with several @option{-i} options.
@item
For audio, to put all channels together in a single stream (example: two
mono streams into one stereo stream): this is sometimes called to
@emph{merge} them, and can be done using the
@url{ffmpeg-filters.html#amerge, @code{amerge}} filter.
@item
For audio, to play one on top of the other: this is called to @emph{mix}
them, and can be done by first merging them into a single stream and then
using the @url{ffmpeg-filters.html#pan, @code{pan}} filter to mix
the channels at will.
@item
For video, to display both together, side by side or one on top of a part of
the other; it can be done using the
@url{ffmpeg-filters.html#overlay, @code{overlay}} video filter.
@end itemize
@anchor{How can I concatenate video files}
@section How can I concatenate video files?
There are several solutions, depending on the exact circumstances.
@subsection Concatenating using the concat @emph{filter}
FFmpeg has a @url{ffmpeg-filters.html#concat,
@code{concat}} filter designed specifically for that, with examples in the
documentation. This operation is recommended if you need to re-encode.
@subsection Concatenating using the concat @emph{demuxer}
FFmpeg has a @url{ffmpeg-formats.html#concat,
@code{concat}} demuxer which you can use when you want to avoid a re-encode and
your format doesn't support file level concatenation.
@subsection Concatenating using the concat @emph{protocol} (file level)
FFmpeg has a @url{ffmpeg-protocols.html#concat,
@code{concat}} protocol designed specifically for that, with examples in the
documentation.
A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
video by merely concatenating the files containing them.
Hence you may concatenate your multimedia files by first transcoding them to
these privileged formats, then using the humble @code{cat} command (or the
equally humble @code{copy} under Windows), and finally transcoding back to your
format of choice.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
cat intermediate1.mpg intermediate2.mpg > intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
Additionally, you can use the @code{concat} protocol instead of @code{cat} or
@code{copy} which will avoid creation of a potentially huge intermediate file.
@example
ffmpeg -i input1.avi -qscale:v 1 intermediate1.mpg
ffmpeg -i input2.avi -qscale:v 1 intermediate2.mpg
ffmpeg -i concat:"intermediate1.mpg|intermediate2.mpg" -c copy intermediate_all.mpg
ffmpeg -i intermediate_all.mpg -qscale:v 2 output.avi
@end example
Note that you may need to escape the character "|" which is special for many
shells.
Another option is usage of named pipes, should your platform support it:
@example
mkfifo intermediate1.mpg
mkfifo intermediate2.mpg
ffmpeg -i input1.avi -qscale:v 1 -y intermediate1.mpg < /dev/null &
ffmpeg -i input2.avi -qscale:v 1 -y intermediate2.mpg < /dev/null &
cat intermediate1.mpg intermediate2.mpg |\
ffmpeg -f mpeg -i - -c:v mpeg4 -c:a libmp3lame output.avi
@end example
@subsection Concatenating using raw audio and video
Similarly, the yuv4mpegpipe format, and the raw video, raw audio codecs also
allow concatenation, and the transcoding step is almost lossless.
When using multiple yuv4mpegpipe(s), the first line needs to be discarded
from all but the first stream. This can be accomplished by piping through
@code{tail} as seen below. Note that when piping through @code{tail} you
must use command grouping, @code{@{ ;@}}, to background properly.
For example, let's say we want to concatenate two FLV files into an
output.flv file:
@example
mkfifo temp1.a
mkfifo temp1.v
mkfifo temp2.a
mkfifo temp2.v
mkfifo all.a
mkfifo all.v
ffmpeg -i input1.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp1.a < /dev/null &
ffmpeg -i input2.flv -vn -f u16le -c:a pcm_s16le -ac 2 -ar 44100 - > temp2.a < /dev/null &
ffmpeg -i input1.flv -an -f yuv4mpegpipe - > temp1.v < /dev/null &
@{ ffmpeg -i input2.flv -an -f yuv4mpegpipe - < /dev/null | tail -n +2 > temp2.v ; @} &
cat temp1.a temp2.a > all.a &
cat temp1.v temp2.v > all.v &
ffmpeg -f u16le -c:a pcm_s16le -ac 2 -ar 44100 -i all.a \
-f yuv4mpegpipe -i all.v \
-y output.flv
rm temp[12].[av] all.[av]
@end example
@section Using @option{-f lavfi}, audio becomes mono for no apparent reason.
Use @option{-dumpgraph -} to find out exactly where the channel layout is
lost.
Most likely, it is through @code{auto-inserted aresample}. Try to understand
why the converting filter was needed at that place.
Just before the output is a likely place, as @option{-f lavfi} currently
only support packed S16.
Then insert the correct @code{aformat} explicitly in the filtergraph,
specifying the exact format.
@example
aformat=sample_fmts=s16:channel_layouts=stereo
@end example
@section Why does FFmpeg not see the subtitles in my VOB file?
VOB and a few other formats do not have a global header that describes
everything present in the file. Instead, applications are supposed to scan
the file to see what it contains. Since VOB files are frequently large, only
the beginning is scanned. If the subtitles happen only later in the file,
they will not be initially detected.
Some applications, including the @code{ffmpeg} command-line tool, can only
work with streams that were detected during the initial scan; streams that
are detected later are ignored.
The size of the initial scan is controlled by two options: @code{probesize}
(default ~5 Mo) and @code{analyzeduration} (default 5,000,000 µs = 5 s). For
the subtitle stream to be detected, both values must be large enough.
@section Why was the @command{ffmpeg} @option{-sameq} option removed? What to use instead?
The @option{-sameq} option meant "same quantizer", and made sense only in a
very limited set of cases. Unfortunately, a lot of people mistook it for
"same quality" and used it in places where it did not make sense: it had
roughly the expected visible effect, but achieved it in a very inefficient
way.
Each encoder has its own set of options to set the quality-vs-size balance,
use the options for the encoder you are using to set the quality level to a
point acceptable for your tastes. The most common options to do that are
@option{-qscale} and @option{-qmax}, but you should peruse the documentation
of the encoder you chose.
@section I have a stretched video, why does scaling does not fix it?
A lot of video codecs and formats can store the @emph{aspect ratio} of the
video: this is the ratio between the width and the height of either the full
image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
SAR.
Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
of video standards, especially from the analogic-numeric transition era, use
non-square pixels.
Most processing filters in FFmpeg handle the aspect ratio to avoid
stretching the image: cropping adjusts the DAR to keep the SAR constant,
scaling adjusts the SAR to keep the DAR constant.
If you want to stretch, or “unstretch”, the image, you need to override the
information with the
@url{ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
Do not forget to examine carefully the original video to check whether the
stretching comes from the image or from the aspect ratio information.
For example, to fix a badly encoded EGA capture, use the following commands,
either the first one to upscale to square pixels or the second one to set
the correct aspect ratio or the third one to avoid transcoding (may not work
depending on the format / codec / player / phase of the moon):
@example
ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
@end example
@anchor{background task}
@section How do I run ffmpeg as a background task?
ffmpeg normally checks the console input, for entries like "q" to stop
and "?" to give help, while performing operations. ffmpeg does not have a way of
detecting when it is running as a background task.
When it checks the console input, that can cause the process running ffmpeg
in the background to suspend.
To prevent those input checks, allowing ffmpeg to run as a background task,
use the @url{ffmpeg.html#stdin-option, @code{-nostdin} option}
in the ffmpeg invocation. This is effective whether you run ffmpeg in a shell
or invoke ffmpeg in its own process via an operating system API.
As an alternative, when you are running ffmpeg in a shell, you can redirect
standard input to @code{/dev/null} (on Linux and macOS)
or @code{NUL} (on Windows). You can do this redirect either
on the ffmpeg invocation, or from a shell script which calls ffmpeg.
For example:
@example
ffmpeg -nostdin -i INPUT OUTPUT
@end example
or (on Linux, macOS, and other UNIX-like shells):
@example
ffmpeg -i INPUT OUTPUT </dev/null
@end example
or (on Windows):
@example
ffmpeg -i INPUT OUTPUT <NUL
@end example
@section How do I prevent ffmpeg from suspending with a message like @emph{suspended (tty output)}?
If you run ffmpeg in the background, you may find that its process suspends.
There may be a message like @emph{suspended (tty output)}. The question is how
to prevent the process from being suspended.
For example:
@example
% ffmpeg -i INPUT OUTPUT &> ~/tmp/log.txt &
[1] 93352
%
[1] + suspended (tty output) ffmpeg -i INPUT OUTPUT &>
@end example
The message "tty output" notwithstanding, the problem here is that
ffmpeg normally checks the console input when it runs. The operating system
detects this, and suspends the process until you can bring it to the
foreground and attend to it.
The solution is to use the right techniques to tell ffmpeg not to consult
console input. You can use the
@url{ffmpeg.html#stdin-option, @code{-nostdin} option},
or redirect standard input with @code{< /dev/null}.
See FAQ
@ref{background task, @emph{How do I run ffmpeg as a background task?}}
for details.
@chapter Development
@section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?
Yes. Check the @file{doc/examples} directory in the source
repository, also available online at:
@url{https://github.com/FFmpeg/FFmpeg/tree/master/doc/examples}.
Examples are also installed by default, usually in
@code{$PREFIX/share/ffmpeg/examples}.
Also you may read the Developers Guide of the FFmpeg documentation. Alternatively,
examine the source code for one of the many open source projects that
already incorporate FFmpeg at (@url{projects.html}).
@section Can you support my C compiler XXX?
It depends. If your compiler is C99-compliant, then patches to support
it are likely to be welcome if they do not pollute the source code
with @code{#ifdef}s related to the compiler.
@section Is Microsoft Visual C++ supported?
Yes. Please see the @uref{platform.html, Microsoft Visual C++}
section in the FFmpeg documentation.
@section Can you add automake, libtool or autoconf support?
No. These tools are too bloated and they complicate the build.
@section Why not rewrite FFmpeg in object-oriented C++?
FFmpeg is already organized in a highly modular manner and does not need to
be rewritten in a formal object language. Further, many of the developers
favor straight C; it works for them. For more arguments on this matter,
read @uref{https://web.archive.org/web/20111004021423/http://kernel.org/pub/linux/docs/lkml/#s15, "Programming Religion"}.
@section Why are the ffmpeg programs devoid of debugging symbols?
The build process creates @command{ffmpeg_g}, @command{ffplay_g}, etc. which
contain full debug information. Those binaries are stripped to create
@command{ffmpeg}, @command{ffplay}, etc. If you need the debug information, use
the *_g versions.
@section I do not like the LGPL, can I contribute code under the GPL instead?
Yes, as long as the code is optional and can easily and cleanly be placed
under #if CONFIG_GPL without breaking anything. So, for example, a new codec
or filter would be OK under GPL while a bug fix to LGPL code would not.
@section I'm using FFmpeg from within my C application but the linker complains about missing symbols from the libraries themselves.
FFmpeg builds static libraries by default. In static libraries, dependencies
are not handled. That has two consequences. First, you must specify the
libraries in dependency order: @code{-lavdevice} must come before
@code{-lavformat}, @code{-lavutil} must come after everything else, etc.
Second, external libraries that are used in FFmpeg have to be specified too.
An easy way to get the full list of required libraries in dependency order
is to use @code{pkg-config}.
@example
c99 -o program program.c $(pkg-config --cflags --libs libavformat libavcodec)
@end example
See @file{doc/example/Makefile} and @file{doc/example/pc-uninstalled} for
more details.
@section I'm using FFmpeg from within my C++ application but the linker complains about missing symbols which seem to be available.
FFmpeg is a pure C project, so to use the libraries within your C++ application
you need to explicitly state that you are using a C library. You can do this by
encompassing your FFmpeg includes using @code{extern "C"}.
See @url{http://www.parashift.com/c++-faq-lite/mixing-c-and-cpp.html#faq-32.3}
@section I'm using libavutil from within my C++ application but the compiler complains about 'UINT64_C' was not declared in this scope
FFmpeg is a pure C project using C99 math features, in order to enable C++
to use them you have to append -D__STDC_CONSTANT_MACROS to your CXXFLAGS
@section I have a file in memory / a API different from *open/*read/ libc how do I use it with libavformat?
You have to create a custom AVIOContext using @code{avio_alloc_context},
see @file{libavformat/aviobuf.c} in FFmpeg and @file{libmpdemux/demux_lavf.c} in MPlayer or MPlayer2 sources.
@section Where is the documentation about ffv1, msmpeg4, asv1, 4xm?
see @url{https://www.ffmpeg.org/~michael/}
@section How do I feed H.263-RTP (and other codecs in RTP) to libavcodec?
Even if peculiar since it is network oriented, RTP is a container like any
other. You have to @emph{demux} RTP before feeding the payload to libavcodec.
In this specific case please look at RFC 4629 to see how it should be done.
@section AVStream.r_frame_rate is wrong, it is much larger than the frame rate.
@code{r_frame_rate} is NOT the average frame rate, it is the smallest frame rate
that can accurately represent all timestamps. So no, it is not
wrong if it is larger than the average!
For example, if you have mixed 25 and 30 fps content, then @code{r_frame_rate}
will be 150 (it is the least common multiple).
If you are looking for the average frame rate, see @code{AVStream.avg_frame_rate}.
@section Why is @code{make fate} not running all tests?
Make sure you have the fate-suite samples and the @code{SAMPLES} Make variable
or @code{FATE_SAMPLES} environment variable or the @code{--samples}
@command{configure} option is set to the right path.
@section Why is @code{make fate} not finding the samples?
Do you happen to have a @code{~} character in the samples path to indicate a
home directory? The value is used in ways where the shell cannot expand it,
causing FATE to not find files. Just replace @code{~} by the full path.
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Automated Testing Environment
@titlepage
@center @titlefont{FFmpeg Automated Testing Environment}
@end titlepage
@node Top
@top
@contents
@chapter Introduction
FATE is an extended regression suite on the client-side and a means
for results aggregation and presentation on the server-side.
The first part of this document explains how you can use FATE from
your FFmpeg source directory to test your ffmpeg binary. The second
part describes how you can run FATE to submit the results to FFmpeg's
FATE server.
In any way you can have a look at the publicly viewable FATE results
by visiting this website:
@url{http://fate.ffmpeg.org/}
This is especially recommended for all people contributing source
code to FFmpeg, as it can be seen if some test on some platform broke
with their recent contribution. This usually happens on the platforms
the developers could not test on.
The second part of this document describes how you can run FATE to
submit your results to FFmpeg's FATE server. If you want to submit your
results be sure to check that your combination of CPU, OS and compiler
is not already listed on the above mentioned website.
In the third part you can find a comprehensive listing of FATE makefile
targets and variables.
@chapter Using FATE from your FFmpeg source directory
If you want to run FATE on your machine you need to have the samples
in place. You can get the samples via the build target fate-rsync.
Use this command from the top-level source directory:
@example
make fate-rsync SAMPLES=fate-suite/
make fate SAMPLES=fate-suite/
@end example
The above commands set the samples location by passing a makefile
variable via command line. It is also possible to set the samples
location at source configuration time by invoking configure with
@option{--samples=<path to the samples directory>}. Afterwards you can
invoke the makefile targets without setting the @var{SAMPLES} makefile
variable. This is illustrated by the following commands:
@example
./configure --samples=fate-suite/
make fate-rsync
make fate
@end example
Yet another way to tell FATE about the location of the sample
directory is by making sure the environment variable FATE_SAMPLES
contains the path to your samples directory. This can be achieved
by e.g. putting that variable in your shell profile or by setting
it in your interactive session.
@example
FATE_SAMPLES=fate-suite/ make fate
@end example
@float NOTE
Do not put a '~' character in the samples path to indicate a home
directory. Because of shell nuances, this will cause FATE to fail.
@end float
To use a custom wrapper to run the test, pass @option{--target-exec} to
@command{configure} or set the @var{TARGET_EXEC} Make variable.
@chapter Submitting the results to the FFmpeg result aggregation server
To submit your results to the server you should run fate through the
shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
to be invoked with a configuration file as its first argument.
@example
tests/fate.sh /path/to/fate_config
@end example
A configuration file template with comments describing the individual
configuration variables can be found at @file{doc/fate_config.sh.template}.
@ifhtml
The mentioned configuration template is also available here:
@verbatiminclude fate_config.sh.template
@end ifhtml
Create a configuration that suits your needs, based on the configuration
template. The @env{slot} configuration variable can be any string that is not
yet used, but it is suggested that you name it adhering to the following
pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
configuration file itself will be sourced in a shell script, therefore all
shell features may be used. This enables you to setup the environment as you
need it for your build.
For your first test runs the @env{fate_recv} variable should be empty or
commented out. This will run everything as normal except that it will omit
the submission of the results to the server. The following files should be
present in $workdir as specified in the configuration file:
@itemize
@item configure.log
@item compile.log
@item test.log
@item report
@item version
@end itemize
When you have everything working properly you can create an SSH key pair
and send the public key to the FATE server administrator who can be contacted
at the email address @email{fate-admin@@ffmpeg.org}.
Configure your SSH client to use public key authentication with that key
when connecting to the FATE server. Also do not forget to check the identity
of the server and to accept its host key. This can usually be achieved by
running your SSH client manually and killing it after you accepted the key.
The FATE server's fingerprint is:
@table @samp
@item RSA
d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
@item ECDSA
76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
@end table
If you have problems connecting to the FATE server, it may help to try out
the @command{ssh} command with one or more @option{-v} options. You should
get detailed output concerning your SSH configuration and the authentication
process.
The only thing left is to automate the execution of the fate.sh script and
the synchronisation of the samples directory.
@chapter Uploading new samples to the fate suite
If you need a sample uploaded send a mail to samples-request.
This is for developers who have an account on the fate suite server.
If you upload new samples, please make sure they are as small as possible,
space on each client, network bandwidth and so on benefit from smaller test cases.
Also keep in mind older checkouts use existing sample files, that means in
practice generally do not replace, remove or overwrite files as it likely would
break older checkouts or releases.
Also all needed samples for a commit should be uploaded, ideally 24
hours, before the push.
If you need an account for frequently uploading samples or you wish to help
others by doing that send a mail to ffmpeg-devel.
@example
#First update your local samples copy:
rsync -vauL --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X fate-suite.ffmpeg.org:/home/samples/fate-suite/ ~/fate-suite
#Then do a dry run checking what would be uploaded:
rsync -vanL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
#Upload the files:
rsync -vaL --no-g --chmod=Dg+s,Duo+x,ug+rw,o+r,o-w,+X ~/fate-suite/ fate-suite.ffmpeg.org:/home/samples/fate-suite
@end example
@chapter FATE makefile targets and variables
@section Makefile targets
@table @option
@item fate-rsync
Download/synchronize sample files to the configured samples directory.
@item fate-list
Will list all fate/regression test targets.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
@end table
@section Makefile variables
@table @env
@item V
Verbosity level, can be set to 0, 1 or 2.
@itemize
@item 0: show just the test arguments
@item 1: show just the command used in the test
@item 2: show everything
@end itemize
@item SAMPLES
Specify or override the path to the FATE samples at make time, it has a
meaning only while running the regression tests.
@item THREADS
Specify how many threads to use while running regression tests, it is
quite useful to detect thread-related regressions.
@item THREAD_TYPE
Specify which threading strategy test, either @samp{slice} or @samp{frame},
by default @samp{slice+frame}
@item CPUFLAGS
Specify CPU flags.
@item TARGET_EXEC
Specify or override the wrapper used to run the tests.
The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
@command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
through @command{ssh}.
@item GEN
Set to @samp{1} to generate the missing or mismatched references.
@item HWACCEL
Specify which hardware acceleration to use while running regression tests,
by default @samp{none} is used.
@item KEEP
Set to @samp{1} to keep temp files generated by fate test(s) when test is successful.
Default is @samp{0}, which removes these files. Files are always kept when a test
fails.
@end table
@section Examples
@example
make V=1 SAMPLES=/var/fate/samples THREADS=2 CPUFLAGS=mmx fate
@end example

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slot= # some unique identifier
repo=git://source.ffmpeg.org/ffmpeg.git # the source repository
#branch=release/2.6 # the branch to test
samples= # path to samples directory
workdir= # directory in which to do all the work
#fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report
comment= # optional description
build_only= # set to "yes" for a compile-only instance that skips tests
ignore_tests=
# the following are optional and map to configure options
arch=
cpu=
cross_prefix=
as=
cc=
ld=
target_os=
sysroot=
target_exec=
target_path=
target_samples=
extra_cflags=
extra_ldflags=
extra_libs=
extra_conf= # extra configure options not covered above
#make= # name of GNU make if not 'make'
makeopts= # extra options passed to 'make'
#makeopts_fate= # extra options passed to 'make' when running tests,
# defaulting to makeopts above if this is not set
#tar= # command to create a tar archive from its arguments on stdout,
# defaults to 'tar c'

