avcodec/noise_bsf: Check for wrapped frames

Wrapped frames contain pointers so they need specific code to
noise them, the generic code would lead to segfaults

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0889ebc577)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

CREDITS

@@ -1,6 +1,6 @@
-See the Git history of the project (git://source.ffmpeg.org/ffmpeg) to
+See the Git history of the project (https://git.ffmpeg.org/ffmpeg) to
 get the names of people who have contributed to FFmpeg.
 
 To check the log, you can type the command "git log" in the FFmpeg
 source directory, or browse the online repository at
-http://source.ffmpeg.org.
+https://git.ffmpeg.org/ffmpeg

MAINTAINERS

@@ -121,7 +121,6 @@ Generic Parts:
     motion*                             Michael Niedermayer
   rate control:
     ratecontrol.c                       Michael Niedermayer
-    libxvid_rc.c                        Michael Niedermayer
   simple IDCT:
     simple_idct.c, simple_idct.h        Michael Niedermayer
   postprocessing:
@@ -220,7 +219,7 @@ Codecs:
   ptx.c                                 Ivo van Poorten
   qcelp*                                Reynaldo H. Verdejo Pinochet
   qdm2.c, qdm2data.h                    Roberto Togni
-  qsv*                                  Mark Thompson
+  qsv*                                  Mark Thompson, Zhong Li
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -333,6 +332,7 @@ Filters:
   vf_bwdif                              Thomas Mundt (CC <thomas.mundt@hr.de>)
   vf_chromakey.c                        Timo Rothenpieler
   vf_colorchannelmixer.c                Paul B Mahol
+  vf_colorconstancy.c                   Mina Sami    (CC <minas.gorgy@gmail.com>)
   vf_colorbalance.c                     Paul B Mahol
   vf_colorkey.c                         Timo Rothenpieler
   vf_colorlevels.c                      Paul B Mahol
@@ -413,7 +413,6 @@ Muxers/Demuxers:
   flvenc.c                              Michael Niedermayer, Steven Liu
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
-  hls.c                                 Anssi Hannula
   hlsenc.c                              Christian Suloway, Steven Liu
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
@@ -524,7 +523,7 @@ Operating systems / CPU architectures
 =====================================
 
 Alpha                                   Falk Hueffner
-MIPS                                    Manojkumar Bhosale
+MIPS                                    Manojkumar Bhosale, Shiyou Yin
 Mac OS X / PowerPC                      Romain Dolbeau, Guillaume Poirier
 Amiga / PowerPC                         Colin Ward
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
@@ -575,8 +574,11 @@ Releases
 If you want to maintain an older release, please contact us
 
 
-GnuPG Fingerprints of maintainers and contributors
-==================================================
+GnuPG Fingerprints and IRC nicknames of maintainers and contributors
+====================================================================
+
+IRC nicknames are in parentheses. These apply
+to the IRC channels listed on the website.
 
 Alexander Strasser            1C96 78B7 83CB 8AA7 9AF5 D1EB A7D8 A57B A876 E58F
 Anssi Hannula                 1A92 FF42 2DD9 8D2E 8AF7 65A9 4278 C520 513D F3CB
@@ -594,8 +596,9 @@ Jaikrishnan Menon             61A1 F09F 01C9 2D45 78E1 C862 25DC 8831 AF70 D368
 James Almer                   7751 2E8C FD94 A169 57E6 9A7A 1463 01AD 7376 59E0
 Jean Delvare                  7CA6 9F44 60F1 BDC4 1FD2 C858 A552 6B9B B3CD 4E6A
 Loren Merritt                 ABD9 08F4 C920 3F65 D8BE 35D7 1540 DAA7 060F 56DE
-Lou Logan                     7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
+Lou Logan (llogan)            7D68 DC73 CBEF EABB 671A B6CF 621C 2E28 82F8 DC3A
 Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
+                              DD1E C9E8 DE08 5C62 9B3E 1846 B18E 8928 B394 8D64
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Nikolay Aleksandrov           8978 1D8C FB71 588E 4B27 EAA8 C4F0 B5FC E011 13B1
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
@@ -611,5 +614,5 @@ Steinar H. Gunderson          C2E9 004F F028 C18E 4EAD DB83 7F61 7561 7797 8F76
 Stephan Hilb                  4F38 0B3A 5F39 B99B F505 E562 8D5C 5554 4E17 8863
 Tiancheng "Timothy" Gu        9456 AFC0 814A 8139 E994 8351 7FE6 B095 B582 B0D4
 Tim Nicholson                 38CF DB09 3ED0 F607 8B67 6CED 0C0B FC44 8B0B FC83
-Tomas Härdin                  A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
+Tomas Härdin (thardin)        A79D 4E3D F38F 763F 91F5 8B33 A01E 8AE0 41BB 2551
 Wei Gao                       4269 7741 857A 0E60 9EC5 08D2 4744 4EFA 62C1 87B9

Makefile

@@ -58,6 +58,7 @@ tools/target_dec_%_fuzzer$(EXESUF): $(FF_DEP_LIBS)
 CONFIGURABLE_COMPONENTS =                                           \
     $(wildcard $(FFLIBS:%=$(SRC_PATH)/lib%/all*.c))                 \
     $(SRC_PATH)/libavcodec/bitstream_filters.c                      \
+    $(SRC_PATH)/libavcodec/parsers.c                                \
     $(SRC_PATH)/libavformat/protocols.c                             \
 
 config.h: ffbuild/.config

RELEASE

@@ -1 +1 @@
-3.4.git
+4.1.11

RELEASE_NOTES

@@ -0,0 +1,15 @@
+
+              ┌─────────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 4.1 "al-Khwarizmi" │
+              └─────────────────────────────────────────────┘
+
+   The FFmpeg Project proudly presents FFmpeg 4.1 "al-Khwarizmi", about 6
+   months after the release of FFmpeg 4.0.
+
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on https://git.ffmpeg.org/gitweb/ffmpeg.git
+
+   We hope you will like this release as much as we enjoyed working on it, and
+   as usual, if you have any questions about it, or any FFmpeg related topic,
+   feel free to join us on the #ffmpeg IRC channel (on irc.libera.chat) or ask
+   on the mailing-lists.

doc/APIchanges

@@ -15,6 +15,41 @@ libavutil:     2017-10-21
 
 API changes, most recent first:
 
+-------- 8< --------- FFmpeg 4.1 was cut here -------- 8< ---------
+
+2018-10-27 - 718044dc19 - lavu 56.21.100 - pixdesc.h
+  Add av_read_image_line2(), av_write_image_line2()
+
+2018-10-24 - f9d4126f28 - lavu 56.20.100 - frame.h
+  Add AV_FRAME_DATA_S12M_TIMECODE
+
+2018-10-11 - f6d48b618a - lavc 58.33.100 - mediacodec.h
+  Add av_mediacodec_render_buffer_at_time().
+
+2018-09-09 - 35498c124a - lavc 58.29.100 - avcodec.h
+  Add AV_PKT_DATA_AFD
+
+2018-08-16 - b33f5299a5 - lavc 58.23.100 - avcodec.h
+  Add av_bsf_flush().
+
+2018-05-18 - 2b2f2f65f3 - lavf 58.15.100 - avformat.h
+  Add pmt_version field to AVProgram
+
+2018-05-17 - 5dfeb7f081 - lavf 58.14.100 - avformat.h
+  Add AV_DISPOSITION_STILL_IMAGE
+
+2018-05-10 - c855683427 - lavu 56.18.101 - hwcontext_cuda.h
+  Add AVCUDADeviceContext.stream.
+
+2018-04-30 - 56b081da57 - lavu 56.18.100 - pixdesc.h
+  Add AV_PIX_FMT_FLAG_ALPHA to AV_PIX_FMT_PAL8.
+
+2018-04-26 - 5be0410cb3 - lavu 56.17.100 - opt.h
+  Add AV_OPT_FLAG_DEPRECATED.
+
+2018-04-26 - 71fa82bed6 - lavu 56.16.100 - threadmessage.h
+  Add av_thread_message_queue_nb_elems().
+
 -------- 8< --------- FFmpeg 4.0 was cut here -------- 8< ---------
 
 2018-04-03 - d6fc031caf - lavu 56.13.100 - pixdesc.h
@@ -889,7 +924,7 @@ API changes, most recent first:
   Add av_opt_get_dict_val/set_dict_val with AV_OPT_TYPE_DICT to support
   dictionary types being set as options.
 
-2014-08-13 - afbd4b8 - lavf 56.01.0 - avformat.h
+2014-08-13 - afbd4b7e09 - lavf 56.01.0 - avformat.h
   Add AVFormatContext.event_flags and AVStream.event_flags for signaling to
   the user when events happen in the file/stream.
 
@@ -906,7 +941,7 @@ API changes, most recent first:
 2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
   Add avio_feof() and deprecate url_feof().
 
-2014-08-07 - bb78903 - lsws 2.1.3 - swscale.h
+2014-08-07 - bb789016d4 - lsws 2.1.3 - swscale.h
   sws_getContext is not going to be removed in the future.
 
 2014-08-07 - a561662 / ad1ee5f - lavc 55.73.101 / 55.57.3 - avcodec.h

doc/Doxyfile

@@ -38,7 +38,7 @@ PROJECT_NAME           = FFmpeg
 # could be handy for archiving the generated documentation or if some version
 # control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 4.1.11
 
 # Using the PROJECT_BRIEF tag one can provide an optional one line description
 # for a project that appears at the top of each page and should give viewer a

doc/authors.texi

@@ -3,9 +3,9 @@
 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
+(https://git.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}.
+online repository at @url{https://git.ffmpeg.org/ffmpeg}.
 
 Maintainers for the specific components are listed in the file
 @file{MAINTAINERS} in the source code tree.

doc/bitstream_filters.texi

@@ -37,6 +37,58 @@ 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.
+
+@end table
+
 @section chomp
 
 Remove zero padding at the end of a packet.
@@ -215,6 +267,15 @@ 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
@@ -505,7 +566,33 @@ 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 H.264, H.265 and MPEG-2.
+Supports H.264, H.265, MPEG-2 and VP9.
+
+@section vp9_metadata
+
+Modify metadata embedded in a VP9 stream.
+
+@table @option
+@item color_space
+Set the color space value in the frame header.
+@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 this cannot
+be set in RGB streams.
+@table @samp
+@item tv
+@item pc
+@end table
+@end table
 
 @section vp9_superframe
 

doc/codecs.texi

@@ -986,10 +986,6 @@ Set chroma qp offset from luma.
 @item trellis @var{integer} (@emph{encoding,audio,video})
 Set rate-distortion optimal quantization.
 
-@item sc_factor @var{integer} (@emph{encoding,video})
-Set value multiplied by qscale for each frame and added to
-scene_change_score.
-
 @item mv0_threshold @var{integer} (@emph{encoding,video})
 @item b_sensitivity @var{integer} (@emph{encoding,video})
 Adjust sensitivity of b_frame_strategy 1.

doc/decoders.texi

@@ -47,6 +47,12 @@ top-field-first is assumed
 
 @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
@@ -248,18 +254,25 @@ configuration. You need to explicitly configure the build with
 
 @table @option
 @item txt_page
-List of teletext page numbers to decode. You may use the special * string to
-match all pages. Pages that do not match the specified list are dropped.
+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_chop_top
 Discards the top teletext line. Default value is 1.
 @item txt_format
-Specifies the format of the decoded subtitles. The teletext decoder is capable
-of decoding the teletext pages to bitmaps or to simple text, you should use
-"bitmap" for teletext pages, because certain graphics and colors cannot be
-expressed in simple text. You might use "text" for teletext based subtitles if
-your application can handle simple text based subtitles. Default value is
-bitmap.
+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
@@ -272,7 +285,8 @@ 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 30000 which is 30 seconds.
+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.

doc/demuxers.texi

@@ -269,6 +269,12 @@ ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
 @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
@@ -538,6 +544,9 @@ This demuxer accepts the following options:
 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
@@ -552,6 +561,10 @@ Show the detected raw packet size, cannot be set by the user.
 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

doc/developer.texi

@@ -128,6 +128,9 @@ 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
 Implementation defined behavior for signed integers is assumed to match the
 expected behavior for two's complement. Non representable values in integer

doc/encoders.texi

@@ -2565,6 +2565,9 @@ The following standard libavcodec options are used:
 @option{bf} / @option{max_b_frames}
 @item
 @option{profile}
+
+If not set, this will be determined automatically from the format of the input
+frames and the profiles supported by the driver.
 @item
 @option{level}
 @item
@@ -2585,7 +2588,8 @@ Speed / quality tradeoff: higher values are faster / worse quality.
 Size / quality tradeoff: higher values are smaller / worse quality.
 @item
 @option{qmin}
-(only: @option{qmax} is not supported)
+@item
+@option{qmax}
 @item
 @option{i_qfactor} / @option{i_quant_factor}
 @item
@@ -2594,8 +2598,22 @@ Size / quality tradeoff: higher values are smaller / worse quality.
 @option{b_qfactor} / @option{b_quant_factor}
 @item
 @option{b_qoffset} / @option{b_quant_offset}
+@item
+@option{slices}
 @end itemize
 
+All encoders support the following options:
+@itemize
+@item
+@option{low_power}
+
+Some drivers/platforms offer a second encoder for some codecs intended to use
+less power than the default encoder; setting this option will attempt to use
+that encoder.  Note that it may support a reduced feature set, so some other
+options may not be available in this mode.
+@end itemize
+
+Each encoder also has its own specific options:
 @table @option
 
 @item h264_vaapi
@@ -2603,8 +2621,6 @@ Size / quality tradeoff: higher values are smaller / worse quality.
 @option{level} sets the value of @emph{level_idc}.
 
 @table @option
-@item low_power
-Use low-power encoding mode.
 @item coder
 Set entropy encoder (default is @emph{cabac}).  Possible values:
 
@@ -2617,21 +2633,70 @@ Use CABAC.
 @item cavlc
 Use CAVLC.
 @end table
+
+@item aud
+Include access unit delimiters in the stream (not included by default).
+
+@item sei
+Set SEI message types to include.
+Some combination of the following values:
+@table @samp
+@item identifier
+Include a @emph{user_data_unregistered} message containing information about
+the encoder.
+@item timing
+Include picture timing parameters (@emph{buffering_period} and
+@emph{pic_timing} messages).
+@item recovery_point
+Include recovery points where appropriate (@emph{recovery_point} messages).
+@end table
+
 @end table
 
 @item hevc_vaapi
 @option{profile} and @option{level} set the values of
 @emph{general_profile_idc} and @emph{general_level_idc} respectively.
 
+@table @option
+@item aud
+Include access unit delimiters in the stream (not included by default).
+
+@item tier
+Set @emph{general_tier_flag}.  This may affect the level chosen for the stream
+if it is not explicitly specified.
+
+@item sei
+Set SEI message types to include.
+Some combination of the following values:
+@table @samp
+@item hdr
+Include HDR metadata if the input frames have it
+(@emph{mastering_display_colour_volume} and @emph{content_light_level}
+messages).
+@end table
+
+@end table
+
 @item mjpeg_vaapi
-Always encodes using the standard quantisation and huffman tables -
-@option{global_quality} scales the standard quantisation table (range 1-100).
+Only baseline DCT encoding is supported.  The encoder always uses the standard
+quantisation and huffman tables - @option{global_quality} scales the standard
+quantisation table (range 1-100).
+
+For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported.  RGB is also
+supported, and will create an RGB JPEG.
+
+@table @option
+@item jfif
+Include JFIF header in each frame (not included by default).
+@item huffman
+Include standard huffman tables (on by default).  Turning this off will save
+a few hundred bytes in each output frame, but may lose compatibility with some
+JPEG decoders which don't fully handle MJPEG.
+@end table
 
 @item mpeg2_vaapi
 @option{profile} and @option{level} set the value of @emph{profile_and_level_indication}.
 
-No rate control is supported.
-
 @item vp8_vaapi
 B-frames are not supported.
 
@@ -2726,6 +2791,52 @@ Reduces detail but attempts to preserve color at extremely low bitrates.
 
 @end table
 
+@section libxavs2
+
+xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.
+
+This encoder requires the presence of the libxavs2 headers and library
+during configuration. You need to explicitly configure the build with
+@option{--enable-libxavs2}.
+
+@subsection Options
+
+@table @option
+@item lcu_row_threads
+Set the number of parallel threads for rows from 1 to 8 (default 5).
+
+@item initial_qp
+Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
+used to set the initial qp for the first frame.
+
+@item qp
+Set the xavs2 quantization parameter from 1 to 63 (default 34). This is
+used to set the qp value under constant-QP mode.
+
+@item max_qp
+Set the max qp for rate control from 1 to 63 (default 55).
+
+@item min_qp
+Set the min qp for rate control from 1 to 63 (default 20).
+
+@item speed_level
+Set the Speed level from 0 to 9 (default 0). Higher is better but slower.
+
+@item log_level
+Set the log level from -1 to 3 (default 0). -1: none, 0: error,
+1: warning, 2: info, 3: debug.
+
+@item xavs2-params
+Set xavs2 options using a list of @var{key}=@var{value} couples separated
+by ":".
+
+For example to specify libxavs2 encoding options with @option{-xavs2-params}:
+
+@example
+ffmpeg -i input -c:v libxavs2 -xavs2-params preset_level=5 output.avs2
+@end example
+@end table
+
 @c man end VIDEO ENCODERS
 
 @chapter Subtitles Encoders

doc/examples/.gitignore

@@ -20,3 +20,5 @@
 /scaling_video
 /transcode_aac
 /transcoding
+/vaapi_encode
+/vaapi_transcode

doc/examples/filtering_audio.c

@@ -74,7 +74,6 @@ static int open_input_file(const char *filename)
     if (!dec_ctx)
         return AVERROR(ENOMEM);
     avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[audio_stream_index]->codecpar);
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the audio decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {

doc/examples/filtering_video.c

@@ -29,6 +29,8 @@
 
 #define _XOPEN_SOURCE 600 /* for usleep */
 #include <unistd.h>
+#include <stdio.h>
+#include <stdlib.h>
 
 #include <libavcodec/avcodec.h>
 #include <libavformat/avformat.h>
@@ -77,7 +79,6 @@ static int open_input_file(const char *filename)
     if (!dec_ctx)
         return AVERROR(ENOMEM);
     avcodec_parameters_to_context(dec_ctx, fmt_ctx->streams[video_stream_index]->codecpar);
-    av_opt_set_int(dec_ctx, "refcounted_frames", 1, 0);
 
     /* init the video decoder */
     if ((ret = avcodec_open2(dec_ctx, dec, NULL)) < 0) {
@@ -210,18 +211,21 @@ int main(int argc, char **argv)
 {
     int ret;
     AVPacket packet;
-    AVFrame *frame = av_frame_alloc();
-    AVFrame *filt_frame = av_frame_alloc();
+    AVFrame *frame;
+    AVFrame *filt_frame;
 
-    if (!frame || !filt_frame) {
-        perror("Could not allocate frame");
-        exit(1);
-    }
     if (argc != 2) {
         fprintf(stderr, "Usage: %s file\n", argv[0]);
         exit(1);
     }
 
+    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)
@@ -248,27 +252,25 @@ int main(int argc, char **argv)
                     goto end;
                 }
 
-                if (ret >= 0) {
-                    frame->pts = frame->best_effort_timestamp;
+                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);
+                /* 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);

doc/examples/hw_decode.c

@@ -4,21 +4,23 @@
  *
  * HW Acceleration API (video decoding) decode sample
  *
- * This file is part of FFmpeg.
+ * 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:
  *
- * 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.
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
  *
- * 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
+ * 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.
  */
 
 /**
@@ -211,7 +213,6 @@ int main(int argc, char *argv[])
         return -1;
 
     decoder_ctx->get_format  = get_hw_format;
-    av_opt_set_int(decoder_ctx, "refcounted_frames", 1, 0);
 
     if (hw_decoder_init(decoder_ctx, type) < 0)
         return -1;

doc/examples/transcoding.c

@@ -172,6 +172,9 @@ static int open_output_file(const char *filename)
                 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) {
@@ -183,8 +186,6 @@ static int open_output_file(const char *filename)
                 av_log(NULL, AV_LOG_ERROR, "Failed to copy encoder parameters to output stream #%u\n", i);
                 return ret;
             }
-            if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-                enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
             out_stream->time_base = enc_ctx->time_base;
             stream_ctx[i].enc_ctx = enc_ctx;

doc/examples/vaapi_encode.c

@@ -1,21 +1,23 @@
 /*
  * Video Acceleration API (video encoding) encode sample
  *
- * This file is part of FFmpeg.
+ * 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:
  *
- * 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.
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
  *
- * 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
+ * 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.
  */
 
 /**

doc/examples/vaapi_transcode.c

@@ -1,21 +1,23 @@
 /*
  * Video Acceleration API (video transcoding) transcode sample
  *
- * This file is part of FFmpeg.
+ * 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:
  *
- * 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.
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
  *
- * 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
+ * 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.
  */
 
 /**
@@ -177,7 +179,7 @@ static int dec_enc(AVPacket *pkt, AVCodec *enc_codec)
             }
             /* set AVCodecContext Parameters for encoder, here we keep them stay
              * the same as decoder.
-             * xxx: now the the sample can't handle resolution change case.
+             * 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;

doc/fate.texi

@@ -155,6 +155,8 @@ space on each client, network bandwidth and so on benefit from smaller test case
 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.
 
 @example
 #First update your local samples copy:
@@ -222,6 +224,11 @@ Set to @samp{1} to generate the missing or mismatched references.
 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

doc/ffmpeg.texi

@@ -216,16 +216,208 @@ filters is obviously also impossible, since filters work on uncompressed data.
 @chapter Stream selection
 @c man begin STREAM SELECTION
 
-By default, @command{ffmpeg} includes only one stream of each type (video, audio, subtitle)
-present in the input files and adds them to each output file.  It picks the
-"best" of each based upon the following criteria: for video, it is the stream
-with the highest resolution, for audio, it is the stream with the most channels, for
-subtitles, it is the first subtitle stream. In the case where several streams of
-the same type rate equally, the stream with the lowest index is chosen.
+@command{ffmpeg} provides the @code{-map} option for manual control of stream selection in each
+output file. Users can skip @code{-map} and let ffmpeg perform automatic stream selection as
+described below. The @code{-vn / -an / -sn / -dn} options can be used to skip inclusion of
+video, audio, subtitle and data streams respectively, whether manually mapped or automatically
+selected, except for those streams which are outputs of complex filtergraphs.
 
-You can disable some of those defaults by using the @code{-vn/-an/-sn/-dn} options. For
-full manual control, use the @code{-map} option, which disables the defaults just
-described.
+@section Description
+The sub-sections that follow describe the various rules that are involved in stream selection.
+The examples that follow next show how these rules are applied in practice.
+
+While every effort is made to accurately reflect the behavior of the program, FFmpeg is under
+continuous development and the code may have changed since the time of this writing.
+
+@subsection Automatic stream selection
+
+In the absence of any map options for a particular output file, ffmpeg inspects the output
+format to check which type of streams can be included in it, viz. video, audio and/or
+subtitles. For each acceptable stream type, ffmpeg will pick one stream, when available,
+from among all the inputs.
+
+It will select that stream based upon the following criteria:
+@itemize
+@item
+for video, it is the stream with the highest resolution,
+@item
+for audio, it is the stream with the most channels,
+@item
+for subtitles, it is the first subtitle stream found but there's a caveat.
+The output format's default subtitle encoder can be either text-based or image-based,
+and only a subtitle stream of the same type will be chosen.
+@end itemize
+
+In the case where several streams of the same type rate equally, the stream with the lowest
+index is chosen.
+
+Data or attachment streams are not automatically selected and can only be included
+using @code{-map}.
+@subsection Manual stream selection
+
+When @code{-map} is used, only user-mapped streams are included in that output file,
+with one possible exception for filtergraph outputs described below.
+
+@subsection Complex filtergraphs
+
+If there are any complex filtergraph output streams with unlabeled pads, they will be added
+to the first output file. This will lead to a fatal error if the stream type is not supported
+by the output format. In the absence of the map option, the inclusion of these streams leads
+to the automatic stream selection of their types being skipped. If map options are present,
+these filtergraph streams are included in addition to the mapped streams.
+
+Complex filtergraph output streams with labeled pads must be mapped once and exactly once.
+
+@subsection Stream handling
+
+Stream handling is independent of stream selection, with an exception for subtitles described
+below. Stream handling is set via the @code{-codec} option addressed to streams within a
+specific @emph{output} file. In particular, codec options are applied by ffmpeg after the
+stream selection process and thus do not influence the latter. If no @code{-codec} option is
+specified for a stream type, ffmpeg will select the default encoder registered by the output
+file muxer.
+
+An exception exists for subtitles. If a subtitle encoder is specified for an output file, the
+first subtitle stream found of any type, text or image, will be included. ffmpeg does not validate
+if the specified encoder can convert the selected stream or if the converted stream is acceptable
+within the output format. This applies generally as well: when the user sets an encoder manually,
+the stream selection process cannot check if the encoded stream can be muxed into the output file.
+If it cannot, ffmpeg will abort and @emph{all} output files will fail to be processed.
+
+@section Examples
+
+The following examples illustrate the behavior, quirks and limitations of ffmpeg's stream
+selection methods.
+
+They assume the following three input files.
+
+@verbatim
+
+input file 'A.avi'
+      stream 0: video 640x360
+      stream 1: audio 2 channels
+
+input file 'B.mp4'
+      stream 0: video 1920x1080
+      stream 1: audio 2 channels
+      stream 2: subtitles (text)
+      stream 3: audio 5.1 channels
+      stream 4: subtitles (text)
+
+input file 'C.mkv'
+      stream 0: video 1280x720
+      stream 1: audio 2 channels
+      stream 2: subtitles (image)
+@end verbatim
+
+@subsubheading Example: automatic stream selection
+@example
+ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov
+@end example
+There are three output files specified, and for the first two, no @code{-map} options
+are set, so ffmpeg will select streams for these two files automatically.
+
+@file{out1.mkv} is a Matroska container file and accepts video, audio and subtitle streams,
+so ffmpeg will try to select one of each type.@*
+For video, it will select @code{stream 0} from @file{B.mp4}, which has the highest
+resolution among all the input video streams.@*
+For audio, it will select @code{stream 3} from @file{B.mp4}, since it has the greatest
+number of channels.@*
+For subtitles, it will select @code{stream 2} from @file{B.mp4}, which is the first subtitle
+stream from among @file{A.avi} and @file{B.mp4}.
+
+@file{out2.wav} accepts only audio streams, so only @code{stream 3} from @file{B.mp4} is
+selected.
+
+For @file{out3.mov}, since a @code{-map} option is set, no automatic stream selection will
+occur. The @code{-map 1:a} option will select all audio streams from the second input
+@file{B.mp4}. No other streams will be included in this output file.
+
+For the first two outputs, all included streams will be transcoded. The encoders chosen will
+be the default ones registered by each output format, which may not match the codec of the
+selected input streams.
+
+For the third output, codec option for audio streams has been set
+to @code{copy}, so no decoding-filtering-encoding operations will occur, or @emph{can} occur.
+Packets of selected streams shall be conveyed from the input file and muxed within the output
+file.
+
+@subsubheading Example: automatic subtitles selection
+@example
+ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv
+@end example
+Although @file{out1.mkv} is a Matroska container file which accepts subtitle streams, only a
+video and audio stream shall be selected. The subtitle stream of @file{C.mkv} is image-based
+and the default subtitle encoder of the Matroska muxer is text-based, so a transcode operation
+for the subtitles is expected to fail and hence the stream isn't selected. However, in
+@file{out2.mkv}, a subtitle encoder is specified in the command and so, the subtitle stream is
+selected, in addition to the video stream. The presence of @code{-an} disables audio stream
+selection for @file{out2.mkv}.
+
+@subsubheading Example: unlabeled filtergraph outputs
+@example
+ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt
+@end example
+A filtergraph is setup here using the @code{-filter_complex} option and consists of a single
+video filter. The @code{overlay} filter requires exactly two video inputs, but none are
+specified, so the first two available video streams are used, those of @file{A.avi} and
+@file{C.mkv}. The output pad of the filter has no label and so is sent to the first output file
+@file{out1.mp4}. Due to this, automatic selection of the video stream is skipped, which would
+have selected the stream in @file{B.mp4}. The audio stream with most channels viz. @code{stream 3}
+in @file{B.mp4}, is chosen automatically. No subtitle stream is chosen however, since the MP4
+format has no default subtitle encoder registered, and the user hasn't specified a subtitle encoder.
+
+The 2nd output file, @file{out2.srt}, only accepts text-based subtitle streams. So, even though
+the first subtitle stream available belongs to @file{C.mkv}, it is image-based and hence skipped.
+The selected stream, @code{stream 2} in @file{B.mp4}, is the first text-based subtitle stream.
+
+@subsubheading Example: labeled filtergraph outputs
+@example
+ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
+       -map '[outv]' -an        out1.mp4 \
+                                out2.mkv \
+       -map '[outv]' -map 1:a:0 out3.mkv
+@end example
+
+The above command will fail, as the output pad labelled @code{[outv]} has been mapped twice.
+None of the output files shall be processed.
+
+@example
+ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
+       -an        out1.mp4 \
+                  out2.mkv \
+       -map 1:a:0 out3.mkv
+@end example
+
+This command above will also fail as the hue filter output has a label, @code{[outv]},
+and hasn't been mapped anywhere.
+
+The command should be modified as follows,
+@example
+ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
+        -map '[outv1]' -an        out1.mp4 \
+                                  out2.mkv \
+        -map '[outv2]' -map 1:a:0 out3.mkv
+@end example
+The video stream from @file{B.mp4} is sent to the hue filter, whose output is cloned once using
+the split filter, and both outputs labelled. Then a copy each is mapped to the first and third
+output files.
+
+The overlay filter, requiring two video inputs, uses the first two unused video streams. Those
+are the streams from @file{A.avi} and @file{C.mkv}. The overlay output isn't labelled, so it is
+sent to the first output file @file{out1.mp4}, regardless of the presence of the @code{-map} option.
+
+The aresample filter is sent the first unused audio stream, that of @file{A.avi}. Since this filter
+output is also unlabelled, it too is mapped to the first output file. The presence of @code{-an}
+only suppresses automatic or manual stream selection of audio streams, not outputs sent from
+filtergraphs. Both these mapped streams shall be ordered before the mapped stream in @file{out1.mp4}.
+
+The video, audio and subtitle streams mapped to @code{out2.mkv} are entirely determined by
+automatic stream selection.
+
+@file{out3.mkv} consists of the cloned video output from the hue filter and the first audio
+stream from @file{B.mp4}.
+@*
 
 @c man end STREAM SELECTION
 
@@ -316,7 +508,7 @@ input until the timestamps reach @var{position}.
 @var{position} must be a time duration specification,
 see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
-@item -sseof @var{position} (@emph{input/output})
+@item -sseof @var{position} (@emph{input})
 
 Like the @code{-ss} option but relative to the "end of file". That is negative
 values are earlier in the file, 0 is at EOF.
@@ -375,22 +567,31 @@ The following dispositions are recognized:
 @item hearing_impaired
 @item visual_impaired
 @item clean_effects
+@item attached_pic
 @item captions
 @item descriptions
+@item dependent
 @item metadata
 @end table
 
 For example, to make the second audio stream the default stream:
 @example
-ffmpeg -i in.mkv -disposition:a:1 default out.mkv
+ffmpeg -i in.mkv -c copy -disposition:a:1 default out.mkv
 @end example
 
 To make the second subtitle stream the default stream and remove the default
 disposition from the first subtitle stream:
 @example
-ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT
+ffmpeg -i in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv
 @end example
 
+To add an embedded cover/thumbnail:
+@example
+ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4
+@end example
+
+Not all muxers support embedded thumbnails, and those who do, only support a few formats, like JPEG or PNG.
+
 @item -program [title=@var{title}:][program_num=@var{program_num}:]st=@var{stream}[:st=@var{stream}...] (@emph{output})
 
 Creates a program with the specified @var{title}, @var{program_num} and adds the specified
@@ -623,8 +824,6 @@ as the input (or graph output) and automatic conversions are disabled.
 
 @item -sws_flags @var{flags} (@emph{input/output})
 Set SwScaler flags.
-@item -vdt @var{n}
-Discard threshold.
 
 @item -rc_override[:@var{stream_specifier}] @var{override} (@emph{output,per-stream})
 Rate control override for specific intervals, formatted as "int,int,int"
@@ -1154,12 +1353,12 @@ disable any chapter copying.
 
 @item -benchmark (@emph{global})
 Show benchmarking information at the end of an encode.
-Shows CPU time used and maximum memory consumption.
+Shows real, system and user time used and maximum memory consumption.
 Maximum memory consumption is not supported on all systems,
 it will usually display as 0 if not supported.
 @item -benchmark_all (@emph{global})
 Show benchmarking information during the encode.
-Shows CPU time used in various steps (audio/video encode/decode).
+Shows real, system and user time used in various steps (audio/video encode/decode).
 @item -timelimit @var{duration} (@emph{global})
 Exit after ffmpeg has been running for @var{duration} seconds.
 @item -dump (@emph{global})

doc/ffplay.texi

@@ -60,6 +60,8 @@ Play @var{duration} seconds of audio/video.
 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
@@ -72,6 +74,10 @@ as 100.
 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}

doc/ffprobe.texi

@@ -584,7 +584,7 @@ value is 0.
 This is required for generating an XML file which can be validated
 through an XSD file.
 
-@item xsd_compliant, x
+@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.

doc/formats.texi

@@ -30,37 +30,43 @@ latency. Must be an integer not lesser than 32. It is 5000000 by default.
 @item packetsize @var{integer} (@emph{output})
 Set packet size.
 
-@item fflags @var{flags} (@emph{input/output})
-Set format flags.
+@item fflags @var{flags}
+Set format flags. Some are implemented for a limited number of formats.
 
-Possible values:
+Possible values for input files:
 @table @samp
-@item ignidx
-Ignore index.
+@item discardcorrupt
+Discard corrupted packets.
 @item fastseek
 Enable fast, but inaccurate seeks for some formats.
 @item genpts
-Generate PTS.
+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 that can be exactly calculated.
+Do not fill in missing values in packet fields that can be exactly calculated.
 @item noparse
 Disable AVParsers, this needs @code{+nofillin} too.
-@item igndts
-Ignore DTS.
-@item discardcorrupt
-Discard corrupted frames.
 @item sortdts
-Try to interleave output packets by DTS.
-@item keepside
-Do not merge side data.
-@item latm
-Enable RTP MP4A-LATM payload.
-@item nobuffer
-Reduce the latency introduced by optional buffering
+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
@@ -214,6 +220,10 @@ ffprobe -dump_separator "
 @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.
 @end table
 
 @c man end FORMAT OPTIONS

doc/general.texi

@@ -17,6 +17,34 @@ for more formats. None of them are used by default, their use has to be
 explicitly requested by passing the appropriate flags to
 @command{./configure}.
 
+@section libxavs2
+
+FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video encoding.
+
+Go to @url{https://github.com/pkuvcl/xavs2} and follow the instructions for
+installing the library. Then pass @code{--enable-libxavs2} to configure to
+enable it.
+
+@float NOTE
+libxavs2 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
+@section libdavs2
+
+FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video decoding.
+
+Go to @url{https://github.com/pkuvcl/davs2} and follow the instructions for
+installing the library. Then pass @code{--enable-libdavs2} to configure to
+enable it.
+
+@float NOTE
+libdavs2 is under the GNU Public License Version 2 or later
+(see @url{http://www.gnu.org/licenses/old-licenses/gpl-2.0.html} for
+details), you must upgrade FFmpeg's license to GPL in order to use it.
+@end float
+
 @section Alliance for Open Media libaom
 
 FFmpeg can make use of the libaom library for AV1 decoding.
@@ -46,9 +74,10 @@ upgrade FFmpeg's license to LGPL version 3 (or if you have enabled
 GPL components, GPL version 3) by passing @code{--enable-version3} to configure in
 order to use it.
 