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Bitstream Filters Documentation
@titlepage
@center @titlefont{FFmpeg Bitstream Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the bitstream filters provided by the
libavcodec library.
A bitstream filter operates on the encoded stream data, and performs
bitstream level modifications without performing decoding.
@c man end DESCRIPTION
@include bitstream_filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-bitstream-filters
@settitle FFmpeg bitstream filters
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Codecs Documentation
@titlepage
@center @titlefont{FFmpeg Codecs Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the codecs (decoders and encoders) provided by
the libavcodec library.
@c man end DESCRIPTION
@include codecs.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-codecs
@settitle FFmpeg codecs
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Devices Documentation
@titlepage
@center @titlefont{FFmpeg Devices Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the input and output devices provided by the
libavdevice library.
@c man end DESCRIPTION
@include devices.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavdevice(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-devices
@settitle FFmpeg devices
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Filters Documentation
@titlepage
@center @titlefont{FFmpeg Filters Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes filters, sources, and sinks provided by the
libavfilter library.
@c man end DESCRIPTION
@include filters.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavfilter.html,libavfilter}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavfilter(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-filters
@settitle FFmpeg filters
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Formats Documentation
@titlepage
@center @titlefont{FFmpeg Formats Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the supported formats (muxers and demuxers)
provided by the libavformat library.
@c man end DESCRIPTION
@include formats.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-formats
@settitle FFmpeg formats
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Protocols Documentation
@titlepage
@center @titlefont{FFmpeg Protocols Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes the input and output protocols provided by the
libavformat library.
@c man end DESCRIPTION
@include protocols.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-protocols
@settitle FFmpeg protocols
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Resampler Documentation
@titlepage
@center @titlefont{FFmpeg Resampler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin 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.
@c man end DESCRIPTION
@include resampler.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libswresample.html,libswresample}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswresample(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-resampler
@settitle FFmpeg Resampler
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Scaler Documentation
@titlepage
@center @titlefont{FFmpeg Scaler Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin 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.
@c man end DESCRIPTION
@include scaler.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libswscale.html,libswscale}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libswscale(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-scaler
@settitle FFmpeg video scaling and pixel format converter
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Utilities Documentation
@titlepage
@center @titlefont{FFmpeg Utilities Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
This document describes some generic features and utilities provided
by the libavutil library.
@c man end DESCRIPTION
@include utils.texi
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1), libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffmpeg-utils
@settitle FFmpeg utilities
@end ignore
@bye

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

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffplay Documentation
@titlepage
@center @titlefont{ffplay Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffplay [@var{options}] [@file{input_url}]
@chapter Description
@c man begin DESCRIPTION
FFplay is a very simple and portable media player using the FFmpeg
libraries and the SDL library. It is mostly used as a testbed for the
various FFmpeg APIs.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -x @var{width}
Force displayed width.
@item -y @var{height}
Force displayed height.
@item -s @var{size}
Set frame size (WxH or abbreviation), needed for videos which do
not contain a header with the frame size like raw YUV. This option
has been deprecated in favor of private options, try -video_size.
@item -fs
Start in fullscreen mode.
@item -an
Disable audio.
@item -vn
Disable video.
@item -sn
Disable subtitles.
@item -ss @var{pos}
Seek to @var{pos}. Note that in most formats it is not possible to seek
exactly, so @command{ffplay} will seek to the nearest seek point to
@var{pos}.
@var{pos} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -t @var{duration}
Play @var{duration} seconds of audio/video.
@var{duration} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
@item -bytes
Seek by bytes.
@item -seek_interval
Set custom interval, in seconds, for seeking using left/right keys. Default is 10 seconds.
@item -nodisp
Disable graphical display.
@item -noborder
Borderless window.
@item -alwaysontop
Window always on top. Available on: X11 with SDL >= 2.0.5, Windows SDL >= 2.0.6.
@item -volume
Set the startup volume. 0 means silence, 100 means no volume reduction or
amplification. Negative values are treated as 0, values above 100 are treated
as 100.
@item -f @var{fmt}
Force format.
@item -window_title @var{title}
Set window title (default is the input filename).
@item -left @var{title}
Set the x position for the left of the window (default is a centered window).
@item -top @var{title}
Set the y position for the top of the window (default is a centered window).
@item -loop @var{number}
Loops movie playback <number> times. 0 means forever.
@item -showmode @var{mode}
Set the show mode to use.
Available values for @var{mode} are:
@table @samp
@item 0, video
show video
@item 1, waves
show audio waves
@item 2, rdft
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
@end table
Default value is "video", if video is not present or cannot be played
"rdft" is automatically selected.
You can interactively cycle through the available show modes by
pressing the key @key{w}.
@item -vf @var{filtergraph}
Create the filtergraph specified by @var{filtergraph} and use it to
filter the video stream.
@var{filtergraph} is a description of the filtergraph to apply to
the stream, and must have a single video input and a single video
output. In the filtergraph, the input is associated to the label
@code{in}, and the output to the label @code{out}. See the
ffmpeg-filters manual for more information about the filtergraph
syntax.
You can specify this parameter multiple times and cycle through the specified
filtergraphs along with the show modes by pressing the key @key{w}.
@item -af @var{filtergraph}
@var{filtergraph} is a description of the filtergraph to apply to
the input audio.
Use the option "-filters" to show all the available filters (including
sources and sinks).
@item -i @var{input_url}
Read @var{input_url}.
@end table
@section Advanced options
@table @option
@item -pix_fmt @var{format}
Set pixel format.
This option has been deprecated in favor of private options, try -pixel_format.
@item -stats
Print several playback statistics, in particular show the stream
duration, the codec parameters, the current position in the stream and
the audio/video synchronisation drift. It is shown by default, unless the
log level is lower than @code{info}. Its display can be forced by manually
specifying this option. To disable it, you need to specify @code{-nostats}.
@item -fast
Non-spec-compliant optimizations.
@item -genpts
Generate pts.
@item -sync @var{type}
Set the master clock to audio (@code{type=audio}), video
(@code{type=video}) or external (@code{type=ext}). Default is audio. The
master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
@item -ast @var{audio_stream_specifier}
Select the desired audio stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" audio stream is selected in the program of the
already selected video stream.
@item -vst @var{video_stream_specifier}
Select the desired video stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" video stream is selected.
@item -sst @var{subtitle_stream_specifier}
Select the desired subtitle stream using the given stream specifier. The stream
specifiers are described in the @ref{Stream specifiers} chapter. If this option
is not specified, the "best" subtitle stream is selected in the program of the
already selected video or audio stream.
@item -autoexit
Exit when video is done playing.
@item -exitonkeydown
Exit if any key is pressed.
@item -exitonmousedown
Exit if any mouse button is pressed.
@item -codec:@var{media_specifier} @var{codec_name}
Force a specific decoder implementation for the stream identified by
@var{media_specifier}, which can assume the values @code{a} (audio),
@code{v} (video), and @code{s} subtitle.
@item -acodec @var{codec_name}
Force a specific audio decoder.
@item -vcodec @var{codec_name}
Force a specific video decoder.
@item -scodec @var{codec_name}
Force a specific subtitle decoder.
@item -autorotate
Automatically rotate the video according to file metadata. Enabled by
default, use @option{-noautorotate} to disable it.
@item -framedrop
Drop video frames if video is out of sync. Enabled by default if the master
clock is not set to video. Use this option to enable frame dropping for all
master clock sources, use @option{-noframedrop} to disable it.
@item -infbuf
Do not limit the input buffer size, read as much data as possible from the
input as soon as possible. Enabled by default for realtime streams, where data
may be dropped if not read in time. Use this option to enable infinite buffers
for all inputs, use @option{-noinfbuf} to disable it.
@item -filter_threads @var{nb_threads}
Defines how many threads are used to process a filter pipeline. Each pipeline
will produce a thread pool with this many threads available for parallel
processing. The default is 0 which means that the thread count will be
determined by the number of available CPUs.
@end table
@section While playing
@table @key
@item q, ESC
Quit.
@item f
Toggle full screen.
@item p, SPC
Pause.
@item m
Toggle mute.
@item 9, 0
Decrease and increase volume respectively.
@item /, *
Decrease and increase volume respectively.
@item a
Cycle audio channel in the current program.
@item v
Cycle video channel.
@item t
Cycle subtitle channel in the current program.
@item c
Cycle program.
@item w
Cycle video filters or show modes.
@item s
Step to the next frame.
Pause if the stream is not already paused, step to the next video
frame, and pause.
@item left/right
Seek backward/forward 10 seconds.
@item down/up
Seek backward/forward 1 minute.
@item page down/page up
Seek to the previous/next chapter.
or if there are no chapters
Seek backward/forward 10 minutes.
@item right mouse click
Seek to percentage in file corresponding to fraction of width.
@item left mouse double-click
Toggle full screen.
@end table
@c man end
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffplay.html,ffplay},
@end ifset
@ifset config-not-all
@url{ffplay-all.html,ffmpeg-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffplay(1),
@end ifset
@ifset config-not-all
ffplay-all(1),
@end ifset
ffmpeg(1), ffprobe(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffplay
@settitle FFplay media player
@end ignore
@bye

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@@ -0,0 +1,690 @@
\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle ffprobe Documentation
@titlepage
@center @titlefont{ffprobe Documentation}
@end titlepage
@top
@contents
@chapter Synopsis
ffprobe [@var{options}] [@file{input_url}]
@chapter Description
@c man begin DESCRIPTION
ffprobe gathers information from multimedia streams and prints it in
human- and machine-readable fashion.
For example it can be used to check the format of the container used
by a multimedia stream and the format and type of each media stream
contained in it.
If a 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.
ffprobe may be employed both as a standalone application or in
combination with a textual filter, which may perform more
sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by ffprobe or
for specifying which information to display, and for setting how
ffprobe will show it.
ffprobe output is designed to be easily parsable by a textual filter,
and consists of one or more sections of a form defined by the selected
writer, which is specified by the @option{print_format} option.
Sections may contain other nested sections, and are identified by a
name (which may be shared by other sections), and an unique
name. See the output of @option{sections}.
Metadata tags stored in the container or in the streams are recognized
and printed in the corresponding "FORMAT", "STREAM" or "PROGRAM_STREAM"
section.
@c man end
@chapter Options
@c man begin OPTIONS
@include fftools-common-opts.texi
@section Main options
@table @option
@item -f @var{format}
Force format to use.
@item -unit
Show the unit of the displayed values.
@item -prefix
Use SI prefixes for the displayed values.
Unless the "-byte_binary_prefix" option is used all the prefixes
are decimal.
@item -byte_binary_prefix
Force the use of binary prefixes for byte values.
@item -sexagesimal
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
@item -pretty
Prettify the format of the displayed values, it corresponds to the
options "-unit -prefix -byte_binary_prefix -sexagesimal".
@item -of, -print_format @var{writer_name}[=@var{writer_options}]
Set the output printing format.
@var{writer_name} specifies the name of the writer, and
@var{writer_options} specifies the options to be passed to the writer.
For example for printing the output in JSON format, specify:
@example
-print_format json
@end example
For more details on the available output printing formats, see the
Writers section below.
@item -sections
Print sections structure and section information, and exit. The output
is not meant to be parsed by a machine.
@item -select_streams @var{stream_specifier}
Select only the streams specified by @var{stream_specifier}. This
option affects only the options related to streams
(e.g. @code{show_streams}, @code{show_packets}, etc.).
For example to show only audio streams, you can use the command:
@example
ffprobe -show_streams -select_streams a INPUT
@end example
To show only video packets belonging to the video stream with index 1:
@example
ffprobe -show_packets -select_streams v:1 INPUT
@end example
@item -show_data
Show payload data, as a hexadecimal and ASCII dump. Coupled with
@option{-show_packets}, it will dump the packets' data. Coupled with
@option{-show_streams}, it will dump the codec extradata.
The dump is printed as the "data" field. It may contain newlines.
@item -show_data_hash @var{algorithm}
Show a hash of payload data, for packets with @option{-show_packets} and for
codec extradata with @option{-show_streams}.
@item -show_error
Show information about the error found when trying to probe the input.
The error information is printed within a section with name "ERROR".
@item -show_format
Show information about the container format of the input multimedia
stream.
All the container format information is printed within a section with
name "FORMAT".
@item -show_format_entry @var{name}
Like @option{-show_format}, but only prints the specified entry of the
container format information, rather than all. This option may be given more
than once, then all specified entries will be shown.
This option is deprecated, use @code{show_entries} instead.
@item -show_entries @var{section_entries}
Set list of entries to show.
Entries are specified according to the following
syntax. @var{section_entries} contains a list of section entries
separated by @code{:}. Each section entry is composed by a section
name (or unique name), optionally followed by a list of entries local
to that section, separated by @code{,}.
If section name is specified but is followed by no @code{=}, all
entries are printed to output, together with all the contained
sections. Otherwise only the entries specified in the local section
entries list are printed. In particular, if @code{=} is specified but
the list of local entries is empty, then no entries will be shown for
that section.
Note that the order of specification of the local section entries is
not honored in the output, and the usual display order will be
retained.
The formal syntax is given by:
@example
@var{LOCAL_SECTION_ENTRIES} ::= @var{SECTION_ENTRY_NAME}[,@var{LOCAL_SECTION_ENTRIES}]
@var{SECTION_ENTRY} ::= @var{SECTION_NAME}[=[@var{LOCAL_SECTION_ENTRIES}]]
@var{SECTION_ENTRIES} ::= @var{SECTION_ENTRY}[:@var{SECTION_ENTRIES}]
@end example
For example, to show only the index and type of each stream, and the PTS
time, duration time, and stream index of the packets, you can specify
the argument:
@example
packet=pts_time,duration_time,stream_index : stream=index,codec_type
@end example
To show all the entries in the section "format", but only the codec
type in the section "stream", specify the argument:
@example
format : stream=codec_type
@end example
To show all the tags in the stream and format sections:
@example
stream_tags : format_tags
@end example
To show only the @code{title} tag (if available) in the stream
sections:
@example
stream_tags=title
@end example
@item -show_packets
Show information about each packet contained in the input multimedia
stream.
The information for each single packet is printed within a dedicated
section with name "PACKET".
@item -show_frames
Show information about each frame 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".
@item -show_log @var{loglevel}
Show logging information from the decoder about each frame according to
the value set in @var{loglevel}, (see @code{-loglevel}). This option requires @code{-show_frames}.
The information for each log message is printed within a dedicated
section with name "LOG".
@item -show_streams
Show information about each media stream contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "STREAM".
@item -show_programs
Show information about programs and their streams contained in the input
multimedia stream.
Each media stream information is printed within a dedicated section
with name "PROGRAM_STREAM".
@item -show_chapters
Show information about chapters stored in the format.
Each chapter is printed within a dedicated section with name "CHAPTER".
@item -count_frames
Count the number of frames per stream and report it in the
corresponding stream section.
@item -count_packets
Count the number of packets per stream and report it in the
corresponding stream section.
@item -read_intervals @var{read_intervals}
Read only the specified intervals. @var{read_intervals} must be a
sequence of interval specifications separated by ",".
@command{ffprobe} will seek to the interval starting point, and will
continue reading from that.
Each interval is specified by two optional parts, separated by "%".
The first part specifies the interval start position. It is
interpreted as an 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:
@example
@var{INTERVAL} ::= [@var{START}|+@var{START_OFFSET}][%[@var{END}|+@var{END_OFFSET}]]
@var{INTERVALS} ::= @var{INTERVAL}[,@var{INTERVALS}]
@end example
A few examples follow.
@itemize
@item
Seek to time 10, read packets until 20 seconds after the found seek
point, then seek to position @code{01:30} (1 minute and thirty
seconds) and read packets until position @code{01:45}.
@example
10%+20,01:30%01:45
@end example
@item
Read only 42 packets after seeking to position @code{01:23}:
@example
01:23%+#42
@end example
@item
Read only the first 20 seconds from the start:
@example
%+20
@end example
@item
Read from the start until position @code{02:30}:
@example
%02:30
@end example
@end itemize
@item -show_private_data, -private
Show private data, that is data depending on the format of the
particular shown element.
This option is enabled by default, but you may need to disable it
for specific uses, for example when creating XSD-compliant XML output.
@item -show_program_version
Show information related to program version.
Version information is printed within a section with name
"PROGRAM_VERSION".
@item -show_library_versions
Show information related to library versions.
Version information for each library is printed within a section with
name "LIBRARY_VERSION".
@item -show_versions
Show information related to program and library versions. This is the
equivalent of setting both @option{-show_program_version} and
@option{-show_library_versions} options.
@item -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".
@item -bitexact
Force bitexact output, useful to produce output which is not dependent
on the specific build.
@item -i @var{input_url}
Read @var{input_url}.
@end table
@c man end
@chapter Writers
@c man begin WRITERS
A writer defines the output format adopted by @command{ffprobe}, and will be
used for printing all the parts of the output.
A writer may accept one or more arguments, which specify the options
to adopt. The options are specified as a list of @var{key}=@var{value}
pairs, separated by ":".
All writers support the following options:
@table @option
@item string_validation, sv
Set string validation mode.
The following values are accepted.
@table @samp
@item 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.
@item ignore
Any validation error will be ignored. This will result in possibly
broken output, especially with the json or xml writer.
@item replace
The writer will substitute invalid UTF-8 sequences or code points with
the string specified with the @option{string_validation_replacement}.
@end table
Default value is @samp{replace}.
@item string_validation_replacement, svr
Set replacement string to use in case @option{string_validation} is
set to @samp{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.
@end table
A description of the currently available writers follows.
@section default
Default format.
Print each section in the form:
@example
[SECTION]
key1=val1
...
keyN=valN
[/SECTION]
@end example
Metadata tags are printed as a line in the corresponding FORMAT, STREAM or
PROGRAM_STREAM section, and are prefixed by the string "TAG:".
A description of the accepted options follows.
@table @option
@item nokey, nk
If set to 1 specify not to print the key of each field. Default value
is 0.
@item noprint_wrappers, nw
If set to 1 specify not to print the section header and footer.
Default value is 0.
@end table
@section compact, csv
Compact and CSV format.
The @code{csv} writer is equivalent to @code{compact}, but supports
different defaults.
Each section is printed on a single line.
If no option is specified, the output has the form:
@example
section|key1=val1| ... |keyN=valN
@end example
Metadata tags are printed in the corresponding "format" or "stream"
section. A metadata tag key, if printed, is prefixed by the string
"tag:".
The description of the accepted options follows.
@table @option
@item item_sep, s
Specify the character to use for separating fields in the output line.
It must be a single printable character, it is "|" by default ("," for
the @code{csv} writer).
@item nokey, nk
If set to 1 specify not to print the key of each field. Its default
value is 0 (1 for the @code{csv} writer).
@item escape, e
Set the escape mode to use, default to "c" ("csv" for the @code{csv}
writer).
It can assume one of the following values:
@table @option
@item c
Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
character (@samp{\}) or the item separator character @var{SEP} are escaped
using C-like fashioned escaping, so that a newline is converted to the
sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
the separator @var{SEP} is converted to @samp{\@var{SEP}}.
@item csv
Perform CSV-like escaping, as described in RFC4180. Strings
containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
(@samp{"}), or @var{SEP} are enclosed in double-quotes.
@item none
Perform no escaping.
@end table
@item print_section, p
Print the section name at the beginning of each line if the value is
@code{1}, disable it with value set to @code{0}. Default value is
@code{1}.
@end table
@section flat
Flat format.
A free-form output where each line contains an explicit key=value, such as
"streams.stream.3.tags.foo=bar". The output is shell escaped, so it can be
directly embedded in sh scripts as long as the separator character is an
alphanumeric character or an underscore (see @var{sep_char} option).
The description of the accepted options follows.
@table @option
@item sep_char, s
Separator character used to separate the chapter, the section name, IDs and
potential tags in the printed field key.
Default value is @samp{.}.
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
@section ini
INI format output.
Print output in an INI based format.
The following conventions are adopted:
@itemize
@item
all key and values are UTF-8
@item
@samp{.} is the subgroup separator
@item
newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
escaped
@item
@samp{\} is the escape character
@item
@samp{#} is the comment indicator
@item
@samp{=} is the key/value separator
@item
@samp{:} is not used but usually parsed as key/value separator
@end itemize
This writer accepts options as a list of @var{key}=@var{value} pairs,
separated by @samp{:}.
The description of the accepted options follows.
@table @option
@item hierarchical, h
Specify if the section name specification should be hierarchical. If
set to 1, and if there is more than one section in the current
chapter, the section name will be prefixed by the name of the
chapter. A value of 0 will disable this behavior.
Default value is 1.
@end table
@section json
JSON based format.
Each section is printed using JSON notation.
The description of the accepted options follows.
@table @option
@item compact, c
If set to 1 enable compact output, that is each section will be
printed on a single line. Default value is 0.
@end table
For more information about JSON, see @url{http://www.json.org/}.
@section xml
XML based format.
The XML output is described in the XML schema description file
@file{ffprobe.xsd} installed in the FFmpeg datadir.
An updated version of the schema can be retrieved at the url
@url{http://www.ffmpeg.org/schema/ffprobe.xsd}, which redirects to the
latest schema committed into the FFmpeg development source code tree.
Note that the output issued will be compliant to the
@file{ffprobe.xsd} schema only when no special global output options
(@option{unit}, @option{prefix}, @option{byte_binary_prefix},
@option{sexagesimal} etc.) are specified.
The description of the accepted options follows.
@table @option
@item fully_qualified, q
If set to 1 specify if the output should be fully qualified. Default
value is 0.
This is required for generating an XML file which can be validated
through an XSD file.
@item xsd_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 @option{fully_qualified} to 1.
@end table
For more information about the XML format, see
@url{https://www.w3.org/XML/}.
@c man end WRITERS
@chapter Timecode
@c man begin TIMECODE
@command{ffprobe} supports Timecode extraction:
@itemize
@item
MPEG1/2 timecode is extracted from the GOP, and is available in the video
stream details (@option{-show_streams}, see @var{timecode}).
@item
MOV timecode is extracted from tmcd track, so is available in the tmcd
stream metadata (@option{-show_streams}, see @var{TAG:timecode}).
@item
DV, GXF and AVI timecodes are available in format metadata
(@option{-show_format}, see @var{TAG:timecode}).
@end itemize
@c man end TIMECODE
@include config.texi
@ifset config-all
@set config-readonly
@ifset config-avutil
@include utils.texi
@end ifset
@ifset config-avcodec
@include codecs.texi
@include bitstream_filters.texi
@end ifset
@ifset config-avformat
@include formats.texi
@include protocols.texi
@end ifset
@ifset config-avdevice
@include devices.texi
@end ifset
@ifset config-swresample
@include resampler.texi
@end ifset
@ifset config-swscale
@include scaler.texi
@end ifset
@ifset config-avfilter
@include filters.texi
@end ifset
@end ifset
@chapter See Also
@ifhtml
@ifset config-all
@url{ffprobe.html,ffprobe},
@end ifset
@ifset config-not-all
@url{ffprobe-all.html,ffprobe-all},
@end ifset
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay},
@url{ffmpeg-utils.html,ffmpeg-utils},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{ffmpeg-resampler.html,ffmpeg-resampler},
@url{ffmpeg-codecs.html,ffmpeg-codecs},
@url{ffmpeg-bitstream-filters.html,ffmpeg-bitstream-filters},
@url{ffmpeg-formats.html,ffmpeg-formats},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{ffmpeg-filters.html,ffmpeg-filters}
@end ifhtml
@ifnothtml
@ifset config-all
ffprobe(1),
@end ifset
@ifset config-not-all
ffprobe-all(1),
@end ifset
ffmpeg(1), ffplay(1),
ffmpeg-utils(1), ffmpeg-scaler(1), ffmpeg-resampler(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename ffprobe
@settitle ffprobe media prober
@end ignore
@bye