-The Fraunhofer AAC library is licensed under a license incompatible to the GPL
-and is not known to be compatible to the LGPL. Therefore, you have to pass
-@code{--enable-nonfree} to configure to use it.
+The license of the Fraunhofer AAC library is incompatible with the GPL.
+Therefore, for GPL builds, you have to pass @code{--enable-nonfree} to
+configure in order to use it. To the best of our knowledge, it is
+compatible with the LGPL.
 @end float
 
 @subsection OpenCORE AMR
@@ -71,7 +100,7 @@ Then pass @code{--enable-libvo-amrwbenc} to configure to enable it.
 
 @subsection Fraunhofer AAC library
 
-FFmpeg can make use of the Fraunhofer AAC library for AAC encoding.
+FFmpeg can make use of the Fraunhofer AAC library for AAC decoding & encoding.
 
 Go to @url{http://sourceforge.net/projects/opencore-amr/} and follow the
 instructions for installing the library.
@@ -516,6 +545,7 @@ library:
 @item raw VC-1                  @tab X @tab X
 @item raw PCM A-law             @tab X @tab X
 @item raw PCM mu-law            @tab X @tab X
+@item raw PCM Archimedes VIDC   @tab X @tab X
 @item raw PCM signed 8 bit      @tab X @tab X
 @item raw PCM signed 16 bit big-endian  @tab X @tab X
 @item raw PCM signed 16 bit little-endian  @tab X @tab X
@@ -559,6 +589,7 @@ library:
 @item SAP                       @tab X @tab X
 @item SBG                       @tab   @tab X
 @item SDP                       @tab   @tab X
+@item SER                       @tab   @tab X
 @item Sega FILM/CPK             @tab X @tab X
     @tab Used in many Sega Saturn console games.
 @item Silicon Graphics Movie    @tab   @tab X
@@ -802,6 +833,7 @@ following image formats are supported:
     @tab fourcc: G2M2, G2M3
 @item Go2Webinar             @tab     @tab  X
     @tab fourcc: G2M4
+@item Gremlin Digital Video  @tab     @tab  X
 @item H.261                  @tab  X  @tab  X
 @item H.263 / H.263-1996     @tab  X  @tab  X
 @item H.263+ / H.263-1998 / H.263 version 2  @tab  X  @tab  X
@@ -822,6 +854,7 @@ following image formats are supported:
     @tab IFF interleaved bitmap
 @item IFF ByteRun1           @tab     @tab  X
     @tab IFF run length encoded bitmap
+@item Infinity IMM4          @tab     @tab  X
 @item Intel H.263            @tab     @tab  X
 @item Intel Indeo 2          @tab     @tab  X
 @item Intel Indeo 3          @tab     @tab  X
@@ -1048,6 +1081,7 @@ following image formats are supported:
 @item ATRAC1                 @tab     @tab  X
 @item ATRAC3                 @tab     @tab  X
 @item ATRAC3+                @tab     @tab  X
+@item ATRAC9                 @tab     @tab  X
 @item Bink Audio             @tab     @tab  X
     @tab Used in Bink and Smacker files in many games.
 @item CELT                   @tab     @tab  E
@@ -1114,6 +1148,7 @@ following image formats are supported:
     @tab encoding supported through external library libopus
 @item PCM A-law              @tab  X  @tab  X
 @item PCM mu-law             @tab  X  @tab  X
+@item PCM Archimedes VIDC    @tab  X  @tab  X
 @item PCM signed 8-bit planar  @tab  X  @tab  X
 @item PCM signed 16-bit big-endian planar  @tab  X  @tab  X
 @item PCM signed 16-bit little-endian planar  @tab  X  @tab  X

doc/git-howto.texi

@@ -53,7 +53,7 @@ Most distribution and operating system provide a package for it.
 @section Cloning the source tree
 
 @example
-git clone git://source.ffmpeg.org/ffmpeg <target>
+git clone https://git.ffmpeg.org/ffmpeg.git <target>
 @end example
 
 This will put the FFmpeg sources into the directory @var{<target>}.
@@ -187,11 +187,18 @@ to make sure you don't have untracked files or deletions.
 git add [-i|-p|-A] <filenames/dirnames>
 @end example
 
-Make sure you have told Git your name and email address
+Make sure you have told Git your name, email address and GPG key
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
+git config --global user.signingkey ABCDEF0123245
+@end example
+
+Enable signing all commits or use -S
+
+@example
+git config --global commit.gpgsign true
 @end example
 
 Use @option{--global} to set the global configuration for all your Git checkouts.
@@ -393,6 +400,19 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
+@chapter gpg key generation
+
+If you have no gpg key yet, we recommend that you create a ed25519 based key as it
+is small, fast and secure. Especially it results in small signatures in git.
+
+@example
+gpg --default-new-key-algo "ed25519/cert,sign+cv25519/encr" --quick-generate-key "human@@server.com"
+@end example
+
+When generating a key, make sure the email specified matches the email used in git as some sites like
+github consider mismatches a reason to declare such commits unverified. After generating a key you
+can add it to the MAINTAINER file and upload it to a keyserver.
+
 @chapter Pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,

doc/indevs.texi

@@ -267,7 +267,8 @@ audio track.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
+Defaults to @option{false}. Alternatively you can use the @code{-sources}
+option of ffmpeg to list the available input devices.
 
 @item list_formats
 If set to @option{true}, print a list of supported formats and exit.
@@ -326,6 +327,12 @@ Defaults to @samp{2}.
 Sets the decklink device duplex mode. Must be @samp{unset}, @samp{half} or @samp{full}.
 Defaults to @samp{unset}.
 
+@item timecode_format
+Timecode type to include in the frame and video stream metadata. Must be
+@samp{none}, @samp{rp188vitc}, @samp{rp188vitc2}, @samp{rp188ltc},
+@samp{rp188any}, @samp{vitc}, @samp{vitc2}, or @samp{serial}. Defaults to
+@samp{none} (not included).
+
 @item video_input
 Sets the video input source. Must be @samp{unset}, @samp{sdi}, @samp{hdmi},
 @samp{optical_sdi}, @samp{component}, @samp{composite} or @samp{s_video}.
@@ -364,6 +371,20 @@ If set to @option{true}, timestamps are forwarded as they are without removing
 the initial offset.
 Defaults to @option{false}.
 
+@item timestamp_align
+Capture start time alignment in seconds. If set to nonzero, input frames are
+dropped till the system timestamp aligns with configured value.
+Alignment difference of up to one frame duration is tolerated.
+This is useful for maintaining input synchronization across N different
+hardware devices deployed for 'N-way' redundancy. The system time of different
+hardware devices should be synchronized with protocols such as NTP or PTP,
+before using this option.
+Note that this method is not foolproof. In some border cases input
+synchronization may not happen due to thread scheduling jitters in the OS.
+Either sync could go wrong by 1 frame or in a rarer case
+@option{timestamp_align} seconds.
+Defaults to @samp{0}.
+
 @end table
 
 @subsection Examples
@@ -402,116 +423,6 @@ ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder'
 
 @end itemize
 
-@section kmsgrab
-
-KMS video input device.
-
-Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
-DRM object that can be passed to other hardware functions.
-
-Requires either DRM master or CAP_SYS_ADMIN to run.
-
-If you don't understand what all of that means, you probably don't want this.  Look at
-@option{x11grab} instead.
-
-@subsection Options
-
-@table @option
-
-@item device
-DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
-
-@item format
-Pixel format of the framebuffer.  Defaults to @option{bgr0}.
-
-@item format_modifier
-Format modifier to signal on output frames.  This is necessary to import correctly into
-some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
-
-@item crtc_id
-KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
-will be used.
-
-@item plane_id
-KMS plane ID to define the capture source.  Defaults to the first active plane found if
-neither @option{crtc_id} nor @option{plane_id} are specified.
-
-@item framerate
-Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
-changes - it just defines the interval at which the framebuffer is sampled.  Sampling
-faster than the framebuffer update rate will generate independent frames with the same
-content.  Defaults to @code{30}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-Capture from the first active plane, download the result to normal frames and encode.
-This will only work if the framebuffer is both linear and mappable - if not, the result
-may be scrambled or fail to download.
-@example
-ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
-@end example
-
-@item
-Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
-@example
-ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
-@end example
-
-@end itemize
-
-@section libndi_newtek
-
-The libndi_newtek input device provides capture capabilities for using NDI (Network
-Device Interface, standard created by NewTek).
-
-Input filename is a NDI source name that could be found by sending -find_sources 1
-to command line - it has no specific syntax but human-readable formatted.
-
-To enable this input device, you need the NDI SDK and you
-need to configure with the appropriate @code{--extra-cflags}
-and @code{--extra-ldflags}.
-
-@subsection Options
-
-@table @option
-
-@item find_sources
-If set to @option{true}, print a list of found/available NDI sources and exit.
-Defaults to @option{false}.
-
-@item wait_sources
-Override time to wait until the number of online sources have changed.
-Defaults to @option{0.5}.
-
-@item allow_video_fields
-When this flag is @option{false}, all video that you receive will be progressive.
-Defaults to @option{true}.
-
-@end table
-
-@subsection Examples
-
-@itemize
-
-@item
-List input devices:
-@example
-ffmpeg -f libndi_newtek -find_sources 1 -i dummy
-@end example
-
-@item
-Restream to NDI:
-@example
-ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
-@end example
-
-@end itemize
-
 @section dshow
 
 Windows DirectShow input device.
@@ -939,6 +850,68 @@ Set the number of channels. Default is 2.
 
 @end table
 
+@section kmsgrab
+
+KMS video input device.
+
+Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a
+DRM object that can be passed to other hardware functions.
+
+Requires either DRM master or CAP_SYS_ADMIN to run.
+
+If you don't understand what all of that means, you probably don't want this.  Look at
+@option{x11grab} instead.
+
+@subsection Options
+
+@table @option
+
+@item device
+DRM device to capture on.  Defaults to @option{/dev/dri/card0}.
+
+@item format
+Pixel format of the framebuffer.  Defaults to @option{bgr0}.
+
+@item format_modifier
+Format modifier to signal on output frames.  This is necessary to import correctly into
+some APIs, but can't be autodetected.  See the libdrm documentation for possible values.
+
+@item crtc_id
+KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
+will be used.
+
+@item plane_id
+KMS plane ID to define the capture source.  Defaults to the first active plane found if
+neither @option{crtc_id} nor @option{plane_id} are specified.
+
+@item framerate
+Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
+changes - it just defines the interval at which the framebuffer is sampled.  Sampling
+faster than the framebuffer update rate will generate independent frames with the same
+content.  Defaults to @code{30}.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Capture from the first active plane, download the result to normal frames and encode.
+This will only work if the framebuffer is both linear and mappable - if not, the result
+may be scrambled or fail to download.
+@example
+ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
+@end example
+
+@item
+Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
+@example
+ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
+@end example
+
+@end itemize
+
 @section lavfi
 
 Libavfilter input virtual device.
@@ -1077,6 +1050,71 @@ IIDC1394 input device, based on libdc1394 and libraw1394.
 
 Requires the configure option @code{--enable-libdc1394}.
 
+@section libndi_newtek
+
+The libndi_newtek input device provides capture capabilities for using NDI (Network
+Device Interface, standard created by NewTek).
+
+Input filename is a NDI source name that could be found by sending -find_sources 1
+to command line - it has no specific syntax but human-readable formatted.
+
+To enable this input device, you need the NDI SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+
+@subsection Options
+
+@table @option
+
+@item find_sources
+If set to @option{true}, print a list of found/available NDI sources and exit.
+Defaults to @option{false}.
+
+@item wait_sources
+Override time to wait until the number of online sources have changed.
+Defaults to @option{0.5}.
+
+@item allow_video_fields
+When this flag is @option{false}, all video that you receive will be progressive.
+Defaults to @option{true}.
+
+@item extra_ips
+If is set to list of comma separated ip addresses, scan for sources not only
+using mDNS but also use unicast ip addresses specified by this list.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List input devices:
+@example
+ffmpeg -f libndi_newtek -find_sources 1 -i dummy
+@end example
+
+@item
+List local and remote input devices:
+@example
+ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -find_sources 1 -i dummy
+@end example
+
+@item
+Restream to NDI:
+@example
+ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+@end example
+
+@item
+Restream remote NDI to local NDI:
+@example
+ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -i "DEV-5.REMOTE.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
+@end example
+
+
+@end itemize
+
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -1195,7 +1233,6 @@ Set the number of channels. Default is 2.
 
 @end table
 
-
 @section pulse
 
 PulseAudio input device.

doc/libav-merge.txt

@@ -95,7 +95,6 @@ Stuff that didn't reach the codebase:
   - 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
-- new bitstream reader (see http://ffmpeg.org/pipermail/ffmpeg-devel/2017-April/209609.html)
 - 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()
@@ -105,7 +104,6 @@ Stuff that didn't reach the codebase:
 Collateral damage that needs work locally:
 ------------------------------------------
 
-- Merge proresdec2.c and proresdec_lgpl.c
 - Merge proresenc_anatoliy.c and proresenc_kostya.c
 - Fix MIPS AC3 downmix
 

doc/mailing-list-faq.texi

@@ -47,7 +47,8 @@ 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?
 
-All you have to do is send an email:
+First you must @ref{How do I subscribe?, subscribe}. Then all you have to do is
+send an email:
 
 @itemize
 @item
@@ -57,49 +58,18 @@ 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
 
-If you are not subscribed to the mailing list then your question must be
-manually approved. Approval may take several days, but the wait is
-usually less. If you want the message to be sent with no delay then you
-must subscribe first. See @ref{How do I subscribe?}
-
-Please do not send a message, subscribe, and re-send the message: this
-results in duplicates, causes more work for the admins, and may lower
-your chance at getting an answer. However, you may do so if you first
-@ref{How do I delete my message in the moderation queue?, delete your original message from the moderation queue}.
+Note that the ffmpeg-devel mailing list does not require you to subscribe
+to send a message or patch, but ffmpeg-user and libav-user do require
+subscription.
 
 @chapter Subscribing / Unsubscribing
 
-@section What does subscribing do?
-
-Subscribing allows two things:
-
-@itemize
-@item
-Your messages will show up in the mailing list without waiting in the
-moderation queue and needing to be manually approved by a mailing list
-admin.
-
-@item
-You will receive all messages to the mailing list including replies to
-your messages. Non-subscribed users do not receive any messages.
-@end itemize
-
-@section Do I need to subscribe?
-
-No. You can still send a message to the mailing list without
-subscribing. See @ref{How do I ask a question or send a message to a mailing list?}
-
-However, your message will need to be manually approved by a mailing
-list admin, and you will not receive any mailing list messages or
-replies.
-
-You can ask to be CCd in your message, but replying users will
-sometimes forget to do so.
-
-You may also view and reply to messages via the @ref{Where are the archives?, archives}.
-
 @anchor{How do I subscribe?}
 @section How do I subscribe?
 
@@ -134,8 +104,6 @@ must be manually approved by a mailing list admin:
 These are:
 
 @itemize
-@item
-Messages from users who are @strong{not} subscribed.
 
 @item
 Messages that exceed the @ref{What is the message size limit?, message size limit}.
@@ -148,13 +116,12 @@ or is abusive towards others).
 
 @section How long does it take for my message in the moderation queue to be approved?
 
-The queue is usually checked once or twice a day, but on occasion
-several days may pass before someone checks the queue.
+The queue is usually checked daily to several times a week.
 
 @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 ffmpeg-user awaits moderator approval}.
+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.
 
@@ -175,6 +142,9 @@ 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:
@@ -203,9 +173,8 @@ Instead, use trimmed interleaved/inline replies (@url{https://lists.ffmpeg.org/p
 @anchor{What is the message size limit?}
 @section What is the message size limit?
 
-The message size limit is 500 kilobytes for the user lists and 1000
-kilobytes for ffmpeg-devel. Please provide links to larger files instead
-of attaching them.
+The message size limit is 1000 kilobytes. Please provide links to larger files
+instead of attaching them.
 
 @section Where can I upload sample files?
 

doc/muxers.texi

@@ -226,7 +226,12 @@ ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
 
 @table @option
 @item -min_seg_duration @var{microseconds}
-Set the segment length in microseconds.
+This is a deprecated option to set the segment length in microseconds, use @var{seg_duration} instead.
+@item -seg_duration @var{duration}
+Set the segment length in seconds (fractional value can be set). The value is
+treated as average segment duration when @var{use_template} is enabled and
+@var{use_timeline} is disabled and as minimum segment duration for all the other
+use cases.
 @item -window_size @var{size}
 Set the maximum number of segments kept in the manifest.
 @item -extra_window_size @var{size}
@@ -247,6 +252,8 @@ DASH-templated name to used for the initialization segment. Default is "init-str
 DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
 @item -utc_timing_url @var{utc_url}
 URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
+@item method @var{method}
+Use the given HTTP method to create output files. Generally set to PUT or POST.
 @item -http_user_agent @var{user_agent}
 Override User-Agent field in HTTP header. Applicable only for HTTP output.
 @item -http_persistent @var{http_persistent}
@@ -266,6 +273,30 @@ To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be use
 When no assignment is defined, this defaults to an AdaptationSet for each stream.
 @item -timeout @var{timeout}
 Set timeout for socket I/O operations. Applicable only for HTTP output.
+@item -index_correction @var{index_correction}
+Enable (1) or Disable (0) segment index correction logic. Applicable only when
+@var{use_template} is enabled and @var{use_timeline} is disabled.
+
+When enabled, the logic monitors the flow of segment indexes. If a streams's
+segment index value is not at the expected real time position, then the logic
+corrects that index value.
+
+Typically this logic is needed in live streaming use cases. The network bandwidth
+fluctuations are common during long run streaming. Each fluctuation can cause
+the segment indexes fall behind the expected real time position.
+@item -format_options @var{options_list}
+Set container format (mp4/webm) options using a @code{:} separated list of
+key=value parameters. Values containing @code{:} special characters must be
+escaped.
+
+@item dash_segment_type @var{dash_segment_type}
+Possible values:
+@item mp4
+If this flag is set, the dash segment files will be in in ISOBMFF format. This is the default format.
+
+@item webm
+If this flag is set, the dash segment files will be in in WebM format.
+
 @end table
 
 @anchor{framecrc}
@@ -586,7 +617,7 @@ This example will produce the playlist, @file{out.m3u8}, and segment files:
 but only the file name part without any path info will be contained in the m3u8 segment list.
 Should a relative path be specified, the path of the created segment
 files will be relative to the current working directory.
-When use_localtime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
+When strftime_mkdir is set, the whole expanded value of @var{filename} will be written into the m3u8 segment list.
 