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<?xml version="1.0" encoding="UTF-8"?>
<xsd:schema xmlns:xsd="http://www.w3.org/2001/XMLSchema"
targetNamespace="http://www.ffmpeg.org/schema/ffprobe"
xmlns:ffprobe="http://www.ffmpeg.org/schema/ffprobe">
<xsd:element name="ffprobe" type="ffprobe:ffprobeType"/>
<xsd:complexType name="ffprobeType">
<xsd:sequence>
<xsd:element name="program_version" type="ffprobe:programVersionType" minOccurs="0" maxOccurs="1" />
<xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="pixel_formats" type="ffprobe:pixelFormatsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets" type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="frames" type="ffprobe:framesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
<xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
<xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
<xsd:element name="format" type="ffprobe:formatType" minOccurs="0" maxOccurs="1" />
<xsd:element name="error" type="ffprobe:errorType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsType">
<xsd:sequence>
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="framesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetsAndFramesType">
<xsd:sequence>
<xsd:choice minOccurs="0" maxOccurs="unbounded">
<xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:choice>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="codec_type" type="xsd:string" use="required" />
<xsd:attribute name="stream_index" type="xsd:int" use="required" />
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float" />
<xsd:attribute name="dts" type="xsd:long" />
<xsd:attribute name="dts_time" type="xsd:float" />
<xsd:attribute name="duration" type="xsd:long" />
<xsd:attribute name="duration_time" type="xsd:float" />
<xsd:attribute name="convergence_duration" type="xsd:long" />
<xsd:attribute name="convergence_duration_time" type="xsd:float" />
<xsd:attribute name="size" type="xsd:long" use="required" />
<xsd:attribute name="pos" type="xsd:long" />
<xsd:attribute name="flags" type="xsd:string" use="required" />
<xsd:attribute name="data" type="xsd:string" />
<xsd:attribute name="data_hash" type="xsd:string" />
</xsd:complexType>
<xsd:complexType name="packetSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="packetSideDataType">
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="frameType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="logs" type="ffprobe:logsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="side_data_list" type="ffprobe:frameSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="media_type" type="xsd:string" use="required"/>
<xsd:attribute name="stream_index" type="xsd:int" />
<xsd:attribute name="key_frame" type="xsd:int" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_pts" type="xsd:long" />
<xsd:attribute name="pkt_pts_time" type="xsd:float"/>
<xsd:attribute name="pkt_dts" type="xsd:long" />
<xsd:attribute name="pkt_dts_time" type="xsd:float"/>
<xsd:attribute name="best_effort_timestamp" type="xsd:long" />
<xsd:attribute name="best_effort_timestamp_time" type="xsd:float" />
<xsd:attribute name="pkt_duration" type="xsd:long" />
<xsd:attribute name="pkt_duration_time" type="xsd:float"/>
<xsd:attribute name="pkt_pos" type="xsd:long" />
<xsd:attribute name="pkt_size" type="xsd:int" />
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="nb_samples" type="xsd:long" />
<xsd:attribute name="channels" type="xsd:int" />
<xsd:attribute name="channel_layout" type="xsd:string"/>
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:long" />
<xsd:attribute name="height" type="xsd:long" />
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pict_type" type="xsd:string"/>
<xsd:attribute name="coded_picture_number" type="xsd:long" />
<xsd:attribute name="display_picture_number" type="xsd:long" />
<xsd:attribute name="interlaced_frame" type="xsd:int" />
<xsd:attribute name="top_field_first" type="xsd:int" />
<xsd:attribute name="repeat_pict" type="xsd:int" />
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="logsType">
<xsd:sequence>
<xsd:element name="log" type="ffprobe:logType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="logType">
<xsd:attribute name="context" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int" />
<xsd:attribute name="category" type="xsd:int" />
<xsd:attribute name="parent_context" type="xsd:string"/>
<xsd:attribute name="parent_category" type="xsd:int" />
<xsd:attribute name="message" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataListType">
<xsd:sequence>
<xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataType">
<xsd:sequence>
<xsd:element name="timecodes" type="ffprobe:frameSideDataTimecodeList" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="side_data_type" type="xsd:string"/>
<xsd:attribute name="side_data_size" type="xsd:int" />
<xsd:attribute name="timecode" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeList">
<xsd:sequence>
<xsd:element name="timecode" type="ffprobe:frameSideDataTimecodeType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="frameSideDataTimecodeType">
<xsd:attribute name="value" type="xsd:string"/>
</xsd:complexType>
<xsd:complexType name="subtitleType">
<xsd:attribute name="media_type" type="xsd:string" fixed="subtitle" use="required"/>
<xsd:attribute name="pts" type="xsd:long" />
<xsd:attribute name="pts_time" type="xsd:float"/>
<xsd:attribute name="format" type="xsd:int" />
<xsd:attribute name="start_display_time" type="xsd:int" />
<xsd:attribute name="end_display_time" type="xsd:int" />
<xsd:attribute name="num_rects" type="xsd:int" />
</xsd:complexType>
<xsd:complexType name="streamsType">
<xsd:sequence>
<xsd:element name="stream" type="ffprobe:streamType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="programsType">
<xsd:sequence>
<xsd:element name="program" type="ffprobe:programType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="streamDispositionType">
<xsd:attribute name="default" type="xsd:int" use="required" />
<xsd:attribute name="dub" type="xsd:int" use="required" />
<xsd:attribute name="original" type="xsd:int" use="required" />
<xsd:attribute name="comment" type="xsd:int" use="required" />
<xsd:attribute name="lyrics" type="xsd:int" use="required" />
<xsd:attribute name="karaoke" type="xsd:int" use="required" />
<xsd:attribute name="forced" type="xsd:int" use="required" />
<xsd:attribute name="hearing_impaired" type="xsd:int" use="required" />
<xsd:attribute name="visual_impaired" type="xsd:int" use="required" />
<xsd:attribute name="clean_effects" type="xsd:int" use="required" />
<xsd:attribute name="attached_pic" type="xsd:int" use="required" />
<xsd:attribute name="timed_thumbnails" type="xsd:int" use="required" />
</xsd:complexType>
<xsd:complexType name="streamType">
<xsd:sequence>
<xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="side_data_list" type="ffprobe:packetSideDataListType" minOccurs="0" maxOccurs="1" />
</xsd:sequence>
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="codec_name" type="xsd:string" />
<xsd:attribute name="codec_long_name" type="xsd:string" />
<xsd:attribute name="profile" type="xsd:string" />
<xsd:attribute name="codec_type" type="xsd:string" />
<xsd:attribute name="codec_time_base" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag" type="xsd:string" use="required"/>
<xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
<xsd:attribute name="extradata" type="xsd:string" />
<xsd:attribute name="extradata_hash" type="xsd:string" />
<!-- video attributes -->
<xsd:attribute name="width" type="xsd:int"/>
<xsd:attribute name="height" type="xsd:int"/>
<xsd:attribute name="coded_width" type="xsd:int"/>
<xsd:attribute name="coded_height" type="xsd:int"/>
<xsd:attribute name="closed_captions" type="xsd:boolean"/>
<xsd:attribute name="has_b_frames" type="xsd:int"/>
<xsd:attribute name="sample_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
<xsd:attribute name="pix_fmt" type="xsd:string"/>
<xsd:attribute name="level" type="xsd:int"/>
<xsd:attribute name="color_range" type="xsd:string"/>
<xsd:attribute name="color_space" type="xsd:string"/>
<xsd:attribute name="color_transfer" type="xsd:string"/>
<xsd:attribute name="color_primaries" type="xsd:string"/>
<xsd:attribute name="chroma_location" type="xsd:string"/>
<xsd:attribute name="field_order" type="xsd:string"/>
<xsd:attribute name="timecode" type="xsd:string"/>
<xsd:attribute name="refs" type="xsd:int"/>
<!-- audio attributes -->
<xsd:attribute name="sample_fmt" type="xsd:string"/>
<xsd:attribute name="sample_rate" type="xsd:int"/>
<xsd:attribute name="channels" type="xsd:int"/>
<xsd:attribute name="channel_layout" type="xsd:string"/>
<xsd:attribute name="bits_per_sample" type="xsd:int"/>
<xsd:attribute name="id" type="xsd:string"/>
<xsd:attribute name="r_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="avg_frame_rate" type="xsd:string" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration_ts" type="xsd:long"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="bit_rate" type="xsd:int"/>
<xsd:attribute name="max_bit_rate" type="xsd:int"/>
<xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
<xsd:attribute name="nb_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_frames" type="xsd:int"/>
<xsd:attribute name="nb_read_packets" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="programType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
<xsd:element name="streams" type="ffprobe:streamsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="program_id" type="xsd:int" use="required"/>
<xsd:attribute name="program_num" type="xsd:int" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="start_pts" type="xsd:long"/>
<xsd:attribute name="end_time" type="xsd:float"/>
<xsd:attribute name="end_pts" type="xsd:long"/>
<xsd:attribute name="pmt_pid" type="xsd:int" use="required"/>
<xsd:attribute name="pcr_pid" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="formatType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="filename" type="xsd:string" use="required"/>
<xsd:attribute name="nb_streams" type="xsd:int" use="required"/>
<xsd:attribute name="nb_programs" type="xsd:int" use="required"/>
<xsd:attribute name="format_name" type="xsd:string" use="required"/>
<xsd:attribute name="format_long_name" type="xsd:string"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="duration" type="xsd:float"/>
<xsd:attribute name="size" type="xsd:long"/>
<xsd:attribute name="bit_rate" type="xsd:long"/>
<xsd:attribute name="probe_score" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="tagType">
<xsd:attribute name="key" type="xsd:string" use="required"/>
<xsd:attribute name="value" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="errorType">
<xsd:attribute name="code" type="xsd:int" use="required"/>
<xsd:attribute name="string" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="programVersionType">
<xsd:attribute name="version" type="xsd:string" use="required"/>
<xsd:attribute name="copyright" type="xsd:string" use="required"/>
<xsd:attribute name="build_date" type="xsd:string"/>
<xsd:attribute name="build_time" type="xsd:string"/>
<xsd:attribute name="compiler_ident" type="xsd:string" use="required"/>
<xsd:attribute name="configuration" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="chaptersType">
<xsd:sequence>
<xsd:element name="chapter" type="ffprobe:chapterType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="chapterType">
<xsd:sequence>
<xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
<xsd:attribute name="id" type="xsd:int" use="required"/>
<xsd:attribute name="time_base" type="xsd:string" use="required"/>
<xsd:attribute name="start" type="xsd:int" use="required"/>
<xsd:attribute name="start_time" type="xsd:float"/>
<xsd:attribute name="end" type="xsd:int" use="required"/>
<xsd:attribute name="end_time" type="xsd:float" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionType">
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="major" type="xsd:int" use="required"/>
<xsd:attribute name="minor" type="xsd:int" use="required"/>
<xsd:attribute name="micro" type="xsd:int" use="required"/>
<xsd:attribute name="version" type="xsd:int" use="required"/>
<xsd:attribute name="ident" type="xsd:string" use="required"/>
</xsd:complexType>
<xsd:complexType name="libraryVersionsType">
<xsd:sequence>
<xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatFlagsType">
<xsd:attribute name="big_endian" type="xsd:int" use="required"/>
<xsd:attribute name="palette" type="xsd:int" use="required"/>
<xsd:attribute name="bitstream" type="xsd:int" use="required"/>
<xsd:attribute name="hwaccel" type="xsd:int" use="required"/>
<xsd:attribute name="planar" type="xsd:int" use="required"/>
<xsd:attribute name="rgb" type="xsd:int" use="required"/>
<xsd:attribute name="pseudopal" type="xsd:int" use="required"/>
<xsd:attribute name="alpha" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentType">
<xsd:attribute name="index" type="xsd:int" use="required"/>
<xsd:attribute name="bit_depth" type="xsd:int" use="required"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatComponentsType">
<xsd:sequence>
<xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
<xsd:complexType name="pixelFormatType">
<xsd:sequence>
<xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
<xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
</xsd:sequence>
<xsd:attribute name="name" type="xsd:string" use="required"/>
<xsd:attribute name="nb_components" type="xsd:int" use="required"/>
<xsd:attribute name="log2_chroma_w" type="xsd:int"/>
<xsd:attribute name="log2_chroma_h" type="xsd:int"/>
<xsd:attribute name="bits_per_pixel" type="xsd:int"/>
</xsd:complexType>
<xsd:complexType name="pixelFormatsType">
<xsd:sequence>
<xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
</xsd:sequence>
</xsd:complexType>
</xsd:schema>

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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.
@anchor{Stream specifiers}
@section Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers
are used to precisely specify which stream(s) a given option belongs to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} contains the
@code{a:1} stream specifier, which matches the second audio stream. Therefore, it
would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, so that the option is applied to all
of them. E.g. the stream specifier in @code{-b:a 128k} matches all audio
streams.
An empty stream specifier matches all streams. For example, @code{-codec copy}
or @code{-codec: copy} would copy all the streams without reencoding.
Possible forms of stream specifiers are:
@table @option
@item @var{stream_index}
Matches the stream with this index. E.g. @code{-threads:1 4} would set the
thread count for the second stream to 4. If @var{stream_index} is used as an
additional stream specifier (see below), then it selects stream number
@var{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.
@item @var{stream_type}[:@var{additional_stream_specifier}]
@var{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 @var{additional_stream_specifier} is used, then
it matches streams which both have this type and match the
@var{additional_stream_specifier}. Otherwise, it matches all streams of the
specified type.
@item p:@var{program_id}[:@var{additional_stream_specifier}]
Matches streams which are in the program with the id @var{program_id}. If
@var{additional_stream_specifier} is used, then it matches streams which both
are part of the program and match the @var{additional_stream_specifier}.
@item #@var{stream_id} or i:@var{stream_id}
Match the stream by stream id (e.g. PID in MPEG-TS container).
@item m:@var{key}[:@var{value}]
Matches streams with the metadata tag @var{key} having the specified value. If
@var{value} is not given, matches streams that contain the given tag with any
value.
@item 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 @command{ffmpeg}, matching by metadata will only work properly for
input files.
@end table
@section Generic options
These options are shared amongst the ff* tools.
@table @option
@item -L
Show license.
@item -h, -?, -help, --help [@var{arg}]
Show help. An optional parameter may be specified to print help about a specific
item. If no argument is specified, only basic (non advanced) tool
options are shown.
Possible values of @var{arg} are:
@table @option
@item long
Print advanced tool options in addition to the basic tool options.
@item full
Print complete list of options, including shared and private options
for encoders, decoders, demuxers, muxers, filters, etc.
@item decoder=@var{decoder_name}
Print detailed information about the decoder named @var{decoder_name}. Use the
@option{-decoders} option to get a list of all decoders.
@item encoder=@var{encoder_name}
Print detailed information about the encoder named @var{encoder_name}. Use the
@option{-encoders} option to get a list of all encoders.
@item demuxer=@var{demuxer_name}
Print detailed information about the demuxer named @var{demuxer_name}. Use the
@option{-formats} option to get a list of all demuxers and muxers.
@item muxer=@var{muxer_name}
Print detailed information about the muxer named @var{muxer_name}. Use the
@option{-formats} option to get a list of all muxers and demuxers.
@item filter=@var{filter_name}
Print detailed information about the filter name @var{filter_name}. Use the
@option{-filters} option to get a list of all filters.
@item bsf=@var{bitstream_filter_name}
Print detailed information about the bitstream filter name @var{bitstream_filter_name}.
Use the @option{-bsfs} option to get a list of all bitstream filters.
@end table
@item -version
Show version.
@item -formats
Show available formats (including devices).
@item -demuxers
Show available demuxers.
@item -muxers
Show available muxers.
@item -devices
Show available devices.
@item -codecs
Show all codecs known to libavcodec.
Note that the term 'codec' is used throughout this documentation as a shortcut
for what is more correctly called a media bitstream format.
@item -decoders
Show available decoders.
@item -encoders
Show all available encoders.
@item -bsfs
Show available bitstream filters.
@item -protocols
Show available protocols.
@item -filters
Show available libavfilter filters.
@item -pix_fmts
Show available pixel formats.
@item -sample_fmts
Show available sample formats.
@item -layouts
Show channel names and standard channel layouts.
@item -colors
Show recognized color names.
@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{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.
@example
ffmpeg -sources pulse,server=192.168.0.4
@end example
@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{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.
@example
ffmpeg -sinks pulse,server=192.168.0.4
@end example
@item -loglevel [@var{flags}+]@var{loglevel} | -v [@var{flags}+]@var{loglevel}
Set logging level and flags used by the library.
The optional @var{flags} prefix can consist of the following values:
@table @samp
@item 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.
@item level
Indicates that log output should add a @code{[level]} prefix to each message
line. This can be used as an alternative to log coloring, e.g. when dumping the
log to file.
@end table
Flags can also be used alone by adding a '+'/'-' prefix to set/reset a single
flag without affecting other @var{flags} or changing @var{loglevel}. When
setting both @var{flags} and @var{loglevel}, a '+' separator is expected
between the last @var{flags} value and before @var{loglevel}.
@var{loglevel} is a string or a number containing one of the following values:
@table @samp
@item quiet, -8
Show nothing at all; be silent.
@item 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.
@item fatal, 8
Only show fatal errors. These are errors after which the process absolutely
cannot continue.
@item error, 16
Show all errors, including ones which can be recovered from.
@item warning, 24
Show all warnings and errors. Any message related to possibly
incorrect or unexpected events will be shown.
@item info, 32
Show informative messages during processing. This is in addition to
warnings and errors. This is the default value.
@item verbose, 40
Same as @code{info}, except more verbose.
@item debug, 48
Show everything, including debugging information.
@item trace, 56
@end table
For example to enable repeated log output, add the @code{level} prefix, and set
@var{loglevel} to @code{verbose}:
@example
ffmpeg -loglevel repeat+level+verbose -i input output
@end example
Another example that enables repeated log output without affecting current
state of @code{level} prefix flag or @var{loglevel}:
@example
ffmpeg [...] -loglevel +repeat
@end example
By default the program logs to stderr. If coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
@env{AV_LOG_FORCE_NOCOLOR}, or can be forced setting
the environment variable @env{AV_LOG_FORCE_COLOR}.
@item -report
Dump full command line and log output to a file named
@code{@var{program}-@var{YYYYMMDD}-@var{HHMMSS}.log} in the current
directory.
This file can be useful for bug reports.
It also implies @code{-loglevel debug}.
Setting the environment variable @env{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:
@table @option
@item file
set the file name to use for the report; @code{%p} is expanded to the name
of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
to a plain @code{%}
@item level
set the log verbosity level using a numerical value (see @code{-loglevel}).
@end table
For example, to output a report to a file named @file{ffreport.log}
using a log level of @code{32} (alias for log level @code{info}):
@example
FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
@end example
Errors in parsing the environment variable are not fatal, and will not
appear in the report.
@item -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.
@item -cpuflags flags (@emph{global})
Allows setting and clearing cpu flags. This option is intended
for testing. Do not use it unless you know what you're doing.
@example
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
@end example
Possible flags for this option are:
@table @samp
@item x86
@table @samp
@item mmx
@item mmxext
@item sse
@item sse2
@item sse2slow
@item sse3
@item sse3slow
@item ssse3
@item atom
@item sse4.1
@item sse4.2
@item avx
@item avx2
@item xop
@item fma3
@item fma4
@item 3dnow
@item 3dnowext
@item bmi1
@item bmi2
@item cmov
@end table
@item ARM
@table @samp
@item armv5te
@item armv6
@item armv6t2
@item vfp
@item vfpv3
@item neon
@item setend
@end table
@item AArch64
@table @samp
@item armv8
@item vfp
@item neon
@end table
@item PowerPC
@table @samp
@item altivec
@end table
@item Specific Processors
@table @samp
@item pentium2
@item pentium3
@item pentium4
@item k6
@item k62
@item athlon
@item athlonxp
@item k8
@end table
@end table
@end table
@section AVOptions
These options are provided directly by the libavformat, libavdevice and
libavcodec libraries. To see the list of available AVOptions, use the
@option{-help} option. They are separated into two categories:
@table @option
@item generic
These options can be set for any container, codec or device. Generic options
are listed under AVFormatContext options for containers/devices and under
AVCodecContext options for codecs.
@item private
These options are specific to the given container, device or codec. Private
options are listed under their corresponding containers/devices/codecs.
@end table
For example to write an ID3v2.3 header instead of a default ID3v2.4 to
an MP3 file, use the @option{id3v2_version} private option of the MP3
muxer:
@example
ffmpeg -i input.flac -id3v2_version 3 out.mp3
@end example
All codec AVOptions are per-stream, and thus a stream specifier
should be attached to them:
@example
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
@end example
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 @option{-nooption} syntax cannot be used for boolean
AVOptions, use @option{-option 0}/@option{-option 1}.
Note: the old undocumented way of specifying per-stream AVOptions by
prepending v/a/s to the options name is now obsolete and will be
removed soon.