 When @code{var_stream_map} is set with two or more variant streams, the
 @var{filename} pattern must contain the string "%v", this string specifies
@@ -615,34 +646,40 @@ This example will produce the playlists segment file sets:
 @file{vs1/file_000.ts}, @file{vs1/file_001.ts}, @file{vs1/file_002.ts}, etc.
 
 @item use_localtime
+Same as strftime option, will be deprecated.
+
+@item strftime
 Use strftime() on @var{filename} to expand the segment filename with localtime.
 The segment number is also available in this mode, but to use it, you need to specify second_level_segment_index
 hls_flag and %%d will be the specifier.
 @example
-ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
 @file{file-20160215-1455569023.ts}, @file{file-20160215-1455569024.ts}, etc.
 Note: On some systems/environments, the @code{%s} specifier is not available. See
   @code{strftime()} documentation.
 @example
-ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
+ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
 @end example
 This example will produce the playlist, @file{out.m3u8}, and segment files:
 @file{file-20160215-0001.ts}, @file{file-20160215-0002.ts}, etc.
 
 @item use_localtime_mkdir
-Used together with -use_localtime, it will create all subdirectories which
+Same as strftime_mkdir option, will be deprecated .
+
+@item strftime_mkdir
+Used together with -strftime_mkdir, it will create all subdirectories which
 is expanded in @var{filename}.
 @example
-ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will create a directory 201560215 (if it does not exist), and then
 produce the playlist, @file{out.m3u8}, and segment files:
 @file{20160215/file-20160215-1455569023.ts}, @file{20160215/file-20160215-1455569024.ts}, etc.
 
 @example
-ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
+ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
 @end example
 This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then
 produce the playlist, @file{out.m3u8}, and segment files:
@@ -727,17 +764,17 @@ Possible values:
 
 @table @samp
 @item mpegts
-If this flag is set, the hls segment files will format to mpegts.
-the mpegts files is used in all hls versions.
+Output segment files in MPEG-2 Transport Stream format. This is
+compatible with all HLS versions.
 
 @item fmp4
-If this flag is set, the hls segment files will format to fragment mp4 looks like dash.
-the fmp4 files is used in hls after version 7.
+Output segment files in fragmented MP4 format, similar to MPEG-DASH.
+fmp4 files may be used in HLS version 7 and above.
 
 @end table
 
 @item hls_fmp4_init_filename @var{filename}
-set filename to the fragment files header file, default filename is @file{init.mp4}.
+Set filename to the fragment files header file, default filename is @file{init.mp4}.
 
 When @code{var_stream_map} is set with two or more variant streams, the
 @var{filename} pattern must contain the string "%v", this string specifies
@@ -802,24 +839,24 @@ Generate @code{EXT-X-PROGRAM-DATE-TIME} tags.
 
 @item second_level_segment_index
 Makes it possible to use segment indexes as %%d in hls_segment_filename expression
-besides date/time values when use_localtime is on.
+besides date/time values when strftime is on.
 To get fixed width numbers with trailing zeroes, %%0xd format is available where x is the required width.
 
 @item second_level_segment_size
 Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename
-expression besides date/time values when use_localtime is on.
+expression besides date/time values when strftime is on.
 To get fixed width numbers with trailing zeroes, %%0xs format is available where x is the required width.
 
 @item second_level_segment_duration
 Makes it possible to use segment duration (calculated  in microseconds) as %%t in hls_segment_filename
-expression besides date/time values when use_localtime is on.
+expression besides date/time values when strftime is on.
 To get fixed width numbers with trailing zeroes, %%0xt format is available where x is the required width.
 
 @example
 ffmpeg -i sample.mpeg \
    -f hls -hls_time 3 -hls_list_size 5 \
    -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
-   -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
+   -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
 @end example
 This will produce segments like this:
 @file{segment_20170102194334_0003_00122200_0000003000000.ts}, @file{segment_20170102194334_0004_00120072_0000003000000.ts} etc.
@@ -1313,6 +1350,18 @@ be negative. This enables the initial sample to have DTS/CTS of zero, and
 reduces the need for edit lists for some cases such as video tracks with
 B-frames. Additionally, eases conformance with the DASH-IF interoperability
 guidelines.
+
+This option is implicitly set when writing ismv (Smooth Streaming) files.
+@item -write_prft
+Write producer time reference box (PRFT) with a specified time source for the
+NTP field in the PRFT box. Set value as @samp{wallclock} to specify timesource
+as wallclock time and @samp{pts} to specify timesource as input packets' PTS
+values.
+
+Setting value to @samp{pts} is applicable only for a live encoding use case,
+where PTS values are set as as wallclock time at the source. For example, an
+encoding use case with decklink capture source where @option{video_pts} and
+@option{audio_pts} are set to @samp{abs_wallclock}.
 @end table
 
 @subsection Example
@@ -2021,20 +2070,35 @@ ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0
 @anchor{tee}
 @section tee
 
-The tee muxer can be used to write the same data to several files or any
-other kind of muxer. It can be used, for example, to both stream a video to
-the network and save it to disk at the same time.
+The tee muxer can be used to write the same data to several outputs, such as files or streams.
+It can be used, for example, to stream a video over a network and save it to disk at the same time.
 
 It is different from specifying several outputs to the @command{ffmpeg}
-command-line tool because the audio and video data will be encoded only once
-with the tee muxer; encoding can be a very expensive process. It is not
-useful when using the libavformat API directly because it is then possible
-to feed the same packets to several muxers directly.
+command-line tool. With the tee muxer, the audio and video data will be encoded only once.
+With conventional multiple outputs, multiple encoding operations in parallel are initiated,
+which can be a very expensive process. The tee muxer is not useful when using the libavformat API
+directly because it is then possible to feed the same packets to several muxers directly.
+
+Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-select
+output streams. So all streams intended for output must be specified using @code{-map}. See
+the examples below.
+
+Some encoders may need different options depending on the output format;
+the auto-detection of this can not work with the tee muxer, so they need to be explicitly specified.
+The main example is the @option{global_header} flag.
+
+The slave outputs are specified in the file name given to the muxer,
+separated by '|'. If any of the slave name contains the '|' separator,
+leading or trailing spaces or any special character, those must be
+escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
+section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
+
+@subsection Options
 
 @table @option
 
 @item use_fifo @var{bool}
-If set to 1, slave outputs will be processed in separate thread using @ref{fifo}
+If set to 1, slave outputs will be processed in separate threads using the @ref{fifo}
 muxer. This allows to compensate for different speed/latency/reliability of
 outputs and setup transparent recovery. By default this feature is turned off.
 
@@ -2043,12 +2107,6 @@ Options to pass to fifo pseudo-muxer instances. See @ref{fifo}.
 
 @end table
 
-The slave outputs are specified in the file name given to the muxer,
-separated by '|'. If any of the slave name contains the '|' separator,
-leading or trailing spaces or any special character, it must be
-escaped (see @ref{quoting_and_escaping,,the "Quoting and escaping"
-section in the ffmpeg-utils(1) manual,ffmpeg-utils}).
-
 Muxer options can be specified for each slave by prepending them as a list of
 @var{key}=@var{value} pairs separated by ':', between square brackets. If
 the options values contain a special character or the ':' separator, they
@@ -2057,13 +2115,27 @@ must be escaped; note that this is a second level escaping.
 The following special options are also recognized:
 @table @option
 @item f
-Specify the format name. Useful if it cannot be guessed from the
-output name suffix.
+Specify the format name. Required if it cannot be guessed from the
+output URL.
 
 @item bsfs[/@var{spec}]
 Specify a list of bitstream filters to apply to the specified
 output.
 
+It is possible to specify to which streams a given bitstream filter
+applies, by appending a stream specifier to the option separated by
+@code{/}. @var{spec} must be a stream specifier (see @ref{Format
+stream specifiers}).
+
+If the stream specifier is not specified, the bitstream filters will be
+applied to all streams in the output. This will cause that output operation
+to fail if the output contains streams to which the bitstream filter cannot
+be applied e.g. @code{h264_mp4toannexb} being applied to an output containing an audio stream.
+
+Options for a bitstream filter must be specified in the form of @code{opt=value}.
+
+Several bitstream filters can be specified, separated by ",".
+
 @item use_fifo @var{bool}
 This allows to override tee muxer use_fifo option for individual slave muxer.
 
@@ -2071,19 +2143,13 @@ This allows to override tee muxer use_fifo option for individual slave muxer.
 This allows to override tee muxer fifo_options for individual slave muxer.
 See @ref{fifo}.
 
-It is possible to specify to which streams a given bitstream filter
-applies, by appending a stream specifier to the option separated by
-@code{/}. @var{spec} must be a stream specifier (see @ref{Format
-stream specifiers}).  If the stream specifier is not specified, the
-bitstream filters will be applied to all streams in the output.
-
-Several bitstream filters can be specified, separated by ",".
-
 @item select
 Select the streams that should be mapped to the slave output,
 specified by a stream specifier. If not specified, this defaults to
-all the input streams. You may use multiple stream specifiers
-separated by commas (@code{,}) e.g.: @code{a:0,v}
+all the mapped streams. This will cause that output operation to fail
+if the output format does not accept all mapped streams.
+
+You may use multiple stream specifiers separated by commas (@code{,}) e.g.: @code{a:0,v}
 
 @item onfail
 Specify behaviour on output failure. This can be set to either @code{abort} (which is
@@ -2097,7 +2163,7 @@ will continue without being affected.
 @itemize
 @item
 Encode something and both archive it in a WebM file and stream it
-as MPEG-TS over UDP (the streams need to be explicitly mapped):
+as MPEG-TS over UDP:
 @example
 ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
   "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
@@ -2120,23 +2186,19 @@ option is applied to @file{out.aac} in order to make it contain only
 audio packets.
 @example
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
+       -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
 @end example
 
 @item
-As below, but select only stream @code{a:1} for the audio output. Note
+As above, but select only stream @code{a:1} for the audio output. Note
 that a second level escaping must be performed, as ":" is a special
 character used to separate options.
 @example
 ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-       -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
+       -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"
 @end example
 @end itemize
 
-Note: some codecs may need different options depending on the output format;
-the auto-detection of this can not work with the tee muxer. The main example
-is the @option{global_header} flag.
-
 @section webm_dash_manifest
 
 WebM DASH Manifest muxer.

doc/outdevs.texi

@@ -140,7 +140,8 @@ device with @command{-list_formats 1}. Audio sample rate is always 48 kHz.
 
 @item list_devices
 If set to @option{true}, print a list of devices and exit.
-Defaults to @option{false}.
+Defaults to @option{false}. Alternatively you can 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.
@@ -150,6 +151,10 @@ Defaults to @option{false}.
 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}.
+
 @end table
 
 @subsection Examples
@@ -182,6 +187,35 @@ ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLi
 
 @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 libndi_newtek
 
 The libndi_newtek output device provides playback capabilities for using NDI (Network
@@ -227,35 +261,6 @@ ffmpeg -i "udp://@@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "sc
 
 @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.
 
@@ -393,9 +398,18 @@ Set the SDL window size, can be a string of the form
 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
@@ -420,6 +434,10 @@ ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL out
 
 sndio audio output device.
 
+@section v4l2
+
+Video4Linux2 output device.
+
 @section xv
 
 XV (XVideo) output device.

doc/platform.texi

@@ -148,16 +148,11 @@ 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 2012 or earlier using a C99-to-C89 conversion utility
-and wrapper, or with MSVC 2013 and ICL natively.
+FFmpeg can be built with MSVC 2013 or later.
 
 You will need the following prerequisites:
 
 @itemize
-@item @uref{https://github.com/libav/c99-to-c89/, C99-to-C89 Converter & Wrapper}
-(if using MSVC 2012 or earlier)
-@item @uref{http://code.google.com/p/msinttypes/, msinttypes}
-(if using MSVC 2012 or earlier)
 @item @uref{http://msys2.github.io/, MSYS2}
 @item @uref{http://www.nasm.us/, NASM}
 (Also available via MSYS2's package manager.)
@@ -166,16 +161,13 @@ You will need the following prerequisites:
 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}. If using MSVC 2012 or
-earlier, place @code{c99wrap.exe} and @code{c99conv.exe} somewhere in your
-@code{PATH} as well.
+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. If using MSVC
-2012 or earlier, place @code{inttypes.h} somewhere the compiler can see too.
+@code{--extra-cflags}/@code{--extra-ldflags} configure options.
 
 Finally, run:
 
@@ -217,8 +209,6 @@ can see.
 
 @item FFmpeg has been tested with the following on i686 and x86_64:
 @itemize
-@item Visual Studio 2010 Pro and Express
-@item Visual Studio 2012 Pro and Express
 @item Visual Studio 2013 Pro and Express
 @item Intel Composer XE 2013
 @item Intel Composer XE 2013 SP1

doc/protocols.texi

@@ -1210,6 +1210,17 @@ IP Type of Service. Applies to sender only. Default value is 0xB8.
 @item ipttl=@var{ttl}
 IP Time To Live. Applies to sender only. Default value is 64.
 
+@item latency
+Timestamp-based Packet Delivery Delay.
+Used to absorb bursts of missed packet retransmissions.
+This flag sets both @option{rcvlatency} and @option{peerlatency}
+to the same value. Note that prior to version 1.3.0
+this is the only flag to set the latency, however
+this is effectively equivalent to setting @option{peerlatency},
+when side is sender and @option{rcvlatency}
+when side is receiver, and the bidirectional stream
+sending is not supported.
+
 @item listen_timeout
 Set socket listen timeout.
 
@@ -1255,6 +1266,25 @@ only if @option{pbkeylen} is non-zero. It is used on
 the receiver only if the received data is encrypted.
 The configured passphrase cannot be recovered (write-only).
 
+@item payload_size=@var{bytes}
+Sets the maximum declared size of a packet transferred
+during the single call to the sending function in Live
+mode. Use 0 if this value isn't used (which is default in
+file mode).
+Default is -1 (automatic), which typically means MPEG-TS;
+if you are going to use SRT
+to send any different kind of payload, such as, for example,
+wrapping a live stream in very small frames, then you can
+use a bigger maximum frame size, though not greater than
+1456 bytes.
+
+@item pkt_size=@var{bytes}
+Alias for @samp{payload_size}.
+
+@item peerlatency
+The latency value (as described in @option{rcvlatency}) that is
+set by the sender side as a minimum value for the receiver.
+
 @item pbkeylen=@var{bytes}
 Sender encryption key length, in bytes.
 Only can be set to 0, 16, 24 and 32.
@@ -1263,11 +1293,23 @@ Not required on receiver (set to 0),
 key size obtained from sender in HaiCrypt handshake.
 Default value is 0.
 
+@item rcvlatency
+The time that should elapse since the moment when the
+packet was sent and the moment when it's delivered to
+the receiver application in the receiving function.
+This time should be a buffer time large enough to cover
+the time spent for sending, unexpectedly extended RTT
+time, and the time needed to retransmit the lost UDP
+packet. The effective latency value will be the maximum
+of this options' value and the value of @option{peerlatency}
+set by the peer side. Before version 1.3.0 this option
+is only available as @option{latency}.
+
 @item recv_buffer_size=@var{bytes}
-Set receive buffer size, expressed in bytes.
+Set UDP receive buffer size, expressed in bytes.
 
 @item send_buffer_size=@var{bytes}
-Set send buffer size, expressed in bytes.
+Set UDP send buffer size, expressed in bytes.
 
 @item rw_timeout
 Set raise error timeout for read/write optations.
@@ -1287,9 +1329,86 @@ have no chance of being delivered in time. It was
 automatically enabled in the sender if the receiver
 supports it.
 
-@item tsbpddelay
-Timestamp-based Packet Delivery Delay.
-Used to absorb burst of missed packet retransmission.
+@item sndbuf=@var{bytes}
+Set send buffer size, expressed in bytes.
+
+@item rcvbuf=@var{bytes}
+Set receive buffer size, expressed in bytes.
+
+Receive buffer must not be greater than @option{ffs}.
+
+@item lossmaxttl=@var{packets}
+The value up to which the Reorder Tolerance may grow. When
+Reorder Tolerance is > 0, then packet loss report is delayed
+until that number of packets come in. Reorder Tolerance
+increases every time a "belated" packet has come, but it
+wasn't due to retransmission (that is, when UDP packets tend
+to come out of order), with the difference between the latest
+sequence and this packet's sequence, and not more than the
+value of this option. By default it's 0, which means that this
+mechanism is turned off, and the loss report is always sent
+immediately upon experiencing a "gap" in sequences.
+
+@item minversion
+The minimum SRT version that is required from the peer. A connection
+to a peer that does not satisfy the minimum version requirement
+will be rejected.
+
+The version format in hex is 0xXXYYZZ for x.y.z in human readable
+form.
+
+@item streamid=@var{string}
+A string limited to 512 characters that can be set on the socket prior
+to connecting. This stream ID will be able to be retrieved by the
+listener side from the socket that is returned from srt_accept and
+was connected by a socket with that set stream ID. SRT does not enforce
+any special interpretation of the contents of this string.
+This option doesn’t make sense in Rendezvous connection; the result
+might be that simply one side will override the value from the other
+side and it’s the matter of luck which one would win
+
+@item smoother=@var{live|file}
+The type of Smoother used for the transmission for that socket, which
+is responsible for the transmission and congestion control. The Smoother
+type must be exactly the same on both connecting parties, otherwise
+the connection is rejected.
+
+@item messageapi=@var{1|0}
+When set, this socket uses the Message API, otherwise it uses Buffer
+API. Note that in live mode (see @option{transtype}) there’s only
+message API available. In File mode you can chose to use one of two modes:
+
+Stream API (default, when this option is false). In this mode you may
+send as many data as you wish with one sending instruction, or even use
+dedicated functions that read directly from a file. The internal facility
+will take care of any speed and congestion control. When receiving, you
+can also receive as many data as desired, the data not extracted will be
+waiting for the next call. There is no boundary between data portions in
+the Stream mode.
+
+Message API. In this mode your single sending instruction passes exactly
+one piece of data that has boundaries (a message). Contrary to Live mode,
+this message may span across multiple UDP packets and the only size
+limitation is that it shall fit as a whole in the sending buffer. The
+receiver shall use as large buffer as necessary to receive the message,
+otherwise the message will not be given up. When the message is not
+complete (not all packets received or there was a packet loss) it will
+not be given up.
+
+@item transtype=@var{live|file}
+Sets the transmission type for the socket, in particular, setting this
+option sets multiple other parameters to their default values as required
+for a particular transmission type.
+
+live: Set options as for live transmission. In this mode, you should
+send by one sending instruction only so many data that fit in one UDP packet,
+and limited to the value defined first in @option{payload_size} (1316 is
+default in this mode). There is no speed control in this mode, only the
+bandwidth control, if configured, in order to not exceed the bandwidth with
+the overhead transmission (retransmitted and control packets).
+
+file: Set options as for non-live transmission. See @option{messageapi}
+for further explanations
 
 @end table
 
@@ -1397,6 +1516,9 @@ Set send buffer size, expressed bytes.
 
 @item tcp_nodelay=@var{1|0}
 Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.
+
+@item tcp_mss=@var{bytes}
+Set maximum segment size for outgoing TCP packets, expressed in bytes.
 @end table
 
 The following example shows how to setup a listening TCP connection
@@ -1503,9 +1625,8 @@ packet bursts.
 Override the local UDP port to bind with.
 
 @item localaddr=@var{addr}
-Choose the local IP address. This is useful e.g. if sending multicast
-and the host has multiple interfaces, where the user can choose
-which interface to send on by specifying the IP address of that interface.
+Local IP address of a network interface used for sending packets or joining
+multicast groups.
 
 @item pkt_size=@var{size}
 Set the size in bytes of UDP packets.
@@ -1528,12 +1649,12 @@ For receiving, this gives the benefit of only receiving packets from
 the specified peer address/port.
 
 @item sources=@var{address}[,@var{address}]
-Only receive packets sent to the multicast group from one of the
-specified sender IP addresses.
+Only receive packets sent from the specified addresses. In case of multicast,
+also subscribe to multicast traffic coming from these addresses only.
 
 @item block=@var{address}[,@var{address}]
-Ignore packets sent to the multicast group from the specified
-sender IP addresses.
+Ignore packets sent from the specified addresses. In case of multicast, also
+exclude the source addresses in the multicast subscription.
 
 @item fifo_size=@var{units}
 Set the UDP receiving circular buffer size, expressed as a number of

doc/writing_filters.txt

@@ -418,4 +418,4 @@ done:
 
 When all of this is done, you can submit your patch to the ffmpeg-devel
 mailing-list for review.  If you need any help, feel free to come on our IRC
-channel, #ffmpeg-devel on irc.freenode.net.
+channel, #ffmpeg-devel on irc.libera.chat.

fftools/cmdutils.c

@@ -1018,7 +1018,7 @@ static int init_report(const char *env)
         av_free(key);
     }
 
-    av_bprint_init(&filename, 0, 1);
+    av_bprint_init(&filename, 0, AV_BPRINT_SIZE_AUTOMATIC);
     expand_filename_template(&filename,
                              av_x_if_null(filename_template, "%p-%t.log"), tm);
     av_free(filename_template);
@@ -1414,6 +1414,16 @@ static void print_codec(const AVCodec *c)
                            AV_CODEC_CAP_SLICE_THREADS |
                            AV_CODEC_CAP_AUTO_THREADS))
         printf("threads ");
+    if (c->capabilities & AV_CODEC_CAP_AVOID_PROBING)
+        printf("avoidprobe ");
+    if (c->capabilities & AV_CODEC_CAP_INTRA_ONLY)
+        printf("intraonly ");
+    if (c->capabilities & AV_CODEC_CAP_LOSSLESS)
+        printf("lossless ");
+    if (c->capabilities & AV_CODEC_CAP_HARDWARE)
+        printf("hardware ");
+    if (c->capabilities & AV_CODEC_CAP_HYBRID)
+        printf("hybrid ");
     if (!c->capabilities)
         printf("none");
     printf("\n");
@@ -1434,6 +1444,17 @@ static void print_codec(const AVCodec *c)
         printf("\n");
     }
 
+    if (avcodec_get_hw_config(c, 0)) {
+        printf("    Supported hardware devices: ");
+        for (int i = 0;; i++) {
+            const AVCodecHWConfig *config = avcodec_get_hw_config(c, i);
+            if (!config)
+                break;
+            printf("%s ", av_hwdevice_get_type_name(config->device_type));
+        }
+        printf("\n");
+    }
+
     if (c->supported_framerates) {
         const AVRational *fps = c->supported_framerates;
 
@@ -1935,7 +1956,10 @@ static void show_help_bsf(const char *name)
 {
     const AVBitStreamFilter *bsf = av_bsf_get_by_name(name);
 
-    if (!bsf) {
+    if (!name) {
+        av_log(NULL, AV_LOG_ERROR, "No bitstream filter name specified.\n");
+        return;
+    } else if (!bsf) {
         av_log(NULL, AV_LOG_ERROR, "Unknown bit stream filter '%s'.\n", name);
         return;
     }

fftools/ffmpeg.c

@@ -120,8 +120,14 @@ const char *const forced_keyframes_const_names[] = {
     NULL
 };
 
+typedef struct BenchmarkTimeStamps {
+    int64_t real_usec;
+    int64_t user_usec;
+    int64_t sys_usec;
+} BenchmarkTimeStamps;
+
 static void do_video_stats(OutputStream *ost, int frame_size);
-static int64_t getutime(void);
+static BenchmarkTimeStamps get_benchmark_time_stamps(void);
 static int64_t getmaxrss(void);
 static int ifilter_has_all_input_formats(FilterGraph *fg);
 
@@ -133,7 +139,7 @@ static int64_t decode_error_stat[2];
 
 static int want_sdp = 1;
 
-static int current_time;
+static BenchmarkTimeStamps current_time;
 AVIOContext *progress_avio = NULL;
 
 static uint8_t *subtitle_out;
@@ -561,6 +567,7 @@ static void ffmpeg_cleanup(int ret)
         ost->audio_channels_mapped = 0;
 
         av_dict_free(&ost->sws_dict);
+        av_dict_free(&ost->swr_opts);
 
         avcodec_free_context(&ost->enc_ctx);
         avcodec_parameters_free(&ost->ref_par);
@@ -653,7 +660,7 @@ static void abort_codec_experimental(AVCodec *c, int encoder)
 static void update_benchmark(const char *fmt, ...)
 {
     if (do_benchmark_all) {
-        int64_t t = getutime();
+        BenchmarkTimeStamps t = get_benchmark_time_stamps();
         va_list va;
         char buf[1024];
 
@@ -661,7 +668,11 @@ static void update_benchmark(const char *fmt, ...)
             va_start(va, fmt);
             vsnprintf(buf, sizeof(buf), fmt, va);
             va_end(va);
-            av_log(NULL, AV_LOG_INFO, "bench: %8"PRIu64" %s \n", t - current_time, buf);
+            av_log(NULL, AV_LOG_INFO,
+                   "bench: %8" PRIu64 " user %8" PRIu64 " sys %8" PRIu64 " real %s \n",
+                   t.user_usec - current_time.user_usec,
+                   t.sys_usec - current_time.sys_usec,
+                   t.real_usec - current_time.real_usec, buf);
         }
         current_time = t;
     }
@@ -714,11 +725,11 @@ static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int u
             if (ret < 0)
                 exit_program(1);
         }
-        ret = av_packet_ref(&tmp_pkt, pkt);
+        ret = av_packet_make_refcounted(pkt);
         if (ret < 0)
             exit_program(1);
+        av_packet_move_ref(&tmp_pkt, pkt);
         av_fifo_generic_write(ost->muxing_queue, &tmp_pkt, sizeof(tmp_pkt), NULL);
-        av_packet_unref(pkt);
         return;
     }
 
@@ -762,7 +773,7 @@ static void write_packet(OutputFile *of, AVPacket *pkt, OutputStream *ost, int u
                      - FFMIN3(pkt->pts, pkt->dts, ost->last_mux_dts + 1)
                      - FFMAX3(pkt->pts, pkt->dts, ost->last_mux_dts + 1);
         }
-        if ((st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO || st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO) &&
+        if ((st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO || st->codecpar->codec_type == AVMEDIA_TYPE_VIDEO || st->codecpar->codec_type == AVMEDIA_TYPE_SUBTITLE) &&
             pkt->dts != AV_NOPTS_VALUE &&
             !(st->codecpar->codec_id == AV_CODEC_ID_VP9 && ost->stream_copy) &&
             ost->last_mux_dts != AV_NOPTS_VALUE) {
@@ -1111,7 +1122,7 @@ static void do_video_out(OutputFile *of,
             format_video_sync != VSYNC_PASSTHROUGH &&
             format_video_sync != VSYNC_DROP) {
             if (delta0 < -0.6) {
-                av_log(NULL, AV_LOG_WARNING, "Past duration %f too large\n", -delta0);
+                av_log(NULL, AV_LOG_VERBOSE, "Past duration %f too large\n", -delta0);
             } else
                 av_log(NULL, AV_LOG_DEBUG, "Clipping frame in rate conversion by %f\n", -delta0);
             sync_ipts = ost->sync_opts;
@@ -1226,8 +1237,12 @@ static void do_video_out(OutputFile *of,
         in_picture->quality = enc->global_quality;
         in_picture->pict_type = 0;
 
+        if (ost->forced_kf_ref_pts == AV_NOPTS_VALUE &&
+            in_picture->pts != AV_NOPTS_VALUE)
+            ost->forced_kf_ref_pts = in_picture->pts;
+
         pts_time = in_picture->pts != AV_NOPTS_VALUE ?
-            in_picture->pts * av_q2d(enc->time_base) : NAN;
+            (in_picture->pts - ost->forced_kf_ref_pts) * av_q2d(enc->time_base) : NAN;
         if (ost->forced_kf_index < ost->forced_kf_count &&
             in_picture->pts >= ost->forced_kf_pts[ost->forced_kf_index]) {
             ost->forced_kf_index++;
@@ -1675,7 +1690,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
 
     vid = 0;
     av_bprint_init(&buf, 0, AV_BPRINT_SIZE_AUTOMATIC);
-    av_bprint_init(&buf_script, 0, 1);
+    av_bprint_init(&buf_script, 0, AV_BPRINT_SIZE_AUTOMATIC);
     for (i = 0; i < nb_output_streams; i++) {
         float q = -1;
         ost = output_streams[i];
@@ -1696,7 +1711,7 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
             av_bprintf(&buf, "frame=%5d fps=%3.*f q=%3.1f ",
                      frame_number, fps < 9.95, fps, q);
             av_bprintf(&buf_script, "frame=%d\n", frame_number);
-            av_bprintf(&buf_script, "fps=%.1f\n", fps);
+            av_bprintf(&buf_script, "fps=%.2f\n", fps);
             av_bprintf(&buf_script, "stream_%d_%d_q=%.1f\n",
                        ost->file_index, ost->index, q);
             if (is_last_report)
@@ -1780,9 +1795,11 @@ static void print_report(int is_last_report, int64_t timer_start, int64_t cur_ti
     if (total_size < 0) av_bprintf(&buf_script, "total_size=N/A\n");
     else                av_bprintf(&buf_script, "total_size=%"PRId64"\n", total_size);
     if (pts == AV_NOPTS_VALUE) {
+        av_bprintf(&buf_script, "out_time_us=N/A\n");
         av_bprintf(&buf_script, "out_time_ms=N/A\n");
         av_bprintf(&buf_script, "out_time=N/A\n");
     } else {
+        av_bprintf(&buf_script, "out_time_us=%"PRId64"\n", pts);
         av_bprintf(&buf_script, "out_time_ms=%"PRId64"\n", pts);
         av_bprintf(&buf_script, "out_time=%s%02d:%02d:%02d.%06d\n",
                    hours_sign, hours, mins, secs, us);
@@ -2094,10 +2111,12 @@ static void check_decode_result(InputStream *ist, int *got_output, int ret)
     if (ret < 0 && exit_on_error)
         exit_program(1);
 
-    if (exit_on_error && *got_output && ist) {
+    if (*got_output && ist) {
         if (ist->decoded_frame->decode_error_flags || (ist->decoded_frame->flags & AV_FRAME_FLAG_CORRUPT)) {
-            av_log(NULL, AV_LOG_FATAL, "%s: corrupt decoded frame in stream %d\n", input_files[ist->file_index]->ctx->url, ist->st->index);
-            exit_program(1);
+            av_log(NULL, exit_on_error ? AV_LOG_FATAL : AV_LOG_WARNING,
+                   "%s: corrupt decoded frame in stream %d\n", input_files[ist->file_index]->ctx->url, ist->st->index);
+            if (exit_on_error)
+                exit_program(1);
         }
     }
 }
@@ -2121,9 +2140,6 @@ static int ifilter_send_frame(InputFilter *ifilter, AVFrame *frame)
 
     /* determine if the parameters for this input changed */
     need_reinit = ifilter->format != frame->format;
-    if (!!ifilter->hw_frames_ctx != !!frame->hw_frames_ctx ||
-        (ifilter->hw_frames_ctx && ifilter->hw_frames_ctx->data != frame->hw_frames_ctx->data))
-        need_reinit = 1;
 
     switch (ifilter->ist->st->codecpar->codec_type) {
     case AVMEDIA_TYPE_AUDIO:
@@ -2137,6 +2153,13 @@ static int ifilter_send_frame(InputFilter *ifilter, AVFrame *frame)
         break;
     }
 
+    if (!ifilter->ist->reinit_filters && fg->graph)
+        need_reinit = 0;
+
+    if (!!ifilter->hw_frames_ctx != !!frame->hw_frames_ctx ||
+        (ifilter->hw_frames_ctx && ifilter->hw_frames_ctx->data != frame->hw_frames_ctx->data))
+        need_reinit = 1;
+
     if (need_reinit) {
         ret = ifilter_parameters_from_frame(ifilter, frame);
         if (ret < 0)
@@ -2696,8 +2719,13 @@ static int process_input_packet(InputStream *ist, const AVPacket *pkt, int no_eo
         ist->dts = ist->next_dts;
         switch (ist->dec_ctx->codec_type) {
         case AVMEDIA_TYPE_AUDIO:
-            ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
-                             ist->dec_ctx->sample_rate;
+            av_assert1(pkt->duration >= 0);
+            if (ist->dec_ctx->sample_rate) {
+                ist->next_dts += ((int64_t)AV_TIME_BASE * ist->dec_ctx->frame_size) /
+                                  ist->dec_ctx->sample_rate;
+            } else {
+                ist->next_dts += av_rescale_q(pkt->duration, ist->st->time_base, AV_TIME_BASE_Q);
+            }
             break;
         case AVMEDIA_TYPE_VIDEO:
             if (ist->framerate.num) {
@@ -3051,7 +3079,13 @@ static int init_output_stream_streamcopy(OutputStream *ost)
                "Error setting up codec context options.\n");
         return ret;
     }
-    avcodec_parameters_from_context(par_src, ost->enc_ctx);
+
+    ret = avcodec_parameters_from_context(par_src, ost->enc_ctx);
+    if (ret < 0) {
+        av_log(NULL, AV_LOG_FATAL,
+               "Error getting reference codec parameters.\n");
+        return ret;
+    }
 
     if (!codec_tag) {
         unsigned int codec_tag_tmp;
@@ -3379,6 +3413,12 @@ static int init_output_stream_encode(OutputStream *ost)
             enc_ctx->bits_per_raw_sample = frame_bits_per_raw_sample;
         }
 
+        if (ost->top_field_first == 0) {
+            enc_ctx->field_order = AV_FIELD_BB;
+        } else if (ost->top_field_first == 1) {
+            enc_ctx->field_order = AV_FIELD_TT;
+        }
+
         if (ost->forced_keyframes) {
             if (!strncmp(ost->forced_keyframes, "expr:", 5)) {
                 ret = av_expr_parse(&ost->forced_keyframes_pexpr, ost->forced_keyframes+5,
@@ -3465,6 +3505,23 @@ static int init_output_stream(OutputStream *ost, char *error, int error_len)
                 return ret;
             }
         }
+        if (ist && ist->dec->type == AVMEDIA_TYPE_SUBTITLE && ost->enc->type == AVMEDIA_TYPE_SUBTITLE) {
+            int input_props = 0, output_props = 0;
+            AVCodecDescriptor const *input_descriptor =
+                avcodec_descriptor_get(dec->codec_id);
+            AVCodecDescriptor const *output_descriptor =
+                avcodec_descriptor_get(ost->enc_ctx->codec_id);
+            if (input_descriptor)
+                input_props = input_descriptor->props & (AV_CODEC_PROP_TEXT_SUB | AV_CODEC_PROP_BITMAP_SUB);
+            if (output_descriptor)
+                output_props = output_descriptor->props & (AV_CODEC_PROP_TEXT_SUB | AV_CODEC_PROP_BITMAP_SUB);
+            if (input_props && output_props && input_props != output_props) {
+                snprintf(error, error_len,
+                         "Subtitle encoding currently only possible from text to text "
+                         "or bitmap to bitmap");
+                return AVERROR_INVALIDDATA;
+            }
+        }
 
         if ((ret = avcodec_open2(ost->enc_ctx, codec, &ost->encoder_opts)) < 0) {
             if (ret == AVERROR_EXPERIMENTAL)
@@ -4011,49 +4068,63 @@ static void *input_thread(void *arg)
     return NULL;
 }
 