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Filter design
=============
This document explains guidelines that should be observed (or ignored with
good reason) when writing filters for libavfilter.
In this document, the word “frame” indicates either a video frame or a group
of audio samples, as stored in an AVFrame structure.
Format negotiation
==================
The query_formats method should set, for each input and each output links,
the list of supported formats.
For video links, that means pixel format. For audio links, that means
channel layout, sample format (the sample packing is implied by the sample
format) and sample rate.
The lists are not just lists, they are references to shared objects. When
the negotiation mechanism computes the intersection of the formats
supported at each end of a link, all references to both lists are replaced
with a reference to the intersection. And when a single format is
eventually chosen for a link amongst the remaining list, again, all
references to the list are updated.
That means that if a filter requires that its input and output have the
same format amongst a supported list, all it has to do is use a reference
to the same list of formats.
query_formats can leave some formats unset and return AVERROR(EAGAIN) to
cause the negotiation mechanism to try again later. That can be used by
filters with complex requirements to use the format negotiated on one link
to set the formats supported on another.
Frame references ownership and permissions
==========================================
Principle
---------
Audio and video data are voluminous; the frame and frame reference
mechanism is intended to avoid, as much as possible, expensive copies of
that data while still allowing the filters to produce correct results.
The data is stored in buffers represented by AVFrame structures.
Several references can point to the same frame buffer; the buffer is
automatically deallocated once all corresponding references have been
destroyed.
The characteristics of the data (resolution, sample rate, etc.) are
stored in the reference; different references for the same buffer can
show different characteristics. In particular, a video reference can
point to only a part of a video buffer.
A reference is usually obtained as input to the filter_frame method or
requested using the ff_get_video_buffer or ff_get_audio_buffer
functions. A new reference on an existing buffer can be created with
av_frame_ref(). A reference is destroyed using
the av_frame_free() function.
Reference ownership
-------------------
At any time, a reference “belongs” to a particular piece of code,
usually a filter. With a few caveats that will be explained below, only
that piece of code is allowed to access it. It is also responsible for
destroying it, although this is sometimes done automatically (see the
section on link reference fields).
Here are the (fairly obvious) rules for reference ownership:
* A reference received by the filter_frame method belongs to the
corresponding filter.
* A reference passed to ff_filter_frame is given away and must no longer
be used.
* A reference created with av_frame_ref() belongs to the code that
created it.
* A reference obtained with ff_get_video_buffer or ff_get_audio_buffer
belongs to the code that requested it.
* A reference given as return value by the get_video_buffer or
get_audio_buffer method is given away and must no longer be used.
Link reference fields
---------------------
The AVFilterLink structure has a few AVFrame fields.
partial_buf is used by libavfilter internally and must not be accessed
by filters.
fifo contains frames queued in the filter's input. They belong to the
framework until they are taken by the filter.
Reference permissions
---------------------
Since the same frame data can be shared by several frames, modifying may
have unintended consequences. A frame is considered writable if only one
reference to it exists. The code owning that reference it then allowed
to modify the data.
A filter can check if a frame is writable by using the
av_frame_is_writable() function.
A filter can ensure that a frame is writable at some point of the code
by using the ff_inlink_make_frame_writable() function. It will duplicate
the frame if needed.
A filter can ensure that the frame passed to the filter_frame() callback
is writable by setting the needs_writable flag on the corresponding
input pad. It does not apply to the activate() callback.
Frame scheduling
================
The purpose of these rules is to ensure that frames flow in the filter
graph without getting stuck and accumulating somewhere.
Simple filters that output one frame for each input frame should not have
to worry about it.
There are two design for filters: one using the filter_frame() and
request_frame() callbacks and the other using the activate() callback.
The design using filter_frame() and request_frame() is legacy, but it is
suitable for filters that have a single input and process one frame at a
time. New filters with several inputs, that treat several frames at a time
or that require a special treatment at EOF should probably use the design
using activate().
activate
--------
This method is called when something must be done in a filter; the
definition of that "something" depends on the semantic of the filter.
The callback must examine the status of the filter's links and proceed
accordingly.
The status of output links is stored in the frame_wanted_out, status_in
and status_out fields and tested by the ff_outlink_frame_wanted()
function. If this function returns true, then the processing requires a
frame on this link and the filter is expected to make efforts in that
direction.
The status of input links is stored by the status_in, fifo and
status_out fields; they must not be accessed directly. The fifo field
contains the frames that are queued in the input for processing by the
filter. The status_in and status_out fields contains the queued status
(EOF or error) of the link; status_in is a status change that must be
taken into account after all frames in fifo have been processed;
status_out is the status that have been taken into account, it is final
when it is not 0.
The typical task of an activate callback is to first check the backward
status of output links, and if relevant forward it to the corresponding
input. Then, if relevant, for each input link: test the availability of
frames in fifo and process them; if no frame is available, test and
acknowledge a change of status using ff_inlink_acknowledge_status(); and
forward the result (frame or status change) to the corresponding input.
If nothing is possible, test the status of outputs and forward it to the
corresponding input(s). If still not possible, return FFERROR_NOT_READY.
If the filters stores internally one or a few frame for some input, it
can consider them to be part of the FIFO and delay acknowledging a
status change accordingly.
Example code:
ret = ff_outlink_get_status(outlink);
if (ret) {
ff_inlink_set_status(inlink, ret);
return 0;
}
if (priv->next_frame) {
/* use it */
return 0;
}
ret = ff_inlink_consume_frame(inlink, &frame);
if (ret < 0)
return ret;
if (ret) {
/* use it */
return 0;
}
ret = ff_inlink_acknowledge_status(inlink, &status, &pts);
if (ret) {
/* flush */
ff_outlink_set_status(outlink, status, pts);
return 0;
}
if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(inlink);
return 0;
}
return FFERROR_NOT_READY;
The exact code depends on how similar the /* use it */ blocks are and
how related they are to the /* flush */ block, and needs to apply these
operations to the correct inlink or outlink if there are several.
Macros are available to factor that when no extra processing is needed:
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
FF_FILTER_FORWARD_STATUS_ALL(outlink, filter);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_STATUS_ALL(inlink, filter);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
filter_frame
------------
For filters that do not use the activate() callback, this method is
called when a frame is pushed to the filter's input. It can be called at
any time except in a reentrant way.
If the input frame is enough to produce output, then the filter should
push the output frames on the output link immediately.
As an exception to the previous rule, if the input frame is enough to
produce several output frames, then the filter needs output only at
least one per link. The additional frames can be left buffered in the
filter; these buffered frames must be flushed immediately if a new input
produces new output.
(Example: frame rate-doubling filter: filter_frame must (1) flush the
second copy of the previous frame, if it is still there, (2) push the
first copy of the incoming frame, (3) keep the second copy for later.)
If the input frame is not enough to produce output, the filter must not
call request_frame to get more. It must just process the frame or queue
it. The task of requesting more frames is left to the filter's
request_frame method or the application.
If a filter has several inputs, the filter must be ready for frames
arriving randomly on any input. Therefore, any filter with several inputs
will most likely require some kind of queuing mechanism. It is perfectly
acceptable to have a limited queue and to drop frames when the inputs
are too unbalanced.
request_frame
-------------
For filters that do not use the activate() callback, this method is
called when a frame is wanted on an output.
For a source, it should directly call filter_frame on the corresponding
output.
For a filter, if there are queued frames already ready, one of these
frames should be pushed. If not, the filter should request a frame on
one of its inputs, repeatedly until at least one frame has been pushed.
Return values:
if request_frame could produce a frame, or at least make progress
towards producing a frame, it should return 0;
if it could not for temporary reasons, it should return AVERROR(EAGAIN);
if it could not because there are no more frames, it should return
AVERROR_EOF.
The typical implementation of request_frame for a filter with several
inputs will look like that:
if (frames_queued) {
push_one_frame();
return 0;
}
input = input_where_a_frame_is_most_needed();
ret = ff_request_frame(input);
if (ret == AVERROR_EOF) {
process_eof_on_input();
} else if (ret < 0) {
return ret;
}
return 0;
Note that, except for filters that can have queued frames and sources,
request_frame does not push frames: it requests them to its input, and
as a reaction, the filter_frame method possibly will be called and do
the work.

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@chapter Format Options
@c man begin FORMAT OPTIONS
The libavformat library provides some generic global options, which
can be set on all the muxers and demuxers. In addition each muxer or
demuxer may support so-called private options, which are specific for
that component.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, or by setting the value explicitly in the
@code{AVFormatContext} options or using the @file{libavutil/opt.h} API
for programmatic use.
The list of supported options follows:
@table @option
@item avioflags @var{flags} (@emph{input/output})
Possible values:
@table @samp
@item direct
Reduce buffering.
@end table
@item probesize @var{integer} (@emph{input})
Set probing size in bytes, i.e. the size of the data to analyze to get
stream information. A higher value will enable detecting more
information in case it is dispersed into the stream, but will increase
latency. Must be an integer not lesser than 32. It is 5000000 by default.
@item max_probe_packets @var{integer} (@emph{input})
Set the maximum number of buffered packets when probing a codec.
Default is 2500 packets.
@item packetsize @var{integer} (@emph{output})
Set packet size.
@item fflags @var{flags}
Set format flags. Some are implemented for a limited number of formats.
Possible values for input files:
@table @samp
@item discardcorrupt
Discard corrupted packets.
@item fastseek
Enable fast, but inaccurate seeks for some formats.
@item genpts
Generate missing PTS if DTS is present.
@item igndts
Ignore DTS if PTS is set. Inert when nofillin is set.
@item ignidx
Ignore index.
@item keepside (@emph{deprecated},@emph{inert})
@item nobuffer
Reduce the latency introduced by buffering during initial input streams analysis.
@item nofillin
Do not fill in missing values in packet fields that can be exactly calculated.
@item noparse
Disable AVParsers, this needs @code{+nofillin} too.
@item sortdts
Try to interleave output packets by DTS. At present, available only for AVIs with an index.
@end table
Possible values for output files:
@table @samp
@item autobsf
Automatically apply bitstream filters as required by the output format. Enabled by default.
@item bitexact
Only write platform-, build- and time-independent data.
This ensures that file and data checksums are reproducible and match between
platforms. Its primary use is for regression testing.
@item flush_packets
Write out packets immediately.
@item latm (@emph{deprecated},@emph{inert})
@item shortest
Stop muxing at the end of the shortest stream.
It may be needed to increase max_interleave_delta to avoid flushing the longer
streams before EOF.
@end table
@item seek2any @var{integer} (@emph{input})
Allow seeking to non-keyframes on demuxer level when supported if set to 1.
Default is 0.
@item analyzeduration @var{integer} (@emph{input})
Specify how many microseconds are analyzed to probe the input. A
higher value will enable detecting more accurate information, but will
increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
@item cryptokey @var{hexadecimal string} (@emph{input})
Set decryption key.
@item indexmem @var{integer} (@emph{input})
Set max memory used for timestamp index (per stream).
@item rtbufsize @var{integer} (@emph{input})
Set max memory used for buffering real-time frames.
@item fdebug @var{flags} (@emph{input/output})
Print specific debug info.
Possible values:
@table @samp
@item ts
@end table
@item max_delay @var{integer} (@emph{input/output})
Set maximum muxing or demuxing delay in microseconds.
@item fpsprobesize @var{integer} (@emph{input})
Set number of frames used to probe fps.
@item audio_preload @var{integer} (@emph{output})
Set microseconds by which audio packets should be interleaved earlier.
@item chunk_duration @var{integer} (@emph{output})
Set microseconds for each chunk.
@item chunk_size @var{integer} (@emph{output})
Set size in bytes for each chunk.
@item err_detect, f_err_detect @var{flags} (@emph{input})
Set error detection flags. @code{f_err_detect} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item crccheck
Verify embedded CRCs.
@item bitstream
Detect bitstream specification deviations.
@item buffer
Detect improper bitstream length.
@item explode
Abort decoding on minor error detection.
@item careful
Consider things that violate the spec and have not been seen in the
wild as errors.
@item compliant
Consider all spec non compliancies as errors.
@item aggressive
Consider things that a sane encoder should not do as an error.
@end table
@item max_interleave_delta @var{integer} (@emph{output})
Set maximum buffering duration for interleaving. The duration is
expressed in microseconds, and defaults to 10000000 (10 seconds).
To ensure all the streams are interleaved correctly, libavformat will
wait until it has at least one packet for each stream before actually
writing any packets to the output file. When some streams are
"sparse" (i.e. there are large gaps between successive packets), this
can result in excessive buffering.
This field specifies the maximum difference between the timestamps of the
first and the last packet in the muxing queue, above which libavformat
will output a packet regardless of whether it has queued a packet for all
the streams.
If set to 0, libavformat will continue buffering packets until it has
a packet for each stream, regardless of the maximum timestamp
difference between the buffered packets.
@item use_wallclock_as_timestamps @var{integer} (@emph{input})
Use wallclock as timestamps if set to 1. Default is 0.
@item avoid_negative_ts @var{integer} (@emph{output})
Possible values:
@table @samp
@item make_non_negative
Shift timestamps to make them non-negative.
Also note that this affects only leading negative timestamps, and not
non-monotonic negative timestamps.
@item make_zero
Shift timestamps so that the first timestamp is 0.
@item auto (default)
Enables shifting when required by the target format.
@item disabled
Disables shifting of timestamp.
@end table
When shifting is enabled, all output timestamps are shifted by the
same amount. Audio, video, and subtitles desynching and relative
timestamp differences are preserved compared to how they would have
been without shifting.
@item skip_initial_bytes @var{integer} (@emph{input})
Set number of bytes to skip before reading header and frames if set to 1.
Default is 0.
@item correct_ts_overflow @var{integer} (@emph{input})
Correct single timestamp overflows if set to 1. Default is 1.
@item flush_packets @var{integer} (@emph{output})
Flush the underlying I/O stream after each packet. Default is -1 (auto), which
means that the underlying protocol will decide, 1 enables it, and has the
effect of reducing the latency, 0 disables it and may increase IO throughput in
some cases.
@item output_ts_offset @var{offset} (@emph{output})
Set the output time offset.
@var{offset} must be a time duration specification,
see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
The offset is added by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams are
delayed bt the time duration specified in @var{offset}. Default value
is @code{0} (meaning that no offset is applied).
@item format_whitelist @var{list} (@emph{input})
"," separated list of allowed demuxers. By default all are allowed.
@item dump_separator @var{string} (@emph{input})
Separator used to separate the fields printed on the command line about the
Stream parameters.
For example, to separate the fields with newlines and indentation:
@example
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
@end example
@item max_streams @var{integer} (@emph{input})
Specifies the maximum number of streams. This can be used to reject files that
would require too many resources due to a large number of streams.
@item skip_estimate_duration_from_pts @var{bool} (@emph{input})
Skip estimation of input duration when calculated using PTS.
At present, applicable for MPEG-PS and MPEG-TS.
@item strict, f_strict @var{integer} (@emph{input/output})
Specify how strictly to follow the standards. @code{f_strict} is deprecated and
should be used only via the @command{ffmpeg} tool.
Possible values:
@table @samp
@item very
strictly conform to an older more strict version of the spec or reference software
@item strict
strictly conform to all the things in the spec no matter what consequences
@item normal
@item unofficial
allow unofficial extensions
@item experimental
allow non standardized experimental things, experimental
(unfinished/work in progress/not well tested) decoders and encoders.
Note: experimental decoders can pose a security risk, do not use this for
decoding untrusted input.
@end table
@end table
@c man end FORMAT OPTIONS
@anchor{Format stream specifiers}
@section Format stream specifiers
Format stream specifiers allow selection of one or more streams that
match specific properties.
The exact semantics of stream specifiers is defined by the
@code{avformat_match_stream_specifier()} function declared in the
@file{libavformat/avformat.h} header and documented in the
@ref{Stream specifiers,,Stream specifiers section in the ffmpeg(1) manual,ffmpeg}.
@ifclear config-writeonly
@include demuxers.texi
@end ifclear
@ifclear config-readonly
@include muxers.texi
@end ifclear
@include metadata.texi

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Using Git to develop FFmpeg
@titlepage
@center @titlefont{Using Git to develop FFmpeg}
@end titlepage
@top
@contents
@chapter Introduction
This document aims in giving some quick references on a set of useful Git
commands. You should always use the extensive and detailed documentation
provided directly by Git:
@example
git --help
man git
@end example
shows you the available subcommands,
@example
git <command> --help
man git-<command>
@end example
shows information about the subcommand <command>.
Additional information could be found on the
@url{http://gitref.org, Git Reference} website.
For more information about the Git project, visit the
@url{http://git-scm.com/, Git website}.
Consult these resources whenever you have problems, they are quite exhaustive.
What follows now is a basic introduction to Git and some FFmpeg-specific
guidelines to ease the contribution to the project.
@chapter Basics Usage
@section Get Git
You can get Git from @url{http://git-scm.com/}
Most distribution and operating system provide a package for it.
@section Cloning the source tree
@example
git clone git://source.ffmpeg.org/ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>}.
@example
git clone git@@source.ffmpeg.org:ffmpeg <target>
@end example
This will put the FFmpeg sources into the directory @var{<target>} and let
you push back your changes to the remote repository.
@example
git clone gil@@ffmpeg.org:ffmpeg-web <target>
@end example
This will put the source of the FFmpeg website into the directory
@var{<target>} and let you push back your changes to the remote repository.
(Note that @var{gil} stands for GItoLite and is not a typo of @var{git}.)
If you don't have write-access to the ffmpeg-web repository, you can
create patches after making a read-only ffmpeg-web clone:
@example
git clone git://ffmpeg.org/ffmpeg-web <target>
@end example
Make sure that you do not have Windows line endings in your checkouts,
otherwise you may experience spurious compilation failures. One way to
achieve this is to run
@example
git config --global core.autocrlf false
@end example
@anchor{Updating the source tree to the latest revision}
@section Updating the source tree to the latest revision
@example
git pull (--rebase)
@end example
pulls in the latest changes from the tracked branch. The tracked branch
can be remote. By default the master branch tracks the branch master in
the remote origin.
@float IMPORTANT
@command{--rebase} (see below) is recommended.
@end float
@section Rebasing your local branches
@example
git pull --rebase
@end example
fetches the changes from the main repository and replays your local commits
over it. This is required to keep all your local changes at the top of
FFmpeg's master tree. The master tree will reject pushes with merge commits.
@section Adding/removing files/directories
@example
git add [-A] <filename/dirname>
git rm [-r] <filename/dirname>
@end example
Git needs to get notified of all changes you make to your working
directory that makes files appear or disappear.
Line moves across files are automatically tracked.
@section Showing modifications
@example
git diff <filename(s)>
@end example
will show all local modifications in your working directory as unified diff.
@section Inspecting the changelog
@example
git log <filename(s)>
@end example
You may also use the graphical tools like @command{gitview} or @command{gitk}
or the web interface available at @url{http://source.ffmpeg.org/}.
@section Checking source tree status
@example
git status
@end example
detects all the changes you made and lists what actions will be taken in case
of a commit (additions, modifications, deletions, etc.).
@section Committing
@example
git diff --check
@end example
to double check your changes before committing them to avoid trouble later
on. All experienced developers do this on each and every commit, no matter
how small.
Every one of them has been saved from looking like a fool by this many times.
It's very easy for stray debug output or cosmetic modifications to slip in,
please avoid problems through this extra level of scrutiny.
For cosmetics-only commits you should get (almost) empty output from
@example
git diff -w -b <filename(s)>
@end example
Also check the output of
@example
git status
@end example
to make sure you don't have untracked files or deletions.
@example
git add [-i|-p|-A] <filenames/dirnames>
@end example
Make sure you have told Git your name and email address
@example
git config --global user.name "My Name"
git config --global user.email my@@email.invalid
@end example
Use @option{--global} to set the global configuration for all your Git checkouts.
Git will select the changes to the files for commit. Optionally you can use
the interactive or the patch mode to select hunk by hunk what should be
added to the commit.
@example
git commit
@end example
Git will commit the selected changes to your current local branch.
You will be prompted for a log message in an editor, which is either
set in your personal configuration file through
@example
git config --global core.editor
@end example
or set by one of the following environment variables:
@var{GIT_EDITOR}, @var{VISUAL} or @var{EDITOR}.
Log messages should be concise but descriptive. Explain why you made a change,
what you did will be obvious from the changes themselves most of the time.
Saying just "bug fix" or "10l" is bad. Remember that people of varying skill
levels look at and educate themselves while reading through your code. Don't
include filenames in log messages, Git provides that information.
Possibly make the commit message have a terse, descriptive first line, an
empty line and then a full description. The first line will be used to name
the patch by @command{git format-patch}.
@section Preparing a patchset
@example
git format-patch <commit> [-o directory]
@end example
will generate a set of patches for each commit between @var{<commit>} and
current @var{HEAD}. E.g.
@example
git format-patch origin/master
@end example
will generate patches for all commits on current branch which are not
present in upstream.
A useful shortcut is also
@example
git format-patch -n
@end example
which will generate patches from last @var{n} commits.
By default the patches are created in the current directory.
@section Sending patches for review
@example
git send-email <commit list|directory>
@end example
will send the patches created by @command{git format-patch} or directly
generates them. All the email fields can be configured in the global/local
configuration or overridden by command line.
Note that this tool must often be installed separately (e.g. @var{git-email}
package on Debian-based distros).
@section Renaming/moving/copying files or contents of files
Git automatically tracks such changes, making those normal commits.
@example
mv/cp path/file otherpath/otherfile
git add [-A] .
git commit
@end example
@chapter Git configuration
In order to simplify a few workflows, it is advisable to configure both
your personal Git installation and your local FFmpeg repository.
@section Personal Git installation
Add the following to your @file{~/.gitconfig} to help @command{git send-email}
and @command{git format-patch} detect renames:
@example
[diff]
renames = copy
@end example
@section Repository configuration
In order to have @command{git send-email} automatically send patches
to the ffmpeg-devel mailing list, add the following stanza
to @file{/path/to/ffmpeg/repository/.git/config}:
@example
[sendemail]
to = ffmpeg-devel@@ffmpeg.org
@end example
@chapter FFmpeg specific
@section Reverting broken commits
@example
git reset <commit>
@end example
@command{git reset} will uncommit the changes till @var{<commit>} rewriting
the current branch history.
@example
git commit --amend
@end example
allows one to amend the last commit details quickly.
@example
git rebase -i origin/master
@end example
will replay local commits over the main repository allowing to edit, merge
or remove some of them in the process.
@float NOTE
@command{git reset}, @command{git commit --amend} and @command{git rebase}
rewrite history, so you should use them ONLY on your local or topic branches.
The main repository will reject those changes.
@end float
@example
git revert <commit>
@end example
@command{git revert} will generate a revert commit. This will not make the
faulty commit disappear from the history.
@section Pushing changes to remote trees
@example
git push origin master --dry-run
@end example
Will simulate a push of the local master branch to the default remote
(@var{origin}). And list which branches and ranges or commits would have been
pushed.
Git will prevent you from pushing changes if the local and remote trees are
out of sync. Refer to @ref{Updating the source tree to the latest revision}.
@example
git remote add <name> <url>
@end example
Will add additional remote with a name reference, it is useful if you want
to push your local branch for review on a remote host.
@example
git push <remote> <refspec>
@end example
Will push the changes to the @var{<remote>} repository.
Omitting @var{<refspec>} makes @command{git push} update all the remote
branches matching the local ones.
@section Finding a specific svn revision
Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
based on a regular expression. see man gitrevisions
@example
git show :/'as revision 23456'
@end example
will show the svn changeset @samp{r23456}. With older Git versions searching in
the @command{git log} output is the easiest option (especially if a pager with
search capabilities is used).
This commit can be checked out with
@example
git checkout -b svn_23456 :/'as revision 23456'
@end example
or for Git < 1.7.1 with
@example
git checkout -b svn_23456 $SHA1
@end example
where @var{$SHA1} is the commit hash from the @command{git log} output.
@chapter Pre-push checklist
Once you have a set of commits that you feel are ready for pushing,
work through the following checklist to doublecheck everything is in
proper order. This list tries to be exhaustive. In case you are just
pushing a typo in a comment, some of the steps may be unnecessary.
Apply your common sense, but if in doubt, err on the side of caution.
First, make sure that the commits and branches you are going to push
match what you want pushed and that nothing is missing, extraneous or
wrong. You can see what will be pushed by running the git push command
with @option{--dry-run} first. And then inspecting the commits listed with
@command{git log -p 1234567..987654}. The @command{git status} command
may help in finding local changes that have been forgotten to be added.
Next let the code pass through a full run of our test suite.
@itemize
@item @command{make distclean}
@item @command{/path/to/ffmpeg/configure}
@item @command{make fate}
@item if fate fails due to missing samples run @command{make fate-rsync} and retry
@end itemize
Make sure all your changes have been checked before pushing them, the
test suite only checks against regressions and that only to some extend. It does
obviously not check newly added features/code to be working unless you have
added a test for that (which is recommended).
Also note that every single commit should pass the test suite, not just
the result of a series of patches.
Once everything passed, push the changes to your public ffmpeg clone and post a
merge request to ffmpeg-devel. You can also push them directly but this is not
recommended.
@chapter Server Issues
Contact the project admins at @email{root@@ffmpeg.org} if you have technical
problems with the Git server.

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FFmpeg's bug/feature request tracker manual
=================================================
Overview:
---------
FFmpeg uses Trac for tracking issues, new issues and changes to
existing issues can be done through a web interface.
Issues can be different kinds of things we want to keep track of
but that do not belong into the source tree itself. This includes
bug reports, feature requests and license violations. We
might add more items to this list in the future, so feel free to
propose a new `type of issue' on the ffmpeg-devel mailing list if
you feel it is worth tracking.
It is possible to subscribe to individual issues by adding yourself to the
Cc list or to subscribe to the ffmpeg-trac mailing list which receives
a mail for every change to every issue.
(the above does all work already after light testing)
The subscription URL for the ffmpeg-trac list is:
https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
The URL of the webinterface of the tracker is:
https://trac.ffmpeg.org
Type:
-----
art
Artwork such as photos, music, banners, and logos.
bug / defect
An error, flaw, mistake, failure, or fault in FFmpeg or libav* that
prevents it from behaving as intended.
feature request / enhancement
Request of support for encoding or decoding of a new codec, container
or variant.
Request of support for more, less or plain different output or behavior
where the current implementation cannot be considered wrong.
license violation
Ticket to keep track of (L)GPL violations of ffmpeg by others.
sponsoring request
Developer requests for hardware, software, specifications, money,
refunds, etc.
task
A task/reminder such as setting up a FATE client, adding filters to
Trac, etc.
Priority:
---------
critical
Bugs about data loss and security issues.
No feature request can be critical.
important
Bugs which make FFmpeg unusable for a significant number of users.
Examples here might be completely broken MPEG-4 decoding or a build issue
on Linux.
While broken 4xm decoding or a broken OS/2 build would not be important,
the separation to normal is somewhat fuzzy.
For feature requests this priority would be used for things many people
want.
Regressions also should be marked as important, regressions are bugs that
don't exist in a past revision or another branch.
normal
Default setting. Use this if the bug does not match the other
priorities or if you are unsure of what priority to choose.
minor
Bugs about things like spelling errors, "mp2" instead of
"mp3" being shown and such.
Feature requests about things few people want or which do not make a big
difference.
wish
Something that is desirable to have but that there is no urgency at
all to implement, e.g. something completely cosmetic like a website
restyle or a personalized doxy template or the FFmpeg logo.
This priority is not valid for bugs.
Status:
-------
new
initial state
open
intermediate states
closed
final state
Analyzed flag:
--------------
Bugs which have been analyzed and where it is understood what causes them
and which exact chain of events triggers them. This analysis should be
available as a message in the bug report.
Note, do not change the status to analyzed without also providing a clear
and understandable analysis.
This state implicates that the bug either has been reproduced or that
reproduction is not needed as the bug is already understood.
Type/Status:
----------
*/new
Initial state of new bugs and feature requests submitted by
users.
*/open
Issues which have been briefly looked at and which did not look outright
invalid.
This implicates that no real more detailed state applies yet. Conversely,
the more detailed states below implicate that the issue has been briefly
looked at.
*/closed/duplicate
Bugs or feature requests which are duplicates.
Note, if you mark something as duplicate, do not forget setting the
superseder so bug reports are properly linked.
*/closed/invalid
Bugs caused by user errors, random ineligible or otherwise nonsense stuff.
*/closed/needs_more_info
Issues for which some information has been requested by the developers,
but which has not been provided by anyone within reasonable time.
bug/closed/fixed
Bugs which have to the best of our knowledge been fixed.
bug/closed/wontfix
Bugs which we will not fix. Possible reasons include legality, high
complexity for the sake of supporting obscure corner cases, speed loss
for similarly esoteric purposes, et cetera.
This also means that we would reject a patch.
If we are just too lazy to fix a bug then the correct state is open
and unassigned. Closed means that the case is closed which is not
the case if we are just waiting for a patch.
bug/closed/works_for_me
Bugs for which sufficient information was provided to reproduce but
reproduction failed - that is the code seems to work correctly to the
best of our knowledge.
feature_request/closed/fixed
Feature requests which have been implemented.
feature_request/closed/wontfix
Feature requests which will not be implemented. The reasons here could
be legal, philosophical or others.
Note2, if you provide the requested info do not forget to remove the
needs_more_info resolution.
Component:
----------
avcodec
issues in libavcodec/*
avdevice
issues in libavdevice/*
avfilter
issues in libavfilter/*
avformat
issues in libavformat/*
avutil
issues in libavutil/*
build system
issues in or related to configure/Makefile
documentation
issues in or related to doc/*
ffmpeg
issues in or related to ffmpeg.c
ffplay
issues in or related to ffplay.c
ffprobe
issues in or related to ffprobe.c
postproc
issues in libpostproc/*
swresample
issues in libswresample/*
swscale
issues in libswscale/*
trac
issues related to our issue tracker
undetermined
default component; choose this if unsure
website
issues related to the website
wiki
issues related to the wiki

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Common abbreviations/shorthands we use that don't need a comment
================================================================
dsp: digital signal processing
dst/adst: (asymmetric) discrete sine transform
ec: entropy coding or error concealment
er: error resilience
fdct/idct: forward/inverse discrete cosine transform
fft: fast Fourier transform
gop: group of pictures
hw/sw: hardware/software
lp: lowpass
lpf: loop filter
lut: lookup table
mb: macroblock
mc: motion compensation
me: motion estimation
mv: motion vector
nal: network abstraction layer
pel/qpel/epel/hpel/fpel: pixel / quarter-pixel / eighth-pixel / half-pixel / full-pixel
pp: post process
qp: quantization parameter
rc: rate control
sei: supplemental enhancement information
sl: slice
vlc: variable length coding
vq: vector quantization

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CONTEXT
=======
The FFmpeg project merges all the changes from the Libav project
(https://libav.org) since the origin of the fork (around 2011).
With the exceptions of some commits due to technical/political disagreements or
issues, the changes are merged on a more or less regular schedule (daily for
years thanks to Michael, but more sparse nowadays).
WHY
===
The majority of the active developers believe the project needs to keep this
policy for various reasons.
The most important one is that we don't want our users to have to choose
between two distributors of libraries of the exact same name in order to have a
different set of features and bugfixes. By taking the responsibility of
unifying the two codebases, we allow users to benefit from the changes from the
two teams.
Today, FFmpeg has a much larger user database (we are distributed by every
major distribution), so we consider this mission a priority.
A different approach to the merge could have been to pick the changes we are
interested in and drop most of the cosmetics and other less important changes.
Unfortunately, this makes the following picks much harder, especially since the
Libav project is involved in various deep API changes. As a result, we decide
to virtually take everything done there.
Any Libav developer is of course welcome anytime to contribute directly to the
FFmpeg tree. Of course, we fully understand and are forced to accept that very
few Libav developers are interested in doing so, but we still want to recognize
their work. This leads us to create merge commits for every single one from
Libav. The original commit appears totally unchanged with full authorship in
our history (and the conflict are solved in the merge one). That way, not a
single thing from Libav will be lost in the future in case some reunification
happens, or that project disappears one way or another.
DOWNSIDES
=========
Of course, there are many downsides to this approach.
- It causes a non negligible merge commits pollution. We make sure there are
not several level of merges entangled (we do a 1:1 merge/commit), but it's
still a non-linear history.
- Many duplicated work. For instance, we added libavresample in our tree to
keep compatibility with Libav when our libswresample was already covering the
exact same purpose. The same thing happened for various elements such as the
ProRes support (but differences in features, bugs, licenses, ...). There are
many work to do to unify them, and any help is very much welcome.
- So much manpower from both FFmpeg and Libav is lost because of this mess. We
know it, and we don't know how to fix it. It takes incredible time to do
these merges, so we have even less time to work on things we personally care
about. The bad vibes also do not help with keeping our developers motivated.
- There is a growing technical risk factor with the merges due to the codebase
differing more and more.
MERGE GUIDELINES
================
The following gives developer guidelines on how to proceed when merging Libav commits.
Before starting, you can reduce the risk of errors on merge conflicts by using
a different merge conflict style:
$ git config --global merge.conflictstyle diff3
tools/libav-merge-next-commit is a script to help merging the next commit in
the queue. It assumes a remote named libav. It has two modes: merge, and noop.
The noop mode creates a merge with no change to the HEAD. You can pass a hash
as extra argument to reference a justification (it is common that we already
have the change done in FFmpeg).
Also see tools/murge, you can copy and paste a 3 way conflict into its stdin
and it will display colored diffs. Any arguments to murge (like ones to suppress
whitespace differences) are passed into colordiff.
TODO/FIXME/UNMERGED
===================
Stuff that didn't reach the codebase:
-------------------------------------
- HEVC DSP and x86 MC SIMD improvements from Libav (see https://ffmpeg.org/pipermail/ffmpeg-devel/2015-December/184777.html)
- 1f821750f hevcdsp: split the qpel functions by width instead of by the subpixel fraction
- 818bfe7f0 hevcdsp: split the epel functions by width
- 688417399 hevcdsp: split the pred functions by width
- a853388d2 hevc: change the stride of the MC buffer to be in bytes instead of elements
- 0cef06df0 checkasm: add HEVC MC tests
- e7078e842 hevcdsp: add x86 SIMD for MC
- 7993ec19a hevc: Add hevc_get_pixel_4/8/12/16/24/32/48/64
- use av_cpu_max_align() instead of hardcoding alignment requirements (see https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/215834.html)
- f44ec22e0 lavc: use av_cpu_max_align() instead of hardcoding alignment requirements
- 4de220d2e frame: allow align=0 (meaning automatic) for av_frame_get_buffer()
- Support recovery from an already present HLS playlist (see 16cb06bb30)
- Remove all output devices (see 8e7e042d41, 8d3db95f20, 6ce13070bd, d46cd24986 and https://ffmpeg.org/pipermail/ffmpeg-devel/2017-September/216904.html)
- avcodec/libaomenc: export the Sequence Header OBU as extradata (See a024c3ce9a)
Collateral damage that needs work locally:
------------------------------------------
- Merge proresenc_anatoliy.c and proresenc_kostya.c
- Fix MIPS AC3 downmix
Extra changes needed to be aligned with Libav:
----------------------------------------------
- Switching our examples to the new encode/decode API (see 67d28f4a0f)
- HEVC IDCT bit depth 12-bit support (Libav added 8 and 10 but doesn't have 12)

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavcodec Documentation
@titlepage
@center @titlefont{Libavcodec Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavcodec library provides a generic encoding/decoding framework
and contains multiple decoders and encoders for audio, video and
subtitle streams, and several bitstream filters.
The shared architecture provides various services ranging from bit
stream I/O to DSP optimizations, and makes it suitable for
implementing robust and fast codecs as well as for experimentation.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-codecs.html,ffmpeg-codecs}, @url{ffmpeg-bitstream-filters.html,bitstream-filters},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavcodec
@settitle media streams decoding and encoding library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavdevice Documentation
@titlepage
@center @titlefont{Libavdevice Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavdevice library provides a generic framework for grabbing from
and rendering to many common multimedia input/output devices, and
supports several input and output devices, including Video4Linux2,
VfW, DShow, and ALSA.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-devices.html,ffmpeg-devices},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-devices(1),
libavutil(3), libavcodec(3), libavformat(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavdevice
@settitle multimedia device handling library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavfilter Documentation
@titlepage
@center @titlefont{Libavfilter Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavfilter library provides a generic audio/video filtering
framework containing several filters, sources and sinks.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-filters.html,ffmpeg-filters},
@url{libavutil.html,libavutil}, @url{libswscale.html,libswscale}, @url{libswresample.html,libswresample},
@url{libavcodec.html,libavcodec}, @url{libavformat.html,libavformat}, @url{libavdevice.html,libavdevice}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-filters(1),
libavutil(3), libswscale(3), libswresample(3), libavcodec(3), libavformat(3), libavdevice(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavfilter
@settitle multimedia filtering library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavformat Documentation
@titlepage
@center @titlefont{Libavformat Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavformat library provides a generic framework for multiplexing
and demultiplexing (muxing and demuxing) audio, video and subtitle
streams. It encompasses multiple muxers and demuxers for multimedia
container formats.
It also supports several input and output protocols to access a media
resource.
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-formats.html,ffmpeg-formats}, @url{ffmpeg-protocols.html,ffmpeg-protocols},
@url{libavutil.html,libavutil}, @url{libavcodec.html,libavcodec}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-formats(1), ffmpeg-protocols(1),
libavutil(3), libavcodec(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavformat
@settitle multimedia muxing and demuxing library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libavutil Documentation
@titlepage
@center @titlefont{Libavutil Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libavutil library is a utility library to aid portable
multimedia programming. It contains safe portable string functions,
random number generators, data structures, additional mathematics
functions, cryptography and multimedia related functionality (like
enumerations for pixel and sample formats). It is not a library for
code needed by both libavcodec and libavformat.
The goals for this library is to be:
@table @strong
@item Modular
It should have few interdependencies and the possibility of disabling individual
parts during @command{./configure}.
@item Small
Both sources and objects should be small.
@item Efficient
It should have low CPU and memory usage.
@item Useful
It should avoid useless features that almost no one needs.
@end table
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-utils.html,ffmpeg-utils}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-utils(1)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libavutil
@settitle multimedia-biased utility library
@end ignore
@bye

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

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Libswscale Documentation
@titlepage
@center @titlefont{Libswscale Documentation}
@end titlepage
@top
@contents
@chapter Description
@c man begin DESCRIPTION
The libswscale library performs highly optimized image scaling and
colorspace and pixel format conversion operations.
Specifically, this library performs the following conversions:
@itemize
@item
@emph{Rescaling}: is the process of changing the video size. Several
rescaling options and algorithms are available. This is usually a
lossy process.
@item
@emph{Pixel format conversion}: is the process of converting the image
format and colorspace of the image, for example from planar YUV420P to
RGB24 packed. It also handles packing conversion, that is converts
from packed layout (all pixels belonging to distinct planes
interleaved in the same buffer), to planar layout (all samples
belonging to the same plane stored in a dedicated buffer or "plane").
This is usually a lossy process in case the source and destination
colorspaces differ.
@end itemize
@c man end DESCRIPTION
@chapter See Also
@ifhtml
@url{ffmpeg.html,ffmpeg}, @url{ffplay.html,ffplay}, @url{ffprobe.html,ffprobe},
@url{ffmpeg-scaler.html,ffmpeg-scaler},
@url{libavutil.html,libavutil}
@end ifhtml
@ifnothtml
ffmpeg(1), ffplay(1), ffprobe(1),
ffmpeg-scaler(1),
libavutil(3)
@end ifnothtml
@include authors.texi
@ignore
@setfilename libswscale
@settitle video scaling and pixel format conversion library
@end ignore
@bye