+static void free_input_thread(int i)
+{
+    InputFile *f = input_files[i];
+    AVPacket pkt;
+
+    if (!f || !f->in_thread_queue)
+        return;
+    av_thread_message_queue_set_err_send(f->in_thread_queue, AVERROR_EOF);
+    while (av_thread_message_queue_recv(f->in_thread_queue, &pkt, 0) >= 0)
+        av_packet_unref(&pkt);
+
+    pthread_join(f->thread, NULL);
+    f->joined = 1;
+    av_thread_message_queue_free(&f->in_thread_queue);
+}
+
 static void free_input_threads(void)
 {
     int i;
 
-    for (i = 0; i < nb_input_files; i++) {
-        InputFile *f = input_files[i];
-        AVPacket pkt;
+    for (i = 0; i < nb_input_files; i++)
+        free_input_thread(i);
+}
 
-        if (!f || !f->in_thread_queue)
-            continue;
-        av_thread_message_queue_set_err_send(f->in_thread_queue, AVERROR_EOF);
-        while (av_thread_message_queue_recv(f->in_thread_queue, &pkt, 0) >= 0)
-            av_packet_unref(&pkt);
+static int init_input_thread(int i)
+{
+    int ret;
+    InputFile *f = input_files[i];
 
-        pthread_join(f->thread, NULL);
-        f->joined = 1;
+    if (nb_input_files == 1)
+        return 0;
+
+    if (f->ctx->pb ? !f->ctx->pb->seekable :
+        strcmp(f->ctx->iformat->name, "lavfi"))
+        f->non_blocking = 1;
+    ret = av_thread_message_queue_alloc(&f->in_thread_queue,
+                                        f->thread_queue_size, sizeof(AVPacket));
+    if (ret < 0)
+        return ret;
+
+    if ((ret = pthread_create(&f->thread, NULL, input_thread, f))) {
+        av_log(NULL, AV_LOG_ERROR, "pthread_create failed: %s. Try to increase `ulimit -v` or decrease `ulimit -s`.\n", strerror(ret));
         av_thread_message_queue_free(&f->in_thread_queue);
+        return AVERROR(ret);
     }
+
+    return 0;
 }
 
 static int init_input_threads(void)
 {
     int i, ret;
 
-    if (nb_input_files == 1)
-        return 0;
-
     for (i = 0; i < nb_input_files; i++) {
-        InputFile *f = input_files[i];
-
-        if (f->ctx->pb ? !f->ctx->pb->seekable :
-            strcmp(f->ctx->iformat->name, "lavfi"))
-            f->non_blocking = 1;
-        ret = av_thread_message_queue_alloc(&f->in_thread_queue,
-                                            f->thread_queue_size, sizeof(AVPacket));
+        ret = init_input_thread(i);
         if (ret < 0)
             return ret;
-
-        if ((ret = pthread_create(&f->thread, NULL, input_thread, f))) {
-            av_log(NULL, AV_LOG_ERROR, "pthread_create failed: %s. Try to increase `ulimit -v` or decrease `ulimit -s`.\n", strerror(ret));
-            av_thread_message_queue_free(&f->in_thread_queue);
-            return AVERROR(ret);
-        }
     }
     return 0;
 }
@@ -4171,7 +4242,8 @@ static int seek_to_start(InputFile *ifile, AVFormatContext *is)
             ifile->time_base = ist->st->time_base;
         /* the total duration of the stream, max_pts - min_pts is
          * the duration of the stream without the last frame */
-        duration += ist->max_pts - ist->min_pts;
+        if (ist->max_pts > ist->min_pts && ist->max_pts - (uint64_t)ist->min_pts < INT64_MAX - duration)
+            duration += ist->max_pts - ist->min_pts;
         ifile->time_base = duration_max(duration, &ifile->duration, ist->st->time_base,
                                         ifile->time_base);
     }
@@ -4195,7 +4267,7 @@ static int process_input(int file_index)
     AVFormatContext *is;
     InputStream *ist;
     AVPacket pkt;
-    int ret, i, j;
+    int ret, thread_ret, i, j;
     int64_t duration;
     int64_t pkt_dts;
 
@@ -4218,7 +4290,15 @@ static int process_input(int file_index)
                 avcodec_flush_buffers(avctx);
             }
         }
+#if HAVE_THREADS
+        free_input_thread(file_index);
+#endif
         ret = seek_to_start(ifile, is);
+#if HAVE_THREADS
+        thread_ret = init_input_thread(file_index);
+        if (thread_ret < 0)
+            return thread_ret;
+#endif
         if (ret < 0)
             av_log(NULL, AV_LOG_WARNING, "Seek to start failed.\n");
         else
@@ -4278,9 +4358,11 @@ static int process_input(int file_index)
     if (ist->discard)
         goto discard_packet;
 
-    if (exit_on_error && (pkt.flags & AV_PKT_FLAG_CORRUPT)) {
-        av_log(NULL, AV_LOG_FATAL, "%s: corrupt input packet in stream %d\n", is->url, pkt.stream_index);
-        exit_program(1);
+    if (pkt.flags & AV_PKT_FLAG_CORRUPT) {
+        av_log(NULL, exit_on_error ? AV_LOG_FATAL : AV_LOG_WARNING,
+               "%s: corrupt input packet in stream %d\n", is->url, pkt.stream_index);
+        if (exit_on_error)
+            exit_program(1);
     }
 
     if (debug_ts) {
@@ -4715,23 +4797,30 @@ static int transcode(void)
     return ret;
 }
 
-
-static int64_t getutime(void)
+static BenchmarkTimeStamps get_benchmark_time_stamps(void)
 {
+    BenchmarkTimeStamps time_stamps = { av_gettime_relative() };
 #if HAVE_GETRUSAGE
     struct rusage rusage;
 
     getrusage(RUSAGE_SELF, &rusage);
-    return (rusage.ru_utime.tv_sec * 1000000LL) + rusage.ru_utime.tv_usec;
+    time_stamps.user_usec =
+        (rusage.ru_utime.tv_sec * 1000000LL) + rusage.ru_utime.tv_usec;
+    time_stamps.sys_usec =
+        (rusage.ru_stime.tv_sec * 1000000LL) + rusage.ru_stime.tv_usec;
 #elif HAVE_GETPROCESSTIMES
     HANDLE proc;
     FILETIME c, e, k, u;
     proc = GetCurrentProcess();
     GetProcessTimes(proc, &c, &e, &k, &u);
-    return ((int64_t) u.dwHighDateTime << 32 | u.dwLowDateTime) / 10;
+    time_stamps.user_usec =
+        ((int64_t)u.dwHighDateTime << 32 | u.dwLowDateTime) / 10;
+    time_stamps.sys_usec =
+        ((int64_t)k.dwHighDateTime << 32 | k.dwLowDateTime) / 10;
 #else
-    return av_gettime_relative();
+    time_stamps.user_usec = time_stamps.sys_usec = 0;
 #endif
+    return time_stamps;
 }
 
 static int64_t getmaxrss(void)
@@ -4759,7 +4848,7 @@ static void log_callback_null(void *ptr, int level, const char *fmt, va_list vl)
 int main(int argc, char **argv)
 {
     int i, ret;
-    int64_t ti;
+    BenchmarkTimeStamps ti;
 
     init_dynload();
 
@@ -4811,12 +4900,18 @@ int main(int argc, char **argv)
             want_sdp = 0;
     }
 
-    current_time = ti = getutime();
+    current_time = ti = get_benchmark_time_stamps();
     if (transcode() < 0)
         exit_program(1);
-    ti = getutime() - ti;
     if (do_benchmark) {
-        av_log(NULL, AV_LOG_INFO, "bench: utime=%0.3fs\n", ti / 1000000.0);
+        int64_t utime, stime, rtime;
+        current_time = get_benchmark_time_stamps();
+        utime = current_time.user_usec - ti.user_usec;
+        stime = current_time.sys_usec  - ti.sys_usec;
+        rtime = current_time.real_usec - ti.real_usec;
+        av_log(NULL, AV_LOG_INFO,
+               "bench: utime=%0.3fs stime=%0.3fs rtime=%0.3fs\n",
+               utime / 1000000.0, stime / 1000000.0, rtime / 1000000.0);
     }
     av_log(NULL, AV_LOG_DEBUG, "%"PRIu64" frames successfully decoded, %"PRIu64" decoding errors\n",
            decode_error_stat[0], decode_error_stat[1]);

fftools/ffmpeg.h

@@ -484,6 +484,7 @@ typedef struct OutputStream {
     AVRational frame_aspect_ratio;
 
     /* forced key frames */
+    int64_t forced_kf_ref_pts;
     int64_t *forced_kf_pts;
     int forced_kf_count;
     int forced_kf_index;

fftools/ffmpeg_filter.c

@@ -65,6 +65,7 @@ enum AVPixelFormat choose_pixel_fmt(AVStream *st, AVCodecContext *enc_ctx, AVCod
     if (codec && codec->pix_fmts) {
         const enum AVPixelFormat *p = codec->pix_fmts;
         const AVPixFmtDescriptor *desc = av_pix_fmt_desc_get(target);
+        //FIXME: This should check for AV_PIX_FMT_FLAG_ALPHA after PAL8 pixel format without alpha is implemented
         int has_alpha = desc ? desc->nb_components % 2 == 0 : 0;
         enum AVPixelFormat best= AV_PIX_FMT_NONE;
 
@@ -774,7 +775,7 @@ static int configure_input_video_filter(FilterGraph *fg, InputFilter *ifilter,
     sar = ifilter->sample_aspect_ratio;
     if(!sar.den)
         sar = (AVRational){0,1};
-    av_bprint_init(&args, 0, 1);
+    av_bprint_init(&args, 0, AV_BPRINT_SIZE_AUTOMATIC);
     av_bprintf(&args,
              "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:"
              "pixel_aspect=%d/%d:sws_param=flags=%d",

fftools/ffmpeg_opt.c

@@ -1,3 +1,4 @@
+
 /*
  * ffmpeg option parsing
  *
@@ -900,13 +901,14 @@ static void add_input_streams(OptionsContext *o, AVFormatContext *ic)
 
 static void assert_file_overwrite(const char *filename)
 {
+    const char *proto_name = avio_find_protocol_name(filename);
+
     if (file_overwrite && no_file_overwrite) {
         fprintf(stderr, "Error, both -y and -n supplied. Exiting.\n");
         exit_program(1);
     }
 
     if (!file_overwrite) {
-        const char *proto_name = avio_find_protocol_name(filename);
         if (proto_name && !strcmp(proto_name, "file") && avio_check(filename, 0) == 0) {
             if (stdin_interaction && !no_file_overwrite) {
                 fprintf(stderr,"File '%s' already exists. Overwrite ? [y/N] ", filename);
@@ -925,6 +927,19 @@ static void assert_file_overwrite(const char *filename)
             }
         }
     }
+
+    if (proto_name && !strcmp(proto_name, "file")) {
+        for (int i = 0; i < nb_input_files; i++) {
+             InputFile *file = input_files[i];
+             if (file->ctx->iformat->flags & AVFMT_NOFILE)
+                 continue;
+             if (!strcmp(filename, file->ctx->url)) {
+                 av_log(NULL, AV_LOG_FATAL, "Output %s same as Input #%d - exiting\n", filename, i);
+                 av_log(NULL, AV_LOG_WARNING, "FFmpeg cannot edit existing files in-place.\n");
+                 exit_program(1);
+             }
+        }
+    }
 }
 
 static void dump_attachment(AVStream *st, const char *filename)
@@ -1103,9 +1118,22 @@ static int open_input_file(OptionsContext *o, const char *filename)
         }
     }
 
+    if (o->start_time != AV_NOPTS_VALUE && o->start_time_eof != AV_NOPTS_VALUE) {
+        av_log(NULL, AV_LOG_WARNING, "Cannot use -ss and -sseof both, using -ss for %s\n", filename);
+        o->start_time_eof = AV_NOPTS_VALUE;
+    }
+
     if (o->start_time_eof != AV_NOPTS_VALUE) {
-        if (ic->duration>0) {
+        if (o->start_time_eof >= 0) {
+            av_log(NULL, AV_LOG_ERROR, "-sseof value must be negative; aborting\n");
+            exit_program(1);
+        }
+        if (ic->duration > 0) {
             o->start_time = o->start_time_eof + ic->duration;
+            if (o->start_time < 0) {
+                av_log(NULL, AV_LOG_WARNING, "-sseof value seeks to before start of file %s; ignored\n", filename);
+                o->start_time = AV_NOPTS_VALUE;
+            }
         } else
             av_log(NULL, AV_LOG_WARNING, "Cannot use -sseof, duration of %s not known\n", filename);
     }
@@ -1122,8 +1150,10 @@ static int open_input_file(OptionsContext *o, const char *filename)
             int dts_heuristic = 0;
             for (i=0; i<ic->nb_streams; i++) {
                 const AVCodecParameters *par = ic->streams[i]->codecpar;
-                if (par->video_delay)
+                if (par->video_delay) {
                     dts_heuristic = 1;
+                    break;
+                }
             }
             if (dts_heuristic) {
                 seek_timestamp -= 3*AV_TIME_BASE / 23;
@@ -1324,6 +1354,7 @@ static OutputStream *new_output_stream(OptionsContext *o, AVFormatContext *oc, e
     ost->file_index = nb_output_files - 1;
     ost->index      = idx;
     ost->st         = st;
+    ost->forced_kf_ref_pts = AV_NOPTS_VALUE;
     st->codecpar->codec_type = type;
 
     ret = choose_encoder(o, oc, ost);
@@ -2302,12 +2333,14 @@ loop_end:
                    o->attachments[i]);
             exit_program(1);
         }
-        if (!(attachment = av_malloc(len))) {
-            av_log(NULL, AV_LOG_FATAL, "Attachment %s too large to fit into memory.\n",
+        if (len > INT_MAX - AV_INPUT_BUFFER_PADDING_SIZE ||
+            !(attachment = av_malloc(len + AV_INPUT_BUFFER_PADDING_SIZE))) {
+            av_log(NULL, AV_LOG_FATAL, "Attachment %s too large.\n",
                    o->attachments[i]);
             exit_program(1);
         }
         avio_read(pb, attachment, len);
+        memset(attachment + len, 0, AV_INPUT_BUFFER_PADDING_SIZE);
 
         ost = new_attachment_stream(o, oc, -1);
         ost->stream_copy               = 0;
@@ -2699,13 +2732,14 @@ static int opt_target(void *optctx, const char *opt, const char *arg)
     } else {
         /* Try to determine PAL/NTSC by peeking in the input files */
         if (nb_input_files) {
-            int i, j, fr;
+            int i, j;
             for (j = 0; j < nb_input_files; j++) {
                 for (i = 0; i < input_files[j]->nb_streams; i++) {
                     AVStream *st = input_files[j]->ctx->streams[i];
+                    int64_t fr;
                     if (st->codecpar->codec_type != AVMEDIA_TYPE_VIDEO)
                         continue;
-                    fr = st->time_base.den * 1000 / st->time_base.num;
+                    fr = st->time_base.den * 1000LL / st->time_base.num;
                     if (fr == 25000) {
                         norm = PAL;
                         break;
@@ -3157,7 +3191,9 @@ void show_help_default(const char *opt, const char *arg)
 #if CONFIG_SWSCALE
         show_help_children(sws_get_class(), flags);
 #endif
+#if CONFIG_SWRESAMPLE
         show_help_children(swr_get_class(), AV_OPT_FLAG_AUDIO_PARAM);
+#endif
         show_help_children(avfilter_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM);
         show_help_children(av_bsf_get_class(), AV_OPT_FLAG_VIDEO_PARAM | AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_BSF_PARAM);
     }
@@ -3196,6 +3232,7 @@ static int open_files(OptionGroupList *l, const char *inout,
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error parsing options for %s file "
                    "%s.\n", inout, g->arg);
+            uninit_options(&o);
             return ret;
         }
 
@@ -3337,7 +3374,7 @@ const OptionDef options[] = {
                         OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time) },
         "set the start time offset", "time_off" },
     { "sseof",          HAS_ARG | OPT_TIME | OPT_OFFSET |
-                        OPT_INPUT | OPT_OUTPUT,                      { .off = OFFSET(start_time_eof) },
+                        OPT_INPUT,                                   { .off = OFFSET(start_time_eof) },
         "set the start time offset relative to EOF", "time_off" },
     { "seek_timestamp", HAS_ARG | OPT_INT | OPT_OFFSET |
                         OPT_INPUT,                                   { .off = OFFSET(seek_timestamp) },

fftools/ffmpeg_qsv.c

@@ -93,7 +93,7 @@ int qsv_init(AVCodecContext *s)
     frames_ctx->height            = FFALIGN(s->coded_height, 32);
     frames_ctx->format            = AV_PIX_FMT_QSV;
     frames_ctx->sw_format         = s->sw_pix_fmt;
-    frames_ctx->initial_pool_size = 64;
+    frames_ctx->initial_pool_size = 64 + s->extra_hw_frames;
     frames_hwctx->frame_type      = MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET;
 
     ret = av_hwframe_ctx_init(ist->hw_frames_ctx);

fftools/ffplay.c

@@ -314,11 +314,14 @@ static int default_width  = 640;
 static int default_height = 480;
 static int screen_width  = 0;
 static int screen_height = 0;
+static int screen_left = SDL_WINDOWPOS_CENTERED;
+static int screen_top = SDL_WINDOWPOS_CENTERED;
 static int audio_disable;
 static int video_disable;
 static int subtitle_disable;
 static const char* wanted_stream_spec[AVMEDIA_TYPE_NB] = {0};
 static int seek_by_bytes = -1;
+static float seek_interval = 10;
 static int display_disable;
 static int borderless;
 static int startup_volume = 100;
@@ -834,10 +837,11 @@ static int realloc_texture(SDL_Texture **texture, Uint32 new_format, int new_wid
 {
     Uint32 format;
     int access, w, h;
-    if (SDL_QueryTexture(*texture, &format, &access, &w, &h) < 0 || new_width != w || new_height != h || new_format != format) {
+    if (!*texture || SDL_QueryTexture(*texture, &format, &access, &w, &h) < 0 || new_width != w || new_height != h || new_format != format) {
         void *pixels;
         int pitch;
-        SDL_DestroyTexture(*texture);
+        if (*texture)
+            SDL_DestroyTexture(*texture);
         if (!(*texture = SDL_CreateTexture(renderer, new_format, SDL_TEXTUREACCESS_STREAMING, new_width, new_height)))
             return -1;
         if (SDL_SetTextureBlendMode(*texture, blendmode) < 0)
@@ -953,6 +957,22 @@ static int upload_texture(SDL_Texture **tex, AVFrame *frame, struct SwsContext *
     return ret;
 }
 