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle FFmpeg Mailing List FAQ
@titlepage
@center @titlefont{FFmpeg Mailing List FAQ}
@end titlepage
@top
@contents
@chapter General Questions
@section What is a mailing list?
A mailing list is not much different than emailing someone, but the
main difference is that your message is received by everyone who
subscribes to the list. It is somewhat like a forum but in email form.
See the @url{https://lists.ffmpeg.org/pipermail/ffmpeg-user/, ffmpeg-user archives}
for examples.
@section What type of questions can I ask?
@itemize
@item
@url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user}:
For questions involving unscripted usage or compilation of the FFmpeg
command-line tools (@command{ffmpeg}, @command{ffprobe}, @command{ffplay}).
@item
@url{https://lists.ffmpeg.org/mailman/listinfo/libav-user/, libav-user}:
For questions involving the FFmpeg libav* libraries (libavcodec,
libavformat, libavfilter, etc).
@item
@url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel/, ffmpeg-devel}:
For discussions involving the development of FFmpeg and for submitting
patches. User questions should be asked at ffmpeg-user or libav-user.
@end itemize
To report a bug see @url{https://ffmpeg.org/bugreports.html}.
We cannot provide help for scripts and/or third-party tools.
@anchor{How do I ask a question or send a message to a mailing list?}
@section How do I ask a question or send a message to a mailing list?
First you must @ref{How do I subscribe?, subscribe}. Then all you have to do is
send an email:
@itemize
@item
Email @email{ffmpeg-user@@ffmpeg.org} to send a message to the
ffmpeg-user mailing list.
@item
Email @email{libav-user@@ffmpeg.org} to send a message to the
libav-user mailing list.
@item
Email @email{ffmpeg-devel@@ffmpeg.org} to send a message to the
ffmpeg-devel mailing list.
@end itemize
@chapter Subscribing / Unsubscribing
@anchor{How do I subscribe?}
@section How do I subscribe?
Email @email{ffmpeg-user-request@@ffmpeg.org} with the subject
@emph{subscribe}.
Or visit the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page}
and refer to the @emph{Subscribing to ffmpeg-user} section.
The process is the same for the other mailing lists.
@section How do I unsubscribe?
Email @email{ffmpeg-user-request@@ffmpeg.org} with subject @emph{unsubscribe}.
Or visit the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page},
scroll to bottom of page, enter your email address in the box, and click
the @emph{Unsubscribe or edit options} button.
The process is the same for the other mailing lists.
Please avoid asking a mailing list admin to unsubscribe you unless you
are absolutely unable to do so by yourself. See @ref{Who do I contact if I have a problem with the mailing list?}
Note that it is possible to temporarily halt message delivery (vacation mode).
See @ref{How do I disable mail delivery without unsubscribing?}
@chapter Moderation Queue
@anchor{Why is my message awaiting moderator approval?}
@section Why is my message awaiting moderator approval?
Some messages are automatically held in the @emph{moderation queue} and
must be manually approved by a mailing list admin:
These are:
@itemize
@item
Messages that exceed the @ref{What is the message size limit?, message size limit}.
@item
Messages from users whose accounts have been set with the @emph{moderation flag}
(very rarely occurs, but may if a user repeatedly ignores the rules
or is abusive towards others).
@end itemize
@section How long does it take for my message in the moderation queue to be approved?
The queue is not checked on a regular basis. You can ask on the
@t{#ffmpeg-devel} IRC channel on Freenode for someone to approve your message.
@anchor{How do I delete my message in the moderation queue?}
@section How do I delete my message in the moderation queue?
You should have received an email with the subject @emph{Your message to <mailing list name> awaits moderator approval}.
A link is in the message that will allow you to delete your message
unless a mailing list admin already approved or rejected it.
@chapter Archives
@anchor{Where are the archives?}
@section Where are the archives?
See the @emph{Archives} section on the @url{https://ffmpeg.org/contact.html, FFmpeg Contact}
page for links to all FFmpeg mailing list archives.
Note that the archives are split by month. Discussions that span
several months will be split into separate months in the archives.
@section How do I reply to a message in the archives?
Click the email link at the top of the message just under the subject
title. The link will provide the proper headers to keep the message
within the thread.
Note that you must be subscribed to send a message to the ffmpeg-user or
libav-user mailing lists.
@section How do I search the archives?
Perform a site search using your favorite search engine. Example:
@t{site:lists.ffmpeg.org/pipermail/ffmpeg-user/ "search term"}
@chapter Other
@section Is there an alternative to the mailing list?
You can ask for help in the official @t{#ffmpeg} IRC channel on Freenode.
Some users prefer the third-party @url{http://www.ffmpeg-archive.org/, Nabble}
interface which presents the mailing lists in a typical forum layout.
There are also numerous third-party help sites such as
@url{https://superuser.com/tags/ffmpeg, Super User} and
@url{https://www.reddit.com/r/ffmpeg/, r/ffmpeg on reddit}.
@anchor{What is top-posting?}
@section What is top-posting?
See @url{https://en.wikipedia.org/wiki/Posting_style#Top-posting}.
Instead, use trimmed interleaved/inline replies (@url{https://lists.ffmpeg.org/pipermail/ffmpeg-user/2017-April/035849.html, example}).
@anchor{What is the message size limit?}
@section What is the message size limit?
The message size limit is 1000 kilobytes. Please provide links to larger files
instead of attaching them.
@section Where can I upload sample files?
Anywhere that is not too annoying for us to use.
Google Drive and Dropbox are acceptable if you need a file host, and
@url{https://0x0.st/, 0x0.st} is good for files under 256 MiB.
Small, short samples are preferred if possible.
@section Will I receive spam if I send and/or subscribe to a mailing list?
Highly unlikely.
@itemize
@item
The list of subscribed users is not public.
@item
Email addresses in the archives are obfuscated.
@item
Several unique test email accounts were utilized and none have yet
received any spam.
@end itemize
However, you may see a spam in the mailing lists on rare occasions:
@itemize
@item
Spam in the moderation queue may be accidentally approved due to human
error.
@item
There have been a few messages from subscribed users who had their own
email addresses hacked and spam messages from (or appearing to be from)
the hacked account were sent to their contacts (a mailing list being a
contact in these cases).
@item
If you are subscribed to the bug tracker mailing list (ffmpeg-trac) you
may see the occasional spam as a false bug report, but we take measures
to try to prevent this.
@end itemize
@section How do I filter mailing list messages?
Use the @emph{List-Id}. For example, the ffmpeg-user mailing list is
@t{ffmpeg-user.ffmpeg.org}. You can view the List-Id in the raw message
or headers.
You can then filter the mailing list messages to their own folder.
@anchor{How do I disable mail delivery without unsubscribing?}
@section How do I disable mail delivery without unsubscribing?
Sometimes you may want to temporarily stop receiving all mailing list
messages. This "vacation mode" is simple to do:
@enumerate
@item
Go to the @url{https://lists.ffmpeg.org/mailman/listinfo/ffmpeg-user/, ffmpeg-user mailing list info page}
@item
Enter your email address in the box at very bottom of the page and click the
@emph{Unsubscribe or edit options} box.
@item
Enter your password and click the @emph{Log in} button.
@item
Look for the @emph{Mail delivery} option. Here you can disable/enable mail
delivery. If you check @emph{Set globally} it will apply your choice to all
other FFmpeg mailing lists you are subscribed to.
@end enumerate
Alternatively, from your subscribed address, send a message to @email{ffmpeg-user-request@@ffmpeg.org}
with the subject @emph{set delivery off}. To re-enable mail delivery send a
message to @email{ffmpeg-user-request@@ffmpeg.org} with the subject
@emph{set delivery on}.
@anchor{Why is the mailing list munging my address?}
@section Why is the mailing list munging my address?
This is due to subscribers that use an email service with a DMARC reject policy
which adds difficulties to mailing list operators.
The mailing list must re-write (munge) the @emph{From:} header for such users;
otherwise their email service will reject and bounce the message resulting in
automatic unsubscribing from the mailing list.
When sending a message these users will see @emph{via <mailing list name>}
added to their name and the @emph{From:} address munged to the address of
the particular mailing list.
If you want to avoid this then please use a different email service.
Note that ffmpeg-devel does not apply any munging as it causes issues with
patch authorship. As a result users with an email service with a DMARC reject
policy may be automatically unsubscribed due to rejected and bounced messages.
@chapter Rules and Etiquette
@section What are the rules and the proper etiquette?
There may seem to be many things to remember, but we want to help and
following these guidelines will allow you to get answers more quickly
and help avoid getting ignored.
@itemize
@item
Always show your actual, unscripted @command{ffmpeg} command and the
complete, uncut console output from your command.
@item
Use the most simple and minimal command that still shows the issue you
are encountering.
@item
Provide all necessary information so others can attempt to duplicate
your issue. This includes the actual command, complete uncut console
output, and any inputs that are required to duplicate the issue.
@item
Use the latest @command{ffmpeg} build you can get. See the @url{https://ffmpeg.org/download.html, FFmpeg Download}
page for links to recent builds for Linux, macOS, and Windows. Or
compile from the current git master branch.
@item
Avoid @url{https://en.wikipedia.org/wiki/Posting_style#Top-posting, top-posting}.
Also see @ref{What is top-posting?}
@item
Avoid hijacking threads. Thread hijacking is replying to a message and
changing the subject line to something unrelated to the original thread.
Most email clients will still show the renamed message under the
original thread. This can be confusing and these types of messages are
often ignored.
@item
Do not send screenshots. Copy and paste console text instead of making
screenshots of the text.
@item
Avoid sending email disclaimers and legalese if possible as this is a
public list.
@item
Avoid using the @code{-loglevel debug}, @code{-loglevel quiet}, and
@command{-hide_banner} options unless requested to do so.
@item
If you attach files avoid compressing small files. Uncompressed is
preferred.
@item
Please do not send HTML-only messages. The mailing list will ignore the
HTML component of your message. Most mail clients will automatically
include a text component: this is what the mailing list will use.
@item
Configuring your mail client to break lines after 70 or so characters is
recommended.
@item
Avoid sending the same message to multiple mailing lists.
@item
Please follow our @url{https://ffmpeg.org/developer.html#Code-of-conduct, Code of Conduct}.
@end itemize
@chapter Help
@section Why am I not receiving any messages?
Some email providers have blacklists or spam filters that block or mark
the mailing list messages as false positives. Unfortunately, the user is
often not aware of this and is often out of their control.
When possible we attempt to notify the provider to be removed from the
blacklists or filters.
@section Why are my sent messages not showing up?
Excluding @ref{Why is my message awaiting moderator approval?, messages that are held in the moderation queue}
there are a few other reasons why your messages may fail to appear:
@itemize
@item
HTML-only messages are ignored by the mailing lists. Most mail clients
automatically include a text component alongside HTML email: this is what
the mailing list will use. If it does not then consider your client to be
broken, because sending a text component along with the HTML component to
form a multi-part message is recommended by email standards.
@item
Check your spam folder.
@end itemize
@anchor{Why do I keep getting unsubscribed from ffmpeg-devel?}
@section Why do I keep getting unsubscribed from ffmpeg-devel?
Users with an email service that has a DMARC reject or quarantine policy may be
automatically unsubscribed from the ffmpeg-devel mailing list due to the mailing
list messages being continuously rejected and bounced back.
Consider using a different email service.
@anchor{Who do I contact if I have a problem with the mailing list?}
@section Who do I contact if I have a problem with the mailing list?
Send a message to @email{ffmpeg-user-owner@@ffmpeg.org}.
@bye

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

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MIPS optimizations info
===============================================
MIPS optimizations of codecs are targeting MIPS 74k family of
CPUs. Some of these optimizations are relying more on properties of
this architecture and some are relying less (and can be used on most
MIPS architectures without degradation in performance).
Along with FFMPEG copyright notice, there is MIPS copyright notice in
all the files that are created by people from MIPS Technologies.
Example of copyright notice:
===============================================
/*
* Copyright (c) 2012
* MIPS Technologies, Inc., California.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
* contributors may be used to endorse or promote products derived from
* this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
* Author: Author Name (author_name@@mips.com)
*/
Files that have MIPS copyright notice in them:
===============================================
* libavutil/mips/
float_dsp_mips.c
libm_mips.h
softfloat_tables.h
* libavcodec/
fft_fixed_32.c
fft_init_table.c
fft_table.h
mdct_fixed_32.c
* libavcodec/mips/
aacdec_fixed.c
aacsbr_fixed.c
aacsbr_template.c
aaccoder_mips.c
aacpsy_mips.h
ac3dsp_mips.c
acelp_filters_mips.c
acelp_vectors_mips.c
amrwbdec_mips.c
amrwbdec_mips.h
celp_filters_mips.c
celp_math_mips.c
compute_antialias_fixed.h
compute_antialias_float.h
lsp_mips.h
dsputil_mips.c
fft_mips.c
fft_table.h
fft_init_table.c
fmtconvert_mips.c
iirfilter_mips.c
mpegaudiodsp_mips_fixed.c
mpegaudiodsp_mips_float.c

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

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

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optimization Tips (for libavcodec):
===================================
What to optimize:
-----------------
If you plan to do non-x86 architecture specific optimizations (SIMD normally),
then take a look in the x86/ directory, as most important functions are
already optimized for MMX.
If you want to do x86 optimizations then you can either try to fine-tune the
stuff in the x86 directory or find some other functions in the C source to
optimize, but there aren't many left.
Understanding these overoptimized functions:
--------------------------------------------
As many functions tend to be a bit difficult to understand because
of optimizations, it can be hard to optimize them further, or write
architecture-specific versions. It is recommended to look at older
revisions of the interesting files (web frontends for the various FFmpeg
branches are listed at http://ffmpeg.org/download.html).
Alternatively, look into the other architecture-specific versions in
the x86/, ppc/, alpha/ subdirectories. Even if you don't exactly
comprehend the instructions, it could help understanding the functions
and how they can be optimized.
NOTE: If you still don't understand some function, ask at our mailing list!!!
(http://lists.ffmpeg.org/mailman/listinfo/ffmpeg-devel)
When is an optimization justified?
----------------------------------
Normally, clean and simple optimizations for widely used codecs are
justified even if they only achieve an overall speedup of 0.1%. These
speedups accumulate and can make a big difference after awhile. Also, if
none of the following factors get worse due to an optimization -- speed,
binary code size, source size, source readability -- and at least one
factor improves, then an optimization is always a good idea even if the
overall gain is less than 0.1%. For obscure codecs that are not often
used, the goal is more toward keeping the code clean, small, and
readable instead of making it 1% faster.
WTF is that function good for ....:
-----------------------------------
The primary purpose of this list is to avoid wasting time optimizing functions
which are rarely used.
put(_no_rnd)_pixels{,_x2,_y2,_xy2}
Used in motion compensation (en/decoding).
avg_pixels{,_x2,_y2,_xy2}
Used in motion compensation of B-frames.
These are less important than the put*pixels functions.
avg_no_rnd_pixels*
unused
pix_abs16x16{,_x2,_y2,_xy2}
Used in motion estimation (encoding) with SAD.
pix_abs8x8{,_x2,_y2,_xy2}
Used in motion estimation (encoding) with SAD of MPEG-4 4MV only.
These are less important than the pix_abs16x16* functions.
put_mspel8_mc* / wmv2_mspel8*
Used only in WMV2.
it is not recommended that you waste your time with these, as WMV2
is an ugly and relatively useless codec.
mpeg4_qpel* / *qpel_mc*
Used in MPEG-4 qpel motion compensation (encoding & decoding).
The qpel8 functions are used only for 4mv,
the avg_* functions are used only for B-frames.
Optimizing them should have a significant impact on qpel
encoding & decoding.
qpel{8,16}_mc??_old_c / *pixels{8,16}_l4
Just used to work around a bug in an old libavcodec encoder version.
Don't optimize them.
add_bytes/diff_bytes
For huffyuv only, optimize if you want a faster ffhuffyuv codec.
get_pixels / diff_pixels
Used for encoding, easy.
clear_blocks
easiest to optimize
gmc
Used for MPEG-4 gmc.
Optimizing this should have a significant effect on the gmc decoding
speed.
gmc1
Used for chroma blocks in MPEG-4 gmc with 1 warp point
(there are 4 luma & 2 chroma blocks per macroblock, so
only 1/3 of the gmc blocks use this, the other 2/3
use the normal put_pixel* code, but only if there is
just 1 warp point).
Note: DivX5 gmc always uses just 1 warp point.
pix_sum
Used for encoding.
hadamard8_diff / sse / sad == pix_norm1 / dct_sad / quant_psnr / rd / bit
Specific compare functions used in encoding, it depends upon the
command line switches which of these are used.
Don't waste your time with dct_sad & quant_psnr, they aren't
really useful.
put_pixels_clamped / add_pixels_clamped
Used for en/decoding in the IDCT, easy.
Note, some optimized IDCTs have the add/put clamped code included and
then put_pixels_clamped / add_pixels_clamped will be unused.
idct/fdct
idct (encoding & decoding)
fdct (encoding)
difficult to optimize
dct_quantize_trellis
Used for encoding with trellis quantization.
difficult to optimize
dct_quantize
Used for encoding.
dct_unquantize_mpeg1
Used in MPEG-1 en/decoding.
dct_unquantize_mpeg2
Used in MPEG-2 en/decoding.
dct_unquantize_h263
Used in MPEG-4/H.263 en/decoding.
Alignment:
Some instructions on some architectures have strict alignment restrictions,
for example most SSE/SSE2 instructions on x86.
The minimum guaranteed alignment is written in the .h files, for example:
void (*put_pixels_clamped)(const int16_t *block/*align 16*/, uint8_t *pixels/*align 8*/, ptrdiff_t stride);
General Tips:
-------------
Use asm loops like:
__asm__(
"1: ....
...
"jump_instruction ....
Do not use C loops:
do{
__asm__(
...
}while()
For x86, mark registers that are clobbered in your asm. This means both
general x86 registers (e.g. eax) as well as XMM registers. This last one is
particularly important on Win64, where xmm6-15 are callee-save, and not
restoring their contents leads to undefined results. In external asm,
you do this by using:
cglobal function_name, num_args, num_regs, num_xmm_regs
In inline asm, you specify clobbered registers at the end of your asm:
__asm__(".." ::: "%eax").
If gcc is not set to support sse (-msse) it will not accept xmm registers
in the clobber list. For that we use two macros to declare the clobbers.
XMM_CLOBBERS should be used when there are other clobbers, for example:
__asm__(".." ::: XMM_CLOBBERS("xmm0",) "eax");
and XMM_CLOBBERS_ONLY should be used when the only clobbers are xmm registers:
__asm__(".." :: XMM_CLOBBERS_ONLY("xmm0"));
Do not expect a compiler to maintain values in your registers between separate
(inline) asm code blocks. It is not required to. For example, this is bad:
__asm__("movdqa %0, %%xmm7" : src);
/* do something */
__asm__("movdqa %%xmm7, %1" : dst);
- first of all, you're assuming that the compiler will not use xmm7 in
between the two asm blocks. It probably won't when you test it, but it's
a poor assumption that will break at some point for some --cpu compiler flag
- secondly, you didn't mark xmm7 as clobbered. If you did, the compiler would
have restored the original value of xmm7 after the first asm block, thus
rendering the combination of the two blocks of code invalid
Code that depends on data in registries being untouched, should be written as
a single __asm__() statement. Ideally, a single function contains only one
__asm__() block.
Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
The latter requires a good optimizing compiler which gcc is not.
When debugging a x86 external asm compilation issue, if lost in the macro
expansions, add DBG=1 to your make command-line: the input file will be
preprocessed, stripped of the debug/empty lines, then compiled, showing the
actual lines causing issues.
Inline asm vs. external asm
---------------------------
Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)
and external asm (.s or .asm files, handled by an assembler such as nasm/yasm)
are accepted in FFmpeg. Which one to use differs per specific case.
- if your code is intended to be inlined in a C function, inline asm is always
better, because external asm cannot be inlined
- if your code calls external functions, external asm is always better
- if your code takes huge and complex structs as function arguments (e.g.
MpegEncContext; note that this is not ideal and is discouraged if there
are alternatives), then inline asm is always better, because predicting
member offsets in complex structs is almost impossible. It's safest to let
the compiler take care of that
- in many cases, both can be used and it just depends on the preference of the
person writing the asm. For new asm, the choice is up to you. For existing
asm, you'll likely want to maintain whatever form it is currently in unless
there is a good reason to change it.
- if, for some reason, you believe that a particular chunk of existing external
asm could be improved upon further if written in inline asm (or the other
way around), then please make the move from external asm <-> inline asm a
separate patch before your patches that actually improve the asm.
Links:
======
http://www.aggregate.org/MAGIC/
x86-specific:
-------------
http://developer.intel.com/design/pentium4/manuals/248966.htm
The IA-32 Intel Architecture Software Developer's Manual, Volume 2:
Instruction Set Reference
http://developer.intel.com/design/pentium4/manuals/245471.htm
http://www.agner.org/assem/
AMD Athlon Processor x86 Code Optimization Guide:
http://www.amd.com/us-en/assets/content_type/white_papers_and_tech_docs/22007.pdf
ARM-specific:
-------------
ARM Architecture Reference Manual (up to ARMv5TE):
http://www.arm.com/community/university/eulaarmarm.html
Procedure Call Standard for the ARM Architecture:
http://www.arm.com/pdfs/aapcs.pdf
Optimization guide for ARM9E (used in Nokia 770 Internet Tablet):
http://infocenter.arm.com/help/topic/com.arm.doc.ddi0240b/DDI0240A.pdf
Optimization guide for ARM11 (used in Nokia N800 Internet Tablet):
http://infocenter.arm.com/help/topic/com.arm.doc.ddi0211j/DDI0211J_arm1136_r1p5_trm.pdf
Optimization guide for Intel XScale (used in Sharp Zaurus PDA):
http://download.intel.com/design/intelxscale/27347302.pdf
Intel Wireless MMX 2 Coprocessor: Programmers Reference Manual
http://download.intel.com/design/intelxscale/31451001.pdf
PowerPC-specific:
-----------------
PowerPC32/AltiVec PIM:
www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPEM.pdf
PowerPC32/AltiVec PEM:
www.freescale.com/files/32bit/doc/ref_manual/ALTIVECPIM.pdf
CELL/SPU:
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/30B3520C93F437AB87257060006FFE5E/$file/Language_Extensions_for_CBEA_2.4.pdf
http://www-01.ibm.com/chips/techlib/techlib.nsf/techdocs/9F820A5FFA3ECE8C8725716A0062585F/$file/CBE_Handbook_v1.1_24APR2007_pub.pdf
GCC asm links:
--------------
official doc but quite ugly
http://gcc.gnu.org/onlinedocs/gcc/Extended-Asm.html
a bit old (note "+" is valid for input-output, even though the next disagrees)
http://www.cs.virginia.edu/~clc5q/gcc-inline-asm.pdf