+static void set_sdl_yuv_conversion_mode(AVFrame *frame)
+{
+#if SDL_VERSION_ATLEAST(2,0,8)
+    SDL_YUV_CONVERSION_MODE mode = SDL_YUV_CONVERSION_AUTOMATIC;
+    if (frame && (frame->format == AV_PIX_FMT_YUV420P || frame->format == AV_PIX_FMT_YUYV422 || frame->format == AV_PIX_FMT_UYVY422)) {
+        if (frame->color_range == AVCOL_RANGE_JPEG)
+            mode = SDL_YUV_CONVERSION_JPEG;
+        else if (frame->colorspace == AVCOL_SPC_BT709)
+            mode = SDL_YUV_CONVERSION_BT709;
+        else if (frame->colorspace == AVCOL_SPC_BT470BG || frame->colorspace == AVCOL_SPC_SMPTE170M || frame->colorspace == AVCOL_SPC_SMPTE240M)
+            mode = SDL_YUV_CONVERSION_BT601;
+    }
+    SDL_SetYUVConversionMode(mode);
+#endif
+}
+
 static void video_image_display(VideoState *is)
 {
     Frame *vp;
@@ -1014,7 +1034,9 @@ static void video_image_display(VideoState *is)
         vp->flip_v = vp->frame->linesize[0] < 0;
     }
 
+    set_sdl_yuv_conversion_mode(vp->frame);
     SDL_RenderCopyEx(renderer, is->vid_texture, NULL, &rect, 0, NULL, vp->flip_v ? SDL_FLIP_VERTICAL : 0);
+    set_sdl_yuv_conversion_mode(NULL);
     if (sp) {
 #if USE_ONEPASS_SUBTITLE_RENDER
         SDL_RenderCopy(renderer, is->sub_texture, NULL, &rect);
@@ -1326,7 +1348,7 @@ static int video_open(VideoState *is)
     SDL_SetWindowTitle(window, window_title);
 
     SDL_SetWindowSize(window, w, h);
-    SDL_SetWindowPosition(window, SDL_WINDOWPOS_CENTERED, SDL_WINDOWPOS_CENTERED);
+    SDL_SetWindowPosition(window, screen_left, screen_top);
     if (is_full_screen)
         SDL_SetWindowFullscreen(window, SDL_WINDOW_FULLSCREEN_DESKTOP);
     SDL_ShowWindow(window);
@@ -2192,6 +2214,8 @@ static int video_thread(void *arg)
             ret = queue_picture(is, frame, pts, duration, frame->pkt_pos, is->viddec.pkt_serial);
             av_frame_unref(frame);
 #if CONFIG_AVFILTER
+            if (is->videoq.serial != is->viddec.pkt_serial)
+                break;
         }
 #endif
 
@@ -2577,7 +2601,7 @@ static int stream_component_open(VideoState *is, int stream_index)
         if (forced_codec_name) av_log(NULL, AV_LOG_WARNING,
                                       "No codec could be found with name '%s'\n", forced_codec_name);
         else                   av_log(NULL, AV_LOG_WARNING,
-                                      "No codec could be found with id %d\n", avctx->codec_id);
+                                      "No decoder could be found for codec %s\n", avcodec_get_name(avctx->codec_id));
         ret = AVERROR(EINVAL);
         goto fail;
     }
@@ -3250,15 +3274,14 @@ static void event_loop(VideoState *cur_stream)
         refresh_loop_wait_event(cur_stream, &event);
         switch (event.type) {
         case SDL_KEYDOWN:
-            if (exit_on_keydown) {
+            if (exit_on_keydown || event.key.keysym.sym == SDLK_ESCAPE || event.key.keysym.sym == SDLK_q) {
                 do_exit(cur_stream);
                 break;
             }
+            // If we don't yet have a window, skip all key events, because read_thread might still be initializing...
+            if (!cur_stream->width)
+                continue;
             switch (event.key.keysym.sym) {
-            case SDLK_ESCAPE:
-            case SDLK_q:
-                do_exit(cur_stream);
-                break;
             case SDLK_f:
                 toggle_full_screen(cur_stream);
                 cur_stream->force_refresh = 1;
@@ -3323,10 +3346,10 @@ static void event_loop(VideoState *cur_stream)
                 seek_chapter(cur_stream, -1);
                 break;
             case SDLK_LEFT:
-                incr = -10.0;
+                incr = seek_interval ? -seek_interval : -10.0;
                 goto do_seek;
             case SDLK_RIGHT:
-                incr = 10.0;
+                incr = seek_interval ? seek_interval : 10.0;
                 goto do_seek;
             case SDLK_UP:
                 incr = 60.0;
@@ -3562,6 +3585,7 @@ static const OptionDef options[] = {
     { "ss", HAS_ARG, { .func_arg = opt_seek }, "seek to a given position in seconds", "pos" },
     { "t", HAS_ARG, { .func_arg = opt_duration }, "play  \"duration\" seconds of audio/video", "duration" },
     { "bytes", OPT_INT | HAS_ARG, { &seek_by_bytes }, "seek by bytes 0=off 1=on -1=auto", "val" },
+    { "seek_interval", OPT_FLOAT | HAS_ARG, { &seek_interval }, "set seek interval for left/right keys, in seconds", "seconds" },
     { "nodisp", OPT_BOOL, { &display_disable }, "disable graphical display" },
     { "noborder", OPT_BOOL, { &borderless }, "borderless window" },
     { "volume", OPT_INT | HAS_ARG, { &startup_volume}, "set startup volume 0=min 100=max", "volume" },
@@ -3580,6 +3604,8 @@ static const OptionDef options[] = {
     { "framedrop", OPT_BOOL | OPT_EXPERT, { &framedrop }, "drop frames when cpu is too slow", "" },
     { "infbuf", OPT_BOOL | OPT_EXPERT, { &infinite_buffer }, "don't limit the input buffer size (useful with realtime streams)", "" },
     { "window_title", OPT_STRING | HAS_ARG, { &window_title }, "set window title", "window title" },
+    { "left", OPT_INT | HAS_ARG | OPT_EXPERT, { &screen_left }, "set the x position for the left of the window", "x pos" },
+    { "top", OPT_INT | HAS_ARG | OPT_EXPERT, { &screen_top }, "set the y position for the top of the window", "y pos" },
 #if CONFIG_AVFILTER
     { "vf", OPT_EXPERT | HAS_ARG, { .func_arg = opt_add_vfilter }, "set video filters", "filter_graph" },
     { "af", OPT_STRING | HAS_ARG, { &afilters }, "set audio filters", "filter_graph" },
@@ -3632,7 +3658,7 @@ void show_help_default(const char *opt, const char *arg)
            "c                   cycle program\n"
            "w                   cycle video filters or show modes\n"
            "s                   activate frame-step mode\n"
-           "left/right          seek backward/forward 10 seconds\n"
+           "left/right          seek backward/forward 10 seconds or to custom interval if -seek_interval is set\n"
            "down/up             seek backward/forward 1 minute\n"
            "page down/page up   seek backward/forward 10 minutes\n"
            "right mouse click   seek to percentage in file corresponding to fraction of width\n"

fftools/ffprobe.c

@@ -2371,11 +2371,11 @@ static int read_interval_packets(WriterContext *w, InputFile *ifile,
         goto end;
     }
     while (!av_read_frame(fmt_ctx, &pkt)) {
-        if (ifile->nb_streams > nb_streams) {
+        if (fmt_ctx->nb_streams > nb_streams) {
             REALLOCZ_ARRAY_STREAM(nb_streams_frames,  nb_streams, fmt_ctx->nb_streams);
             REALLOCZ_ARRAY_STREAM(nb_streams_packets, nb_streams, fmt_ctx->nb_streams);
             REALLOCZ_ARRAY_STREAM(selected_streams,   nb_streams, fmt_ctx->nb_streams);
-            nb_streams = ifile->nb_streams;
+            nb_streams = fmt_ctx->nb_streams;
         }
         if (selected_streams[pkt.stream_index]) {
             AVRational tb = ifile->streams[pkt.stream_index].st->time_base;
@@ -2521,7 +2521,7 @@ static int show_stream(WriterContext *w, AVFormatContext *fmt_ctx, int stream_id
 #endif
         print_int("has_b_frames", par->video_delay);
         sar = av_guess_sample_aspect_ratio(fmt_ctx, stream, NULL);
-        if (sar.den) {
+        if (sar.num) {
             print_q("sample_aspect_ratio", sar, ':');
             av_reduce(&dar.num, &dar.den,
                       par->width  * sar.num,

libavcodec/012v.c

@@ -131,8 +131,8 @@ static int zero12v_decode_frame(AVCodecContext *avctx, void *data,
             u = x/2 + (uint16_t *)(pic->data[1] + line * pic->linesize[1]);
             v = x/2 + (uint16_t *)(pic->data[2] + line * pic->linesize[2]);
             memcpy(y, y_temp, sizeof(*y) * (width - x));
-            memcpy(u, u_temp, sizeof(*u) * (width - x + 1) / 2);
-            memcpy(v, v_temp, sizeof(*v) * (width - x + 1) / 2);
+            memcpy(u, u_temp, sizeof(*u) * ((width - x + 1) / 2));
+            memcpy(v, v_temp, sizeof(*v) * ((width - x + 1) / 2));
         }
 
         line_end += stride;

libavcodec/4xm.c

@@ -158,7 +158,7 @@ typedef struct FourXContext {
 #define FIX_1_847759065 121095
 #define FIX_2_613125930 171254
 
-#define MULTIPLY(var, const) (((var) * (const)) >> 16)
+#define MULTIPLY(var, const) ((int)((var) * (unsigned)(const)) >> 16)
 
 static void idct(int16_t block[64])
 {
@@ -351,6 +351,8 @@ static int decode_p_block(FourXContext *f, uint16_t *dst, const uint16_t *src,
     index = size2index[log2h][log2w];
     av_assert0(index >= 0);
 
+    if (get_bits_left(&f->gb) < 1)
+        return AVERROR_INVALIDDATA;
     h     = 1 << log2h;
     code  = get_vlc2(&f->gb, block_type_vlc[1 - (f->version > 1)][index].table,
                      BLOCK_TYPE_VLC_BITS, 1);
@@ -496,9 +498,9 @@ static int decode_i_block(FourXContext *f, int16_t *block)
 {
     int code, i, j, level, val;
 
-    if (get_bits_left(&f->gb) < 2){
-        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->gb));
-        return -1;
+    if (get_bits_left(&f->pre_gb) < 2) {
+        av_log(f->avctx, AV_LOG_ERROR, "%d bits left before decode_i_block()\n", get_bits_left(&f->pre_gb));
+        return AVERROR_INVALIDDATA;
     }
 
     /* DC coef */
@@ -523,6 +525,10 @@ static int decode_i_block(FourXContext *f, int16_t *block)
             break;
         if (code == 0xf0) {
             i += 16;
+            if (i >= 64) {
+                av_log(f->avctx, AV_LOG_ERROR, "run %d overflow\n", i);
+                return 0;
+            }
         } else {
             if (code & 0xf) {
                 level = get_xbits(&f->gb, code & 0xf);
@@ -732,7 +738,7 @@ static int decode_i2_frame(FourXContext *f, const uint8_t *buf, int length)
         for (x = 0; x < width; x += 16) {
             unsigned int color[4] = { 0 }, bits;
             if (buf_end - buf < 8)
-                return -1;
+                return AVERROR_INVALIDDATA;
             // warning following is purely guessed ...
             color[0] = bytestream2_get_le16u(&g3);
             color[1] = bytestream2_get_le16u(&g3);

libavcodec/8bps.c

@@ -70,6 +70,9 @@ static int decode_frame(AVCodecContext *avctx, void *data,
     unsigned char *planemap = c->planemap;
     int ret;
 
+    if (buf_size < planes * height *2)
+        return AVERROR_INVALIDDATA;
+
     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
         return ret;
 

libavcodec/Makefile

@@ -44,6 +44,7 @@ OBJS = ac3_parser.o                                                     \
        options.o                                                        \
        mjpegenc_huffman.o                                               \
        parser.o                                                         \
+       parsers.o                                                        \
        profiles.o                                                       \
        qsv_api.o                                                        \
        raw.o                                                            \
@@ -62,9 +63,12 @@ OBJS-$(CONFIG_BLOCKDSP)                += blockdsp.o
 OBJS-$(CONFIG_BSWAPDSP)                += bswapdsp.o
 OBJS-$(CONFIG_CABAC)                   += cabac.o
 OBJS-$(CONFIG_CBS)                     += cbs.o
+OBJS-$(CONFIG_CBS_AV1)                 += cbs_av1.o
 OBJS-$(CONFIG_CBS_H264)                += cbs_h2645.o h2645_parse.o
 OBJS-$(CONFIG_CBS_H265)                += cbs_h2645.o h2645_parse.o
+OBJS-$(CONFIG_CBS_JPEG)                += cbs_jpeg.o
 OBJS-$(CONFIG_CBS_MPEG2)               += cbs_mpeg2.o
+OBJS-$(CONFIG_CBS_VP9)                 += cbs_vp9.o
 OBJS-$(CONFIG_CRYSTALHD)               += crystalhd.o
 OBJS-$(CONFIG_DCT)                     += dct.o dct32_fixed.o dct32_float.o
 OBJS-$(CONFIG_ERROR_RESILIENCE)        += error_resilience.o
@@ -209,6 +213,7 @@ OBJS-$(CONFIG_ATRAC3P_DECODER)         += atrac3plusdec.o atrac3plus.o \
                                           atrac3plusdsp.o atrac.o
 OBJS-$(CONFIG_ATRAC3PAL_DECODER)       += atrac3plusdec.o atrac3plus.o \
                                           atrac3plusdsp.o atrac.o
+OBJS-$(CONFIG_ATRAC9_DECODER)          += atrac9dec.o
 OBJS-$(CONFIG_AURA_DECODER)            += cyuv.o
 OBJS-$(CONFIG_AURA2_DECODER)           += aura.o
 OBJS-$(CONFIG_AVRN_DECODER)            += avrndec.o mjpegdec.o
@@ -351,7 +356,7 @@ OBJS-$(CONFIG_H264_OMX_ENCODER)        += omx.o
 OBJS-$(CONFIG_H264_QSV_DECODER)        += qsvdec_h2645.o
 OBJS-$(CONFIG_H264_QSV_ENCODER)        += qsvenc_h264.o
 OBJS-$(CONFIG_H264_RKMPP_DECODER)      += rkmppdec.o
-OBJS-$(CONFIG_H264_VAAPI_ENCODER)      += vaapi_encode_h264.o
+OBJS-$(CONFIG_H264_VAAPI_ENCODER)      += vaapi_encode_h264.o h264_levels.o
 OBJS-$(CONFIG_H264_VIDEOTOOLBOX_ENCODER) += videotoolboxenc.o
 OBJS-$(CONFIG_H264_V4L2M2M_DECODER)    += v4l2_m2m_dec.o
 OBJS-$(CONFIG_H264_V4L2M2M_ENCODER)    += v4l2_m2m_enc.o
@@ -369,7 +374,7 @@ OBJS-$(CONFIG_HEVC_QSV_DECODER)        += qsvdec_h2645.o
 OBJS-$(CONFIG_HEVC_QSV_ENCODER)        += qsvenc_hevc.o hevc_ps_enc.o       \
                                           hevc_data.o
 OBJS-$(CONFIG_HEVC_RKMPP_DECODER)      += rkmppdec.o
-OBJS-$(CONFIG_HEVC_VAAPI_ENCODER)      += vaapi_encode_h265.o
+OBJS-$(CONFIG_HEVC_VAAPI_ENCODER)      += vaapi_encode_h265.o h265_profile_level.o
 OBJS-$(CONFIG_HEVC_V4L2M2M_DECODER)    += v4l2_m2m_dec.o
 OBJS-$(CONFIG_HEVC_V4L2M2M_ENCODER)    += v4l2_m2m_enc.o
 OBJS-$(CONFIG_HNM4_VIDEO_DECODER)      += hnm4video.o
@@ -381,7 +386,9 @@ OBJS-$(CONFIG_HUFFYUV_ENCODER)         += huffyuv.o huffyuvenc.o
 OBJS-$(CONFIG_IDCIN_DECODER)           += idcinvideo.o
 OBJS-$(CONFIG_IDF_DECODER)             += bintext.o cga_data.o
 OBJS-$(CONFIG_IFF_ILBM_DECODER)        += iff.o
+OBJS-$(CONFIG_ILBC_DECODER)            += ilbcdec.o
 OBJS-$(CONFIG_IMC_DECODER)             += imc.o
+OBJS-$(CONFIG_IMM4_DECODER)            += imm4.o
 OBJS-$(CONFIG_INDEO2_DECODER)          += indeo2.o
 OBJS-$(CONFIG_INDEO3_DECODER)          += indeo3.o
 OBJS-$(CONFIG_INDEO4_DECODER)          += indeo4.o ivi.o
@@ -478,6 +485,7 @@ OBJS-$(CONFIG_MSZH_DECODER)            += lcldec.o
 OBJS-$(CONFIG_MTS2_DECODER)            += mss4.o
 OBJS-$(CONFIG_MVC1_DECODER)            += mvcdec.o
 OBJS-$(CONFIG_MVC2_DECODER)            += mvcdec.o
+OBJS-$(CONFIG_MWSC_DECODER)            += mwsc.o
 OBJS-$(CONFIG_MXPEG_DECODER)           += mxpegdec.o
 OBJS-$(CONFIG_NELLYMOSER_DECODER)      += nellymoserdec.o nellymoser.o
 OBJS-$(CONFIG_NELLYMOSER_ENCODER)      += nellymoserenc.o nellymoser.o
@@ -508,10 +516,10 @@ OBJS-$(CONFIG_PNG_ENCODER)             += png.o pngenc.o
 OBJS-$(CONFIG_PPM_DECODER)             += pnmdec.o pnm.o
 OBJS-$(CONFIG_PPM_ENCODER)             += pnmenc.o
 OBJS-$(CONFIG_PRORES_DECODER)          += proresdec2.o proresdsp.o proresdata.o
-OBJS-$(CONFIG_PRORES_LGPL_DECODER)     += proresdec_lgpl.o proresdsp.o proresdata.o
-OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o
-OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o
+OBJS-$(CONFIG_PRORES_ENCODER)          += proresenc_anatoliy.o proresdata.o
+OBJS-$(CONFIG_PRORES_AW_ENCODER)       += proresenc_anatoliy.o proresdata.o
 OBJS-$(CONFIG_PRORES_KS_ENCODER)       += proresenc_kostya.o proresdata.o
+OBJS-$(CONFIG_PROSUMER_DECODER)        += prosumer.o
 OBJS-$(CONFIG_PSD_DECODER)             += psd.o
 OBJS-$(CONFIG_PTX_DECODER)             += ptx.o
 OBJS-$(CONFIG_QCELP_DECODER)           += qcelpdec.o                     \
@@ -531,6 +539,7 @@ OBJS-$(CONFIG_RA_144_DECODER)          += ra144dec.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_144_ENCODER)          += ra144enc.o ra144.o celp_filters.o
 OBJS-$(CONFIG_RA_288_DECODER)          += ra288.o celp_filters.o
 OBJS-$(CONFIG_RALF_DECODER)            += ralf.o
+OBJS-$(CONFIG_RASC_DECODER)            += rasc.o
 OBJS-$(CONFIG_RAWVIDEO_DECODER)        += rawdec.o
 OBJS-$(CONFIG_RAWVIDEO_ENCODER)        += rawenc.o
 OBJS-$(CONFIG_REALTEXT_DECODER)        += realtextdec.o ass.o
@@ -574,7 +583,7 @@ OBJS-$(CONFIG_SOL_DPCM_DECODER)        += dpcm.o
 OBJS-$(CONFIG_SONIC_DECODER)           += sonic.o
 OBJS-$(CONFIG_SONIC_ENCODER)           += sonic.o
 OBJS-$(CONFIG_SONIC_LS_ENCODER)        += sonic.o
-OBJS-$(CONFIG_SPEEDHQ_DECODER)         += speedhq.o simple_idct.o
+OBJS-$(CONFIG_SPEEDHQ_DECODER)         += speedhq.o mpeg12.o mpeg12data.o simple_idct.o
 OBJS-$(CONFIG_SP5X_DECODER)            += sp5xdec.o
 OBJS-$(CONFIG_SRGC_DECODER)            += mscc.o
 OBJS-$(CONFIG_SRT_DECODER)             += srtdec.o ass.o htmlsubtitles.o
@@ -672,6 +681,7 @@ OBJS-$(CONFIG_VP9_V4L2M2M_DECODER)     += v4l2_m2m_dec.o
 OBJS-$(CONFIG_VQA_DECODER)             += vqavideo.o
 OBJS-$(CONFIG_WAVPACK_DECODER)         += wavpack.o
 OBJS-$(CONFIG_WAVPACK_ENCODER)         += wavpackenc.o
+OBJS-$(CONFIG_WCMV_DECODER)            += wcmv.o
 OBJS-$(CONFIG_WEBP_DECODER)            += webp.o
 OBJS-$(CONFIG_WEBVTT_DECODER)          += webvttdec.o ass.o
 OBJS-$(CONFIG_WEBVTT_ENCODER)          += webvttenc.o ass_split.o
@@ -784,6 +794,8 @@ OBJS-$(CONFIG_PCM_U32BE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32BE_ENCODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32LE_DECODER)          += pcm.o
 OBJS-$(CONFIG_PCM_U32LE_ENCODER)          += pcm.o
+OBJS-$(CONFIG_PCM_VIDC_DECODER)           += pcm.o
+OBJS-$(CONFIG_PCM_VIDC_ENCODER)           += pcm.o
 OBJS-$(CONFIG_PCM_ZORK_DECODER)           += pcm.o
 