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@chapter Output Devices
@c man begin OUTPUT DEVICES
Output devices are configured elements in FFmpeg that can write
multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices
are enabled by default. You can list all available ones using the
configure option "--list-outdevs".
You can disable all the output devices using the configure option
"--disable-outdevs", and selectively enable an output device using the
option "--enable-outdev=@var{OUTDEV}", or you can disable a particular
input device using the option "--disable-outdev=@var{OUTDEV}".
The option "-devices" of the ff* tools will display the list of
enabled output devices.
A description of the currently available output devices follows.
@section alsa
ALSA (Advanced Linux Sound Architecture) output device.
@subsection Examples
@itemize
@item
Play a file on default ALSA device:
@example
ffmpeg -i INPUT -f alsa default
@end example
@item
Play a file on soundcard 1, audio device 7:
@example
ffmpeg -i INPUT -f alsa hw:1,7
@end example
@end itemize
@section 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
@code{--enable-libcaca}.
libcaca is a graphics library that outputs text instead of pixels.
For more information about libcaca, check:
@url{http://caca.zoy.org/wiki/libcaca}
@subsection Options
@table @option
@item window_title
Set the CACA window title, if not specified default to the filename
specified for the output device.
@item window_size
Set the CACA window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video.
@item driver
Set display driver.
@item algorithm
Set dithering algorithm. Dithering is necessary
because the picture being rendered has usually far more colours than
the available palette.
The accepted values are listed with @code{-list_dither algorithms}.
@item antialias
Set antialias method. Antialiasing smoothens the rendered
image and avoids the commonly seen staircase effect.
The accepted values are listed with @code{-list_dither antialiases}.
@item charset
Set which characters are going to be used when rendering text.
The accepted values are listed with @code{-list_dither charsets}.
@item color
Set color to be used when rendering text.
The accepted values are listed with @code{-list_dither colors}.
@item list_drivers
If set to @option{true}, print a list of available drivers and exit.
@item list_dither
List available dither options related to the argument.
The argument must be one of @code{algorithms}, @code{antialiases},
@code{charsets}, @code{colors}.
@end table
@subsection Examples
@itemize
@item
The following command shows the @command{ffmpeg} output is an
CACA window, forcing its size to 80x25:
@example
ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
@end example
@item
Show the list of available drivers and exit:
@example
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
@end example
@item
Show the list of available dither colors and exit:
@example
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
@end example
@end itemize
@section 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 @code{--extra-cflags}
and @code{--extra-ldflags}.
On Windows, you need to run the IDL files through @command{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 @command{-list_formats 1}. Audio sample rate is always 48 kHz.
@subsection Options
@table @option
@item list_devices
If set to @option{true}, print a list of devices and exit.
Defaults to @option{false}. This option is deprecated, please use the
@code{-sinks} option of ffmpeg to list the available output devices.
@item list_formats
If set to @option{true}, print a list of supported formats and exit.
Defaults to @option{false}.
@item preroll
Amount of time to preroll video in seconds.
Defaults to @option{0.5}.
@item duplex_mode
Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
Defaults to @samp{unset}.
@item timing_offset
Sets the genlock timing pixel offset on the used output.
Defaults to @samp{unset}.
@end table
@subsection Examples
@itemize
@item
List output devices:
@example
ffmpeg -sinks decklink
@end example
@item
List supported formats:
@example
ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
@end example
@item
Play video clip:
@example
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
@end example
@item
Play video clip with non-standard framerate or video size:
@example
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
@end example
@end itemize
@section fbdev
Linux framebuffer output device.
The Linux framebuffer is a graphic hardware-independent abstraction
layer to show graphics on a computer monitor, typically on the
console. It is accessed through a file device node, usually
@file{/dev/fb0}.
For more detailed information read the file
@file{Documentation/fb/framebuffer.txt} included in the Linux source tree.
@subsection Options
@table @option
@item xoffset
@item yoffset
Set x/y coordinate of top left corner. Default is 0.
@end table
@subsection Examples
Play a file on framebuffer device @file{/dev/fb0}.
Required pixel format depends on current framebuffer settings.
@example
ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
@end example
See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
@section opengl
OpenGL output device.
To enable this output device you need to configure FFmpeg with @code{--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:
@code{AV_DEV_TO_APP_CREATE_WINDOW_BUFFER} - create OpenGL context on current thread.
@code{AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER} - make OpenGL context current.
@code{AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER} - swap buffers.
@code{AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER} - destroy OpenGL context.
Application is also required to inform a device about current resolution by sending @code{AV_APP_TO_DEV_WINDOW_SIZE} message.
@subsection Options
@table @option
@item background
Set background color. Black is a default.
@item no_window
Disables default SDL window when set to non-zero value.
Application must provide OpenGL context and both @code{window_size_cb} and @code{window_swap_buffers_cb} callbacks when set.
@item window_title
Set the SDL window title, if not specified default to the filename specified for the output device.
Ignored when @option{no_window} is set.
@item 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 @option{no_window} is not set.
@end table
@subsection Examples
Play a file on SDL window using OpenGL rendering:
@example
ffmpeg -i INPUT -f opengl "window title"
@end example
@section oss
OSS (Open Sound System) output device.
@section pulse
PulseAudio output device.
To enable this output device you need to configure FFmpeg with @code{--enable-libpulse}.
More information about PulseAudio can be found on @url{http://www.pulseaudio.org}
@subsection Options
@table @option
@item server
Connect to a specific PulseAudio server, specified by an IP address.
Default server is used when not provided.
@item name
Specify the application name PulseAudio will use when showing active clients,
by default it is the @code{LIBAVFORMAT_IDENT} string.
@item stream_name
Specify the stream name PulseAudio will use when showing active streams,
by default it is set to the specified output name.
@item device
Specify the device to use. Default device is used when not provided.
List of output devices can be obtained with command @command{pactl list sinks}.
@item buffer_size
@item buffer_duration
Control the size and duration of the PulseAudio buffer. A small buffer
gives more control, but requires more frequent updates.
@option{buffer_size} specifies size in bytes while
@option{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.
@item prebuf
Specify pre-buffering size in bytes. The server does not start with
playback before at least @option{prebuf} bytes are available in the
buffer. By default this option is initialized to the same value as
@option{buffer_size} or @option{buffer_duration} (whichever is bigger).
@item minreq
Specify minimum request size in bytes. The server does not request less
than @option{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.
@end table
@subsection Examples
Play a file on default device on default server:
@example
ffmpeg -i INPUT -f pulse "stream name"
@end example
@section sdl
SDL (Simple DirectMedia Layer) output device.
"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:
@url{http://www.libsdl.org/}
@subsection Options
@table @option
@item window_title
Set the SDL window title, if not specified default to the filename
specified for the output device.
@item icon_title
Set the name of the iconified SDL window, if not specified it is set
to the same value of @var{window_title}.
@item window_size
Set the SDL window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation.
If not specified it defaults to the size of the input video,
downscaled according to the aspect ratio.
@item window_x
@item window_y
Set the position of the window on the screen.
@item window_fullscreen
Set fullscreen mode when non-zero value is provided.
Default value is zero.
@item 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)
@end table
@subsection Interactive commands
The window created by the device can be controlled through the
following interactive commands.
@table @key
@item q, ESC
Quit the device immediately.
@end table
@subsection Examples
The following command shows the @command{ffmpeg} output is an
SDL window, forcing its size to the qcif format:
@example
ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
@end example
@section sndio
sndio audio output device.
@section v4l2
Video4Linux2 output device.
@section xv
XV (XVideo) output device.
This output device allows one to show a video stream in a X Window System
window.
@subsection Options
@table @option
@item display_name
Specify the hardware display name, which determines the display and
communications domain to be used.
The display name or DISPLAY environment variable can be a string in
the format @var{hostname}[:@var{number}[.@var{screen_number}]].
@var{hostname} specifies the name of the host machine on which the
display is physically attached. @var{number} specifies the number of
the display server on that host machine. @var{screen_number} specifies
the screen to be used on that server.
If unspecified, it defaults to the value of the DISPLAY environment
variable.
For example, @code{dual-headed:0.1} would specify screen 1 of display
0 on the machine named ``dual-headed''.
Check the X11 specification for more detailed information about the
display name format.
@item window_id
When set to non-zero value then device doesn't create new window,
but uses existing one with provided @var{window_id}. By default
this options is set to zero and device creates its own window.
@item window_size
Set the created window size, can be a string of the form
@var{width}x@var{height} or a video size abbreviation. If not
specified it defaults to the size of the input video.
Ignored when @var{window_id} is set.
@item window_x
@item window_y
Set the X and Y window offsets for the created window. They are both
set to 0 by default. The values may be ignored by the window manager.
Ignored when @var{window_id} is set.
@item window_title
Set the window title, if not specified default to the filename
specified for the output device. Ignored when @var{window_id} is set.
@end table
For more information about XVideo see @url{http://www.x.org/}.
@subsection Examples
@itemize
@item
Decode, display and encode video input with @command{ffmpeg} at the
same time:
@example
ffmpeg -i INPUT OUTPUT -f xv display
@end example
@item
Decode and display the input video to multiple X11 windows:
@example
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
@end example
@end itemize
@c man end OUTPUT DEVICES

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Patchwork states
NEW: Initial state of new patches
Accepted: The patch was pushed to the main master repository
Rejected: The patch has been rejected
Withdrawn: The patch was withdrawn by the author
Not Applicable: The patch does not apply to the main master repository
Superseded: A newer version of the patch has been posted
Changes Requested: The patch has been or is under review and changes have been requested
RFC: The patch is not intended to be applied but only for comments

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\input texinfo @c -*- texinfo -*-
@documentencoding UTF-8
@settitle Platform Specific Information
@titlepage
@center @titlefont{Platform Specific Information}
@end titlepage
@top
@contents
@chapter Unix-like
Some parts of FFmpeg cannot be built with version 2.15 of the GNU
assembler which is still provided by a few AMD64 distributions. To
make sure your compiler really uses the required version of gas
after a binutils upgrade, run:
@example
$(gcc -print-prog-name=as) --version
@end example
If not, then you should install a different compiler that has no
hard-coded path to gas. In the worst case pass @code{--disable-asm}
to configure.
@section Advanced linking configuration
If you compiled FFmpeg libraries statically and you want to use them to
build your own shared library, you may need to force PIC support (with
@code{--enable-pic} during FFmpeg configure) and add the following option
to your project LDFLAGS:
@example
-Wl,-Bsymbolic
@end example
If your target platform requires position independent binaries, you should
pass the correct linking flag (e.g. @code{-pie}) to @code{--extra-ldexeflags}.
@section BSD
BSD make will not build FFmpeg, you need to install and use GNU Make
(@command{gmake}).
@section (Open)Solaris
GNU Make is required to build FFmpeg, so you have to invoke (@command{gmake}),
standard Solaris Make will not work. When building with a non-c99 front-end
(gcc, generic suncc) add either @code{--extra-libs=/usr/lib/values-xpg6.o}
or @code{--extra-libs=/usr/lib/64/values-xpg6.o} to the configure options
since the libc is not c99-compliant by default. The probes performed by
configure may raise an exception leading to the death of configure itself
due to a bug in the system shell. Simply invoke a different shell such as
bash directly to work around this:
@example
bash ./configure
@end example
@anchor{Darwin}
@section Darwin (Mac OS X, iPhone)
The toolchain provided with Xcode is sufficient to build the basic
unaccelerated code.
Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{https://github.com/FFmpeg/gas-preprocessor} or
@url{https://github.com/yuvi/gas-preprocessor}(currently outdated) to build the optimized
assembly functions. Put the Perl script somewhere
in your PATH, FFmpeg's configure will pick it up automatically.
Mac OS X on amd64 and x86 requires @command{nasm} to build most of the
optimized assembly functions. @uref{http://www.finkproject.org/, Fink},
@uref{https://wiki.gentoo.org/wiki/Project:Prefix, Gentoo Prefix},
@uref{https://mxcl.github.com/homebrew/, Homebrew}
or @uref{http://www.macports.org, MacPorts} can easily provide it.
@chapter DOS
Using a cross-compiler is preferred for various reasons.
@url{http://www.delorie.com/howto/djgpp/linux-x-djgpp.html}
@chapter OS/2
For information about compiling FFmpeg on OS/2 see
@url{http://www.edm2.com/index.php/FFmpeg}.
@chapter Windows
To get help and instructions for building FFmpeg under Windows, check out
the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
@section Native Windows compilation using MinGW or MinGW-w64
FFmpeg can be built to run natively on Windows using the MinGW-w64
toolchain. Install the latest versions of MSYS2 and MinGW-w64 from
@url{http://msys2.github.io/} and/or @url{http://mingw-w64.sourceforge.net/}.
You can find detailed installation instructions in the download section and
the FAQ.
Notes:
@itemize
@item Building for the MSYS environment is discouraged, MSYS2 provides a full
MinGW-w64 environment through @file{mingw64_shell.bat} or
@file{mingw32_shell.bat} that should be used instead of the environment
provided by @file{msys2_shell.bat}.
@item Building using MSYS2 can be sped up by disabling implicit rules in the
Makefile by calling @code{make -r} instead of plain @code{make}. This
speed up is close to non-existent for normal one-off builds and is only
noticeable when running make for a second time (for example during
@code{make install}).
@item In order to compile FFplay, you must have the MinGW development library
of @uref{http://www.libsdl.org/, SDL} and @code{pkg-config} installed.
@item By using @code{./configure --enable-shared} when configuring FFmpeg,
you can build the FFmpeg libraries (e.g. libavutil, libavcodec,
libavformat) as DLLs.
@end itemize
@subsection Native Windows compilation using MSYS2
The MSYS2 MinGW-w64 environment provides ready to use toolchains and dependencies
through @command{pacman}.
Make sure to use @file{mingw64_shell.bat} or @file{mingw32_shell.bat} to have
the correct MinGW-w64 environment. The default install provides shortcuts to
them under @command{MinGW-w64 Win64 Shell} and @command{MinGW-w64 Win32 Shell}.
@example
# normal msys2 packages
pacman -S make pkgconf diffutils
# mingw-w64 packages and toolchains
pacman -S mingw-w64-x86_64-nasm mingw-w64-x86_64-gcc mingw-w64-x86_64-SDL2
@end example
To target 32 bits replace @code{x86_64} with @code{i686} in the command above.
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
FFmpeg can be built with MSVC 2013 or later.
You will need the following prerequisites:
@itemize
@item @uref{http://msys2.github.io/, MSYS2}
@item @uref{http://www.nasm.us/, NASM}
(Also available via MSYS2's package manager.)
@end itemize
To set up a proper environment in MSYS2, you need to run @code{msys_shell.bat} from
the Visual Studio or Intel Compiler command prompt.
Place @code{yasm.exe} somewhere in your @code{PATH}.
Next, make sure any other headers and libs you want to use, such as zlib, are
located in a spot that the compiler can see. Do so by modifying the @code{LIB}
and @code{INCLUDE} environment variables to include the @strong{Windows-style}
paths to these directories. Alternatively, you can try to use the
@code{--extra-cflags}/@code{--extra-ldflags} configure options.
Finally, run:
@example
For MSVC:
./configure --toolchain=msvc
For ICL:
./configure --toolchain=icl
make
make install
@end example
If you wish to compile shared libraries, add @code{--enable-shared} to your
configure options. Note that due to the way MSVC and ICL handle DLL imports and
exports, you cannot compile static and shared libraries at the same time, and
enabling shared libraries will automatically disable the static ones.
Notes:
@itemize
@item If you wish to build with zlib support, you will have to grab a compatible
zlib binary from somewhere, with an MSVC import lib, or if you wish to link
statically, you can follow the instructions below to build a compatible
@code{zlib.lib} with MSVC. Regardless of which method you use, you must still
follow step 3, or compilation will fail.
@enumerate
@item Grab the @uref{http://zlib.net/, zlib sources}.
@item Edit @code{win32/Makefile.msc} so that it uses -MT instead of -MD, since
this is how FFmpeg is built as well.
@item Edit @code{zconf.h} and remove its inclusion of @code{unistd.h}. This gets
erroneously included when building FFmpeg.
@item Run @code{nmake -f win32/Makefile.msc}.
@item Move @code{zlib.lib}, @code{zconf.h}, and @code{zlib.h} to somewhere MSVC
can see.
@end enumerate
@item FFmpeg has been tested with the following on i686 and x86_64:
@itemize
@item Visual Studio 2013 Pro and Express
@item Intel Composer XE 2013
@item Intel Composer XE 2013 SP1
@end itemize
Anything else is not officially supported.
@end itemize
@subsection Linking to FFmpeg with Microsoft Visual C++
If you plan to link with MSVC-built static libraries, you will need
to make sure you have @code{Runtime Library} set to
@code{Multi-threaded (/MT)} in your project's settings.
You will need to define @code{inline} to something MSVC understands:
@example
#define inline __inline
@end example
Also note, that as stated in @strong{Microsoft Visual C++}, you will need
an MSVC-compatible @uref{http://code.google.com/p/msinttypes/, inttypes.h}.
If you plan on using import libraries created by dlltool, you must
set @code{References} to @code{No (/OPT:NOREF)} under the linker optimization
settings, otherwise the resulting binaries will fail during runtime.
This is not required when using import libraries generated by @code{lib.exe}.
This issue is reported upstream at
@url{http://sourceware.org/bugzilla/show_bug.cgi?id=12633}.
To create import libraries that work with the @code{/OPT:REF} option
(which is enabled by default in Release mode), follow these steps:
@enumerate
@item Open the @emph{Visual Studio Command Prompt}.
Alternatively, in a normal command line prompt, call @file{vcvars32.bat}
which sets up the environment variables for the Visual C++ tools
(the standard location for this file is something like
@file{C:\Program Files (x86_\Microsoft Visual Studio 10.0\VC\bin\vcvars32.bat}).
@item Enter the @file{bin} directory where the created LIB and DLL files
are stored.
@item Generate new import libraries with @command{lib.exe}:
@example
lib /machine:i386 /def:..\lib\foo-version.def /out:foo.lib
@end example
Replace @code{foo-version} and @code{foo} with the respective library names.
@end enumerate
@anchor{Cross compilation for Windows with Linux}
@section Cross compilation for Windows with Linux
You must use the MinGW cross compilation tools available at
@url{http://www.mingw.org/}.
Then configure FFmpeg with the following options:
@example
./configure --target-os=mingw32 --cross-prefix=i386-mingw32msvc-
@end example
(you can change the cross-prefix according to the prefix chosen for the
MinGW tools).
Then you can easily test FFmpeg with @uref{http://www.winehq.com/, Wine}.
@section Compilation under Cygwin
Please use Cygwin 1.7.x as the obsolete 1.5.x Cygwin versions lack
llrint() in its C library.
Install your Cygwin with all the "Base" packages, plus the
following "Devel" ones:
@example
binutils, gcc4-core, make, git, mingw-runtime, texinfo
@end example
In order to run FATE you will also need the following "Utils" packages:
@example
diffutils
@end example
If you want to build FFmpeg with additional libraries, download Cygwin
"Devel" packages for Ogg and Vorbis from any Cygwin packages repository:
@example
libogg-devel, libvorbis-devel
@end example
These library packages are only available from
@uref{http://sourceware.org/cygwinports/, Cygwin Ports}:
@example
yasm, libSDL-devel, libgsm-devel, libmp3lame-devel,
speex-devel, libtheora-devel, libxvidcore-devel
@end example
The recommendation for x264 is to build it from source, as it evolves too
quickly for Cygwin Ports to be up to date.
@section Crosscompilation for Windows under Cygwin
With Cygwin you can create Windows binaries that do not need the cygwin1.dll.
Just install your Cygwin as explained before, plus these additional
"Devel" packages:
@example
gcc-mingw-core, mingw-runtime, mingw-zlib
@end example
and add some special flags to your configure invocation.
For a static build run
@example
./configure --target-os=mingw32 --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
and for a build with shared libraries
@example
./configure --target-os=mingw32 --enable-shared --disable-static --extra-cflags=-mno-cygwin --extra-libs=-mno-cygwin
@end example
@bye

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

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

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@chapter Resampler Options
@c man begin RESAMPLER OPTIONS
The audio resampler supports the following named options.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, @var{option}=@var{value} for the aresample filter,
by setting the value explicitly in the
@code{SwrContext} options or using the @file{libavutil/opt.h} API for
programmatic use.
@table @option
@item ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{in_channel_layout} is set.
@item och, out_channel_count
Set the number of output channels. Default value is 0. Setting this
value is not mandatory if the corresponding channel layout
@option{out_channel_layout} is set.
@item uch, used_channel_count
Set the number of used input channels. Default value is 0. This option is
only used for special remapping.
@item isr, in_sample_rate
Set the input sample rate. Default value is 0.
@item osr, out_sample_rate
Set the output sample rate. Default value is 0.
@item isf, in_sample_fmt
Specify the input sample format. It is set by default to @code{none}.
@item osf, out_sample_fmt
Specify the output sample format. It is set by default to @code{none}.
@item tsf, internal_sample_fmt
Set the internal sample format. Default value is @code{none}.
This will automatically be chosen when it is not explicitly set.
@item icl, in_channel_layout
@item ocl, out_channel_layout
Set the input/output channel layout.
See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the required syntax.
@item clev, center_mix_level
Set the center mix level. It is a value expressed in deciBel, and must be
in the interval [-32,32].
@item slev, surround_mix_level
Set the surround mix level. It is a value expressed in deciBel, and must
be in the interval [-32,32].
@item lfe_mix_level
Set LFE mix into non LFE level. It is used when there is a LFE input but no
LFE output. It is a value expressed in deciBel, and must
be in the interval [-32,32].
@item rmvol, rematrix_volume
Set rematrix volume. Default value is 1.0.
@item rematrix_maxval
Set maximum output value for rematrixing.
This can be used to prevent clipping vs. preventing volume reduction.
A value of 1.0 prevents clipping.
@item flags, swr_flags
Set flags used by the converter. Default value is 0.
It supports the following individual flags:
@table @option
@item res
force resampling, this flag forces resampling to be used even when the
input and output sample rates match.
@end table
@item dither_scale
Set the dither scale. Default value is 1.
@item dither_method
Set dither method. Default value is 0.
Supported values:
@table @samp
@item rectangular
select rectangular dither
@item triangular
select triangular dither
@item triangular_hp
select triangular dither with high pass
@item lipshitz
select Lipshitz noise shaping dither.
@item shibata
select Shibata noise shaping dither.
@item low_shibata
select low Shibata noise shaping dither.
@item high_shibata
select high Shibata noise shaping dither.
@item f_weighted
select f-weighted noise shaping dither
@item modified_e_weighted
select modified-e-weighted noise shaping dither
@item improved_e_weighted
select improved-e-weighted noise shaping dither
@end table
@item resampler
Set resampling engine. Default value is swr.
Supported values:
@table @samp
@item swr
select the native SW Resampler; filter options precision and cheby are not
applicable in this case.
@item soxr
select the SoX Resampler (where available); compensation, and filter options
filter_size, phase_shift, exact_rational, filter_type & kaiser_beta, are not
applicable in this case.
@end table
@item filter_size
For swr only, set resampling filter size, default value is 32.
@item phase_shift
For swr only, set resampling phase shift, default value is 10, and must be in
the interval [0,30].
@item linear_interp
Use linear interpolation when enabled (the default). Disable it if you want
to preserve speed instead of quality when exact_rational fails.
@item 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 @code{1 << phase_shift},
the phase_count will be @code{1 << phase_shift} as fallback. Default is enabled.
@item cutoff
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float
value between 0 and 1. Default value is 0.97 with swr, and 0.91 with soxr
(which, with a sample-rate of 44100, preserves the entire audio band to 20kHz).
@item precision
For soxr only, the precision in bits to which the resampled signal will be
calculated. The default value of 20 (which, with suitable dithering, is
appropriate for a destination bit-depth of 16) gives SoX's 'High Quality'; a
value of 28 gives SoX's 'Very High Quality'.
@item cheby
For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
approximation for 'irrational' ratios. Default value is 0.
@item async
For swr only, simple 1 parameter audio sync to timestamps using stretching,
squeezing, filling and trimming. Setting this to 1 will enable filling and
trimming, larger values represent the maximum amount in samples that the data
may be stretched or squeezed for each second.
Default value is 0, thus no compensation is applied to make the samples match
the audio timestamps.
@item first_pts
For swr only, assume the first pts should be this value. The time unit is 1 / sample rate.
This allows for padding/trimming at the start of stream. By default, no
assumption is made about the first frame's expected pts, so no padding or
trimming is done. For example, this could be set to 0 to pad the beginning with
silence if an audio stream starts after the video stream or to trim any samples
with a negative pts due to encoder delay.
@item min_comp
For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger stretching/squeezing/filling or trimming of the
data to make it match the timestamps. The default is that
stretching/squeezing/filling and trimming is disabled
(@option{min_comp} = @code{FLT_MAX}).
@item min_hard_comp
For swr only, set the minimum difference between timestamps and audio data (in
seconds) to trigger adding/dropping samples to make it match the
timestamps. This option effectively is a threshold to select between
hard (trim/fill) and soft (squeeze/stretch) compensation. Note that
all compensation is by default disabled through @option{min_comp}.
The default is 0.1.
@item comp_duration
For swr only, set duration (in seconds) over which data is stretched/squeezed
to make it match the timestamps. Must be a non-negative double float value,
default value is 1.0.
@item max_soft_comp
For swr only, set maximum factor by which data is stretched/squeezed to make it
match the timestamps. Must be a non-negative double float value, default value
is 0.
@item matrix_encoding
Select matrixed stereo encoding.
It accepts the following values:
@table @samp
@item none
select none
@item dolby
select Dolby
@item dplii
select Dolby Pro Logic II
@end table
Default value is @code{none}.
@item filter_type
For swr only, select resampling filter type. This only affects resampling
operations.
It accepts the following values:
@table @samp
@item cubic
select cubic
@item blackman_nuttall
select Blackman Nuttall windowed sinc
@item kaiser
select Kaiser windowed sinc
@end table
@item kaiser_beta
For swr only, set Kaiser window beta value. Must be a double float value in the
interval [2,16], default value is 9.
@item output_sample_bits
For swr only, set number of used output sample bits for dithering. Must be an integer in the
interval [0,64], default value is 0, which means it's not used.
@end table
@c man end RESAMPLER OPTIONS

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@anchor{scaler_options}
@chapter Scaler Options
@c man begin SCALER OPTIONS
The video scaler supports the following named options.
Options may be set by specifying -@var{option} @var{value} in the
FFmpeg tools, with a few API-only exceptions noted below.
For programmatic use, they can be set explicitly in the
@code{SwsContext} options or through the @file{libavutil/opt.h} API.
@table @option
@anchor{sws_flags}
@item sws_flags
Set the scaler flags. This is also used to set the scaling
algorithm. Only a single algorithm should be selected. Default
value is @samp{bicubic}.
It accepts the following values:
@table @samp
@item fast_bilinear
Select fast bilinear scaling algorithm.
@item bilinear
Select bilinear scaling algorithm.
@item bicubic
Select bicubic scaling algorithm.
@item experimental
Select experimental scaling algorithm.
@item neighbor
Select nearest neighbor rescaling algorithm.
@item area
Select averaging area rescaling algorithm.
@item bicublin
Select bicubic scaling algorithm for the luma component, bilinear for
chroma components.
@item gauss
Select Gaussian rescaling algorithm.
@item sinc
Select sinc rescaling algorithm.
@item lanczos
Select Lanczos rescaling algorithm. The default width (alpha) is 3 and can be
changed by setting @code{param0}.
@item spline
Select natural bicubic spline rescaling algorithm.
@item print_info
Enable printing/debug logging.
@item accurate_rnd
Enable accurate rounding.
@item full_chroma_int
Enable full chroma interpolation.
@item full_chroma_inp
Select full chroma input.
@item bitexact
Enable bitexact output.
@end table
@item srcw @var{(API only)}
Set source width.
@item srch @var{(API only)}
Set source height.
@item dstw @var{(API only)}
Set destination width.
@item dsth @var{(API only)}
Set destination height.
@item src_format @var{(API only)}
Set source pixel format (must be expressed as an integer).
@item dst_format @var{(API only)}
Set destination pixel format (must be expressed as an integer).
@item src_range @var{(boolean)}
If value is set to @code{1}, indicates source is full range. Default value is
@code{0}, which indicates source is limited range.
@item dst_range @var{(boolean)}
If value is set to @code{1}, enable full range for destination. Default value
is @code{0}, which enables limited range.
@anchor{sws_params}
@item param0, param1
Set scaling algorithm parameters. The specified values are specific of
some scaling algorithms and ignored by others. The specified values
are floating point number values.
@item sws_dither
Set the dithering algorithm. Accepts one of the following
values. Default value is @samp{auto}.
@table @samp
@item auto
automatic choice
@item none
no dithering
@item bayer
bayer dither
@item ed
error diffusion dither
@item a_dither
arithmetic dither, based using addition
@item x_dither
arithmetic dither, based using xor (more random/less apparent patterning that
a_dither).
@end table
@item alphablend
Set the alpha blending to use when the input has alpha but the output does not.
Default value is @samp{none}.
@table @samp
@item uniform_color
Blend onto a uniform background color
@item checkerboard
Blend onto a checkerboard
@item none
No blending
@end table
@end table
@c man end SCALER OPTIONS

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

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

98
externals/ffmpeg/doc/swscale.txt vendored Executable file
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The official guide to swscale for confused developers.
========================================================
Current (simplified) Architecture:
---------------------------------
Input
v
_______OR_________
/ \
/ \
special converter [Input to YUV converter]
| |
| (8-bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 )
| |
| v
| Horizontal scaler
| |
| (15-bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 )
| |
| v
| Vertical scaler and output converter
| |
v v
output
Swscale has 2 scaler paths. Each side must be capable of handling
slices, that is, consecutive non-overlapping rectangles of dimension
(0,slice_top) - (picture_width, slice_bottom).
special converter
These generally are unscaled converters of common
formats, like YUV 4:2:0/4:2:2 -> RGB12/15/16/24/32. Though it could also
in principle contain scalers optimized for specific common cases.
Main path
The main path is used when no special converter can be used. The code
is designed as a destination line pull architecture. That is, for each
output line the vertical scaler pulls lines from a ring buffer. When
the ring buffer does not contain the wanted line, then it is pulled from
the input slice through the input converter and horizontal scaler.
The result is also stored in the ring buffer to serve future vertical
scaler requests.
When no more output can be generated because lines from a future slice
would be needed, then all remaining lines in the current slice are
converted, horizontally scaled and put in the ring buffer.
[This is done for luma and chroma, each with possibly different numbers
of lines per picture.]
Input to YUV Converter
When the input to the main path is not planar 8 bits per component YUV or
8-bit gray, it is converted to planar 8-bit YUV. Two sets of converters
exist for this currently: One performs horizontal downscaling by 2
before the conversion, the other leaves the full chroma resolution,
but is slightly slower. The scaler will try to preserve full chroma
when the output uses it. It is possible to force full chroma with
SWS_FULL_CHR_H_INP even for cases where the scaler thinks it is useless.
Horizontal scaler
There are several horizontal scalers. A special case worth mentioning is
the fast bilinear scaler that is made of runtime-generated MMXEXT code
using specially tuned pshufw instructions.
The remaining scalers are specially-tuned for various filter lengths.
They scale 8-bit unsigned planar data to 16-bit signed planar data.
Future >8 bits per component inputs will need to add a new horizontal
scaler that preserves the input precision.
Vertical scaler and output converter
There is a large number of combined vertical scalers + output converters.
Some are:
* unscaled output converters
* unscaled output converters that average 2 chroma lines
* bilinear converters (C, MMX and accurate MMX)
* arbitrary filter length converters (C, MMX and accurate MMX)
And
* Plain C 8-bit 4:2:2 YUV -> RGB converters using LUTs
* Plain C 17-bit 4:4:4 YUV -> RGB converters using multiplies
* MMX 11-bit 4:2:2 YUV -> RGB converters
* Plain C 16-bit Y -> 16-bit gray
...
RGB with less than 8 bits per component uses dither to improve the
subjective quality and low-frequency accuracy.
Filter coefficients:
--------------------
There are several different scalers (bilinear, bicubic, lanczos, area,
sinc, ...). Their coefficients are calculated in initFilter().
Horizontal filter coefficients have a 1.0 point at 1 << 14, vertical ones at
1 << 12. The 1.0 points have been chosen to maximize precision while leaving
a little headroom for convolutional filters like sharpening filters and
minimizing SIMD instructions needed to apply them.
It would be trivial to use a different 1.0 point if some specific scaler
would benefit from it.
Also, as already hinted at, initFilter() accepts an optional convolutional
filter as input that can be used for contrast, saturation, blur, sharpening
shift, chroma vs. luma shift, ...

128
externals/ffmpeg/doc/t2h.init vendored Executable file
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# Init file for texi2html.
# This is deprecated, and the makeinfo/texi2any version is doc/t2h.pm
# no horiz rules between sections
$end_section = \&FFmpeg_end_section;
sub FFmpeg_end_section($$)
{
}
my $TEMPLATE_HEADER1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="utf-8" />
<meta http-equiv="X-UA-Compatible" content="IE=edge" />
<title>FFmpeg documentation</title>
<link rel="stylesheet" href="bootstrap.min.css" />
<link rel="stylesheet" href="style.min.css" />
EOT
my $TEMPLATE_HEADER2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
</head>
<body>
<div class="container">
EOT
my $TEMPLATE_FOOTER = $ENV{"FFMPEG_FOOTER"} || <<EOT;
</div>
</body>
</html>
EOT
$SMALL_RULE = '';
$BODYTEXT = '';
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
my $program_string = defined &T2H_DEFAULT_program_string ?
T2H_DEFAULT_program_string() : program_string();
print $fh '<footer class="footer pagination-right">' . "\n";
print $fh '<span class="label label-info">' . $program_string;
print $fh "</span></footer></div></div></body>\n";
}
$float = \&FFmpeg_float;
sub FFmpeg_float($$$$)
{
my $text = shift;
my $float = shift;
my $caption = shift;
my $shortcaption = shift;
my $label = '';
if (exists($float->{'id'}))
{
$label = &$anchor($float->{'id'});
}
my $class = '';
my $subject = '';
if ($caption =~ /NOTE/)
{
$class = "alert alert-info";
}
elsif ($caption =~ /IMPORTANT/)
{
$class = "alert alert-warning";
}
return '<div class="float ' . $class . '">' . "$label\n" . $text . '</div>';
}
$print_page_head = \&FFmpeg_print_page_head;
sub FFmpeg_print_page_head($$)
{
my $fh = shift;
my $longtitle = "$Texi2HTML::THISDOC{'fulltitle_no_texi'}";
$longtitle .= ": $Texi2HTML::NO_TEXI{'This'}" if exists $Texi2HTML::NO_TEXI{'This'};
my $description = $DOCUMENT_DESCRIPTION;
$description = $longtitle if (!defined($description));
$description = "<meta name=\"description\" content=\"$description\">" if
($description ne '');
$description = $Texi2HTML::THISDOC{'documentdescription'} if (defined($Texi2HTML::THISDOC{'documentdescription'}));
my $encoding = '';
$encoding = "<meta http-equiv=\"Content-Type\" content=\"text/html; charset=$ENCODING\">" if (defined($ENCODING) and ($ENCODING ne ''));
$longtitle =~ s/Documentation.*//g;
$longtitle = "FFmpeg documentation : " . $longtitle;
print $fh <<EOT;
$TEMPLATE_HEADER1
$description
<meta name="keywords" content="$longtitle">
<meta name="Generator" content="$Texi2HTML::THISDOC{program}">
$Texi2HTML::THISDOC{'copying'}<!-- Created on $Texi2HTML::THISDOC{today} by $Texi2HTML::THISDOC{program} -->
<!--
$Texi2HTML::THISDOC{program_authors}
-->
$encoding
$TEMPLATE_HEADER2
EOT
}
$print_page_foot = \&FFmpeg_print_page_foot;
sub FFmpeg_print_page_foot($$)
{
my $fh = shift;
print $fh <<EOT;
$TEMPLATE_FOOTER
EOT
}
# declare encoding in header
$IN_ENCODING = $ENCODING = "utf-8";
# no navigation elements
$SECTION_NAVIGATION = 0;
# the same for texi2html 5.0
$HEADERS = 0;
# TOC and Chapter headings link
$TOC_LINKS = 1;
# print the TOC where @contents is used
$INLINE_CONTENTS = 1;

339
externals/ffmpeg/doc/t2h.pm vendored Executable file
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# makeinfo HTML output init file
#
# Copyright (c) 2011, 2012 Free Software Foundation, Inc.
# Copyright (c) 2014 Andreas Cadhalpun
# Copyright (c) 2014 Tiancheng "Timothy" Gu
#
# This file is part of FFmpeg.
#
# FFmpeg 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.
#
# FFmpeg is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
#
# You should have received a copy of the GNU General Public
# License along with FFmpeg; if not, write to the Free Software
# Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
# no navigation elements
set_from_init_file('HEADERS', 0);
sub ffmpeg_heading_command($$$$$)
{
my $self = shift;
my $cmdname = shift;
my $command = shift;
my $args = shift;
my $content = shift;
my $result = '';
# not clear that it may really happen
if ($self->in_string) {
$result .= $self->command_string($command) ."\n" if ($cmdname ne 'node');
$result .= $content if (defined($content));
return $result;
}
my $element_id = $self->command_id($command);
$result .= "<a name=\"$element_id\"></a>\n"
if (defined($element_id) and $element_id ne '');
print STDERR "Process $command "
.Texinfo::Structuring::_print_root_command_texi($command)."\n"
if ($self->get_conf('DEBUG'));
my $element;
if ($Texinfo::Common::root_commands{$command->{'cmdname'}}
and $command->{'parent'}
and $command->{'parent'}->{'type'}
and $command->{'parent'}->{'type'} eq 'element') {
$element = $command->{'parent'};
}
if ($element) {
$result .= &{$self->{'format_element_header'}}($self, $cmdname,
$command, $element);
}
my $heading_level;
# node is used as heading if there is nothing else.
if ($cmdname eq 'node') {
if (!$element or (!$element->{'extra'}->{'section'}
and $element->{'extra'}->{'node'}
and $element->{'extra'}->{'node'} eq $command
# bogus node may not have been normalized
and defined($command->{'extra'}->{'normalized'}))) {
if ($command->{'extra'}->{'normalized'} eq 'Top') {
$heading_level = 0;
} else {
$heading_level = 3;
}
}
} else {
$heading_level = $command->{'level'};
}
my $heading = $self->command_text($command);
# $heading not defined may happen if the command is a @node, for example
# if there is an error in the node.
if (defined($heading) and $heading ne '' and defined($heading_level)) {
if ($Texinfo::Common::root_commands{$cmdname}
and $Texinfo::Common::sectioning_commands{$cmdname}) {
my $content_href = $self->command_contents_href($command, 'contents',
$self->{'current_filename'});
if ($content_href) {
my $this_href = $content_href =~ s/^\#toc-/\#/r;
$heading .= '<span class="pull-right">'.
'<a class="anchor hidden-xs" '.
"href=\"$this_href\" aria-hidden=\"true\">".
($ENV{"FA_ICONS"} ? '<i class="fa fa-link"></i>'
: '#').
'</a> '.
'<a class="anchor hidden-xs"'.
"href=\"$content_href\" aria-hidden=\"true\">".
($ENV{"FA_ICONS"} ? '<i class="fa fa-navicon"></i>'
: 'TOC').
'</a>'.
'</span>';
}
}
if ($self->in_preformatted()) {
$result .= $heading."\n";
} else {
# if the level was changed, set the command name right
if ($cmdname ne 'node'
and $heading_level ne $Texinfo::Common::command_structuring_level{$cmdname}) {
$cmdname
= $Texinfo::Common::level_to_structuring_command{$cmdname}->[$heading_level];
}
$result .= &{$self->{'format_heading_text'}}(
$self, $cmdname, $heading,
$heading_level +
$self->get_conf('CHAPTER_HEADER_LEVEL') - 1, $command);
}
}
$result .= $content if (defined($content));
return $result;
}
foreach my $command (keys(%Texinfo::Common::sectioning_commands), 'node') {
texinfo_register_command_formatting($command, \&ffmpeg_heading_command);
}
# print the TOC where @contents is used
set_from_init_file('INLINE_CONTENTS', 1);
# make chapters <h2>
set_from_init_file('CHAPTER_HEADER_LEVEL', 2);
# Do not add <hr>
set_from_init_file('DEFAULT_RULE', '');
set_from_init_file('BIG_RULE', '');
# Customized file beginning
sub ffmpeg_begin_file($$$)
{
my $self = shift;
my $filename = shift;
my $element = shift;
my $command;
if ($element and $self->get_conf('SPLIT')) {
$command = $self->element_command($element);
}
my ($title, $description, $encoding, $date, $css_lines,
$doctype, $bodytext, $copying_comment, $after_body_open,
$extra_head, $program_and_version, $program_homepage,
$program, $generator) = $self->_file_header_informations($command);
my $links = $self->_get_links ($filename, $element);
my $head1 = $ENV{"FFMPEG_HEADER1"} || <<EOT;
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd">
<html>
<!-- Created by $program_and_version, $program_homepage -->
<head>
<meta charset="utf-8">
<title>
EOT
my $head_title = <<EOT;
$title
EOT
my $head2 = $ENV{"FFMPEG_HEADER2"} || <<EOT;
</title>
<meta name="viewport" content="width=device-width,initial-scale=1.0">
<link rel="stylesheet" type="text/css" href="bootstrap.min.css">
<link rel="stylesheet" type="text/css" href="style.min.css">
</head>
<body>
<div class="container">
<h1>
EOT
my $head3 = $ENV{"FFMPEG_HEADER3"} || <<EOT;
</h1>
EOT
return $head1 . $head_title . $head2 . $head_title . $head3;
}
texinfo_register_formatting_function('begin_file', \&ffmpeg_begin_file);
sub ffmpeg_program_string($)
{
my $self = shift;
if (defined($self->get_conf('PROGRAM'))
and $self->get_conf('PROGRAM') ne ''
and defined($self->get_conf('PACKAGE_URL'))) {
return $self->convert_tree(
$self->gdt('This document was generated using @uref{{program_homepage}, @emph{{program}}}.',
{ 'program_homepage' => $self->get_conf('PACKAGE_URL'),
'program' => $self->get_conf('PROGRAM') }));
} else {
return $self->convert_tree(
$self->gdt('This document was generated automatically.'));
}
}
texinfo_register_formatting_function('program_string', \&ffmpeg_program_string);
# Customized file ending
sub ffmpeg_end_file($)
{
my $self = shift;
my $program_string = &{$self->{'format_program_string'}}($self);
my $program_text = <<EOT;
<p style="font-size: small;">
$program_string
</p>
EOT
my $footer = $ENV{FFMPEG_FOOTER} || <<EOT;
</div>
</body>
</html>
EOT
return $program_text . $footer;
}
texinfo_register_formatting_function('end_file', \&ffmpeg_end_file);
# Dummy title command
# Ignore title. Title is handled through ffmpeg_begin_file().
set_from_init_file('USE_TITLEPAGE_FOR_TITLE', 1);
sub ffmpeg_title($$$$)
{
return '';
}
texinfo_register_command_formatting('titlefont',
\&ffmpeg_title);
# Customized float command. Part of code borrowed from GNU Texinfo.
sub ffmpeg_float($$$$$)
{
my $self = shift;
my $cmdname = shift;
my $command = shift;
my $args = shift;
my $content = shift;
my ($caption, $prepended) = Texinfo::Common::float_name_caption($self,
$command);
my $caption_text = '';
my $prepended_text;
my $prepended_save = '';
if ($self->in_string()) {
if ($prepended) {
$prepended_text = $self->convert_tree_new_formatting_context(
$prepended, 'float prepended');
} else {
$prepended_text = '';
}
if ($caption) {
$caption_text = $self->convert_tree_new_formatting_context(
{'contents' => $caption->{'args'}->[0]->{'contents'}},
'float caption');
}
return $prepended.$content.$caption_text;
}
my $id = $self->command_id($command);
my $label;
if (defined($id) and $id ne '') {
$label = "<a name=\"$id\"></a>";
} else {
$label = '';
}
if ($prepended) {
if ($caption) {
# prepend the prepended tree to the first paragraph
my @caption_original_contents = @{$caption->{'args'}->[0]->{'contents'}};
my @caption_contents;
my $new_paragraph;
while (@caption_original_contents) {
my $content = shift @caption_original_contents;
if ($content->{'type'} and $content->{'type'} eq 'paragraph') {
%{$new_paragraph} = %{$content};
$new_paragraph->{'contents'} = [@{$content->{'contents'}}];
unshift (@{$new_paragraph->{'contents'}}, {'cmdname' => 'strong',
'args' => [{'type' => 'brace_command_arg',
'contents' => [$prepended]}]});
push @caption_contents, $new_paragraph;
last;
} else {
push @caption_contents, $content;
}
}
push @caption_contents, @caption_original_contents;
if ($new_paragraph) {
$caption_text = $self->convert_tree_new_formatting_context(
{'contents' => \@caption_contents}, 'float caption');
$prepended_text = '';
}
}
if ($caption_text eq '') {
$prepended_text = $self->convert_tree_new_formatting_context(
$prepended, 'float prepended');
if ($prepended_text ne '') {
$prepended_save = $prepended_text;
$prepended_text = '<p><strong>'.$prepended_text.'</strong></p>';
}
}
} else {
$prepended_text = '';
}
if ($caption and $caption_text eq '') {
$caption_text = $self->convert_tree_new_formatting_context(
$caption->{'args'}->[0], 'float caption');
}
if ($prepended_text.$caption_text ne '') {
$prepended_text = $self->_attribute_class('div','float-caption'). '>'
. $prepended_text;
$caption_text .= '</div>';
}
my $html_class = '';
if ($prepended_save =~ /NOTE/) {
$html_class = 'info';
$prepended_text = '';
$caption_text = '';
} elsif ($prepended_save =~ /IMPORTANT/) {
$html_class = 'warning';
$prepended_text = '';
$caption_text = '';
}
return $self->_attribute_class('div', $html_class). '>' . "\n" .
$prepended_text . $caption_text . $content . '</div>';
}
texinfo_register_command_formatting('float',
\&ffmpeg_float);
1;

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