 OBJS-$(CONFIG_ADPCM_4XM_DECODER)          += adpcm.o adpcm_data.o
@@ -944,6 +956,7 @@ OBJS-$(CONFIG_LIBAOM_AV1_ENCODER)         += libaomenc.o
 OBJS-$(CONFIG_LIBCELT_DECODER)            += libcelt_dec.o
 OBJS-$(CONFIG_LIBCODEC2_DECODER)          += libcodec2.o codec2utils.o
 OBJS-$(CONFIG_LIBCODEC2_ENCODER)          += libcodec2.o codec2utils.o
+OBJS-$(CONFIG_LIBDAVS2_DECODER)           += libdavs2.o
 OBJS-$(CONFIG_LIBFDK_AAC_DECODER)         += libfdk-aacdec.o
 OBJS-$(CONFIG_LIBFDK_AAC_ENCODER)         += libfdk-aacenc.o
 OBJS-$(CONFIG_LIBGSM_DECODER)             += libgsmdec.o
@@ -985,7 +998,8 @@ OBJS-$(CONFIG_LIBX262_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX264_ENCODER)            += libx264.o
 OBJS-$(CONFIG_LIBX265_ENCODER)            += libx265.o
 OBJS-$(CONFIG_LIBXAVS_ENCODER)            += libxavs.o
-OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o libxvid_rc.o
+OBJS-$(CONFIG_LIBXAVS2_ENCODER)           += libxavs2.o
+OBJS-$(CONFIG_LIBXVID_ENCODER)            += libxvid.o
 OBJS-$(CONFIG_LIBZVBI_TELETEXT_DECODER)   += libzvbi-teletextdec.o ass.o
 
 # parsers
@@ -994,6 +1008,8 @@ OBJS-$(CONFIG_AAC_PARSER)              += aac_parser.o aac_ac3_parser.o \
                                           mpeg4audio.o
 OBJS-$(CONFIG_AC3_PARSER)              += ac3tab.o aac_ac3_parser.o
 OBJS-$(CONFIG_ADX_PARSER)              += adx_parser.o adx.o
+OBJS-$(CONFIG_AV1_PARSER)              += av1_parser.o av1_parse.o
+OBJS-$(CONFIG_AVS2_PARSER)             += avs2_parser.o
 OBJS-$(CONFIG_BMP_PARSER)              += bmp_parser.o
 OBJS-$(CONFIG_CAVSVIDEO_PARSER)        += cavs_parser.o
 OBJS-$(CONFIG_COOK_PARSER)             += cook_parser.o
@@ -1040,14 +1056,15 @@ OBJS-$(CONFIG_XMA_PARSER)              += xma_parser.o
 
 # bitstream filters
 OBJS-$(CONFIG_AAC_ADTSTOASC_BSF)          += aac_adtstoasc_bsf.o mpeg4audio.o
+OBJS-$(CONFIG_AV1_METADATA_BSF)           += av1_metadata_bsf.o
 OBJS-$(CONFIG_CHOMP_BSF)                  += chomp_bsf.o
 OBJS-$(CONFIG_DUMP_EXTRADATA_BSF)         += dump_extradata_bsf.o
 OBJS-$(CONFIG_DCA_CORE_BSF)               += dca_core_bsf.o
 OBJS-$(CONFIG_EAC3_CORE_BSF)              += eac3_core_bsf.o
 OBJS-$(CONFIG_EXTRACT_EXTRADATA_BSF)      += extract_extradata_bsf.o    \
-                                             h2645_parse.o
+                                             av1_parse.o h2645_parse.o
 OBJS-$(CONFIG_FILTER_UNITS_BSF)           += filter_units_bsf.o
-OBJS-$(CONFIG_H264_METADATA_BSF)          += h264_metadata_bsf.o
+OBJS-$(CONFIG_H264_METADATA_BSF)          += h264_metadata_bsf.o h264_levels.o
 OBJS-$(CONFIG_H264_MP4TOANNEXB_BSF)       += h264_mp4toannexb_bsf.o
 OBJS-$(CONFIG_H264_REDUNDANT_PPS_BSF)     += h264_redundant_pps_bsf.o
 OBJS-$(CONFIG_HAPQA_EXTRACT_BSF)          += hapqa_extract_bsf.o hap.o
@@ -1066,6 +1083,7 @@ OBJS-$(CONFIG_NULL_BSF)                   += null_bsf.o
 OBJS-$(CONFIG_REMOVE_EXTRADATA_BSF)       += remove_extradata_bsf.o
 OBJS-$(CONFIG_TEXT2MOVSUB_BSF)            += movsub_bsf.o
 OBJS-$(CONFIG_TRACE_HEADERS_BSF)          += trace_headers_bsf.o
+OBJS-$(CONFIG_VP9_METADATA_BSF)           += vp9_metadata_bsf.o
 OBJS-$(CONFIG_VP9_RAW_REORDER_BSF)        += vp9_raw_reorder_bsf.o
 OBJS-$(CONFIG_VP9_SUPERFRAME_BSF)         += vp9_superframe_bsf.o
 OBJS-$(CONFIG_VP9_SUPERFRAME_SPLIT_BSF)   += vp9_superframe_split_bsf.o
@@ -1127,6 +1145,7 @@ TESTPROGS-$(CONFIG_IDCTDSP)               += dct
 TESTPROGS-$(CONFIG_IIRFILTER)             += iirfilter
 TESTPROGS-$(HAVE_MMX)                     += motion
 TESTPROGS-$(CONFIG_MPEGVIDEO)             += mpeg12framerate
+TESTPROGS-$(CONFIG_H264_METADATA_BSF)     += h264_levels
 TESTPROGS-$(CONFIG_RANGECODER)            += rangecoder
 TESTPROGS-$(CONFIG_SNOW_ENCODER)          += snowenc
 

libavcodec/aaccoder.c

@@ -843,25 +843,25 @@ static void search_for_ms(AACEncContext *s, ChannelElement *cpe)
                                                     sce0->ics.swb_sizes[g],
                                                     sce0->sf_idx[w*16+g],
                                                     sce0->band_type[w*16+g],
-                                                    lambda / band0->threshold, INFINITY, &b1, NULL, 0);
+                                                    lambda / (band0->threshold + FLT_MIN), INFINITY, &b1, NULL, 0);
                         dist1 += quantize_band_cost(s, &sce1->coeffs[start + (w+w2)*128],
                                                     R34,
                                                     sce1->ics.swb_sizes[g],
                                                     sce1->sf_idx[w*16+g],
                                                     sce1->band_type[w*16+g],
-                                                    lambda / band1->threshold, INFINITY, &b2, NULL, 0);
+                                                    lambda / (band1->threshold + FLT_MIN), INFINITY, &b2, NULL, 0);
                         dist2 += quantize_band_cost(s, M,
                                                     M34,
                                                     sce0->ics.swb_sizes[g],
                                                     mididx,
                                                     midcb,
-                                                    lambda / minthr, INFINITY, &b3, NULL, 0);
+                                                    lambda / (minthr + FLT_MIN), INFINITY, &b3, NULL, 0);
                         dist2 += quantize_band_cost(s, S,
                                                     S34,
                                                     sce1->ics.swb_sizes[g],
                                                     sididx,
                                                     sidcb,
-                                                    mslambda / (minthr * bmax), INFINITY, &b4, NULL, 0);
+                                                    mslambda / (minthr * bmax + FLT_MIN), INFINITY, &b4, NULL, 0);
                         B0 += b1+b2;
                         B1 += b3+b4;
                         dist1 -= b1+b2;

libavcodec/aacdec.c

@@ -411,6 +411,8 @@ static int read_stream_mux_config(struct LATMContext *latmctx,
             } else {
                 int esc;
                 do {
+                    if (get_bits_left(gb) < 9)
+                        return AVERROR_INVALIDDATA;
                     esc = get_bits(gb, 1);
                     skip_bits(gb, 8);
                 } while (esc);
@@ -561,7 +563,7 @@ AVCodec ff_aac_decoder = {
         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },
     .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
-    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
     .channel_layouts = aac_channel_layout,
     .flush = flush,
     .priv_class      = &aac_decoder_class,
@@ -586,7 +588,7 @@ AVCodec ff_aac_latm_decoder = {
         AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE
     },
     .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
-    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE,
+    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
     .channel_layouts = aac_channel_layout,
     .flush = flush,
     .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),

libavcodec/aacdec_fixed.c

@@ -155,9 +155,9 @@ static void vector_pow43(int *coefs, int len)
     for (i=0; i<len; i++) {
         coef = coefs[i];
         if (coef < 0)
-            coef = -(int)ff_cbrt_tab_fixed[-coef];
+            coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
         else
-            coef = (int)ff_cbrt_tab_fixed[coef];
+            coef =  (int)ff_cbrt_tab_fixed[  coef  & 8191];
         coefs[i] = coef;
     }
 }
@@ -195,12 +195,12 @@ static void subband_scale(int *dst, int *src, int scale, int offset, int len)
 
 static void noise_scale(int *coefs, int scale, int band_energy, int len)
 {
-    int ssign = scale < 0 ? -1 : 1;
-    int s = FFABS(scale);
+    int s = -scale;
     unsigned int round;
     int i, out, c = exp2tab[s & 3];
     int nlz = 0;
 
+    av_assert0(s >= 0);
     while (band_energy > 0x7fff) {
         band_energy >>= 1;
         nlz++;
@@ -216,15 +216,20 @@ static void noise_scale(int *coefs, int scale, int band_energy, int len)
         round = s ? 1 << (s-1) : 0;
         for (i=0; i<len; i++) {
             out = (int)(((int64_t)coefs[i] * c) >> 32);
-            coefs[i] = ((int)(out+round) >> s) * ssign;
+            coefs[i] = -((int)(out+round) >> s);
         }
     }
     else {
         s = s + 32;
-        round = 1 << (s-1);
-        for (i=0; i<len; i++) {
-            out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
-            coefs[i] = out * ssign;
+        if (s > 0) {
+            round = 1 << (s-1);
+            for (i=0; i<len; i++) {
+                out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
+                coefs[i] = -out;
+            }
+        } else {
+            for (i=0; i<len; i++)
+                coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
         }
     }
 }
@@ -385,7 +390,7 @@ static void apply_dependent_coupling_fixed(AACContext *ac,
                         for (k = offsets[i]; k < offsets[i + 1]; k++) {
                             tmp = (int)(((int64_t)src[group * 128 + k] * c + \
                                        (int64_t)0x1000000000) >> 37);
-                            dest[group * 128 + k] += (tmp + round) >> shift;
+                            dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
                         }
                     }
                 }
@@ -436,7 +441,7 @@ static void apply_independent_coupling_fixed(AACContext *ac,
     else {
       for (i = 0; i < len; i++) {
           tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
-          dest[i] += tmp * (1 << shift);
+          dest[i] += tmp * (1U << shift);
       }
     }
 }

libavcodec/aacdec_template.c

@@ -1157,6 +1157,9 @@ static av_cold int aac_decode_init(AVCodecContext *avctx)
     AACContext *ac = avctx->priv_data;
     int ret;
 
+    if (avctx->sample_rate > 96000)
+        return AVERROR_INVALIDDATA;
+
     ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
     if (ret != 0)
         return AVERROR_UNKNOWN;
@@ -1673,25 +1676,24 @@ static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
                 }
             } else if (cbt_m1 == NOISE_BT - 1) {
                 for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
-#if !USE_FIXED
-                    float scale;
-#endif /* !USE_FIXED */
                     INTFLOAT band_energy;
-
+#if USE_FIXED
                     for (k = 0; k < off_len; k++) {
                         ac->random_state  = lcg_random(ac->random_state);
-#if USE_FIXED
                         cfo[k] = ac->random_state >> 3;
-#else
-                        cfo[k] = ac->random_state;
-#endif /* USE_FIXED */
                     }
 
-#if USE_FIXED
                     band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
                     band_energy = fixed_sqrt(band_energy, 31);
                     noise_scale(cfo, sf[idx], band_energy, off_len);
 #else
+                    float scale;
+
+                    for (k = 0; k < off_len; k++) {
+                        ac->random_state  = lcg_random(ac->random_state);
+                        cfo[k] = ac->random_state;
+                    }
+
                     band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
                     scale = sf[idx] / sqrtf(band_energy);
                     ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
@@ -2493,6 +2495,9 @@ static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
     INTFLOAT tmp[TNS_MAX_ORDER+1];
     UINTFLOAT *coef = coef_param;
 
+    if(!mmm)
+        return;
+
     for (w = 0; w < ics->num_windows; w++) {
         bottom = ics->num_swb;
         for (filt = 0; filt < tns->n_filt[w]; filt++) {
@@ -2657,7 +2662,7 @@ static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
         ac->mdct.imdct_half(&ac->mdct, buf, in);
 #if USE_FIXED
         for (i=0; i<1024; i++)
-          buf[i] = (buf[i] + 4) >> 3;
+          buf[i] = (buf[i] + 4LL) >> 3;
 #endif /* USE_FIXED */
     }
 
@@ -2802,7 +2807,7 @@ static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
 
 static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
 {
-    INTFLOAT *in    = sce->coeffs;
+    UINTFLOAT *in   = sce->coeffs;
     INTFLOAT *out   = sce->ret;
     INTFLOAT *saved = sce->saved;
     INTFLOAT *buf  = ac->buf_mdct;
@@ -3122,6 +3127,7 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
     int samples = 0, multiplier, audio_found = 0, pce_found = 0;
     int is_dmono, sce_count = 0;
     int payload_alignment;
+    uint8_t che_presence[4][MAX_ELEM_ID] = {{0}};
 
     ac->frame = data;
 
@@ -3159,6 +3165,17 @@ static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
         }
 
         if (elem_type < TYPE_DSE) {
+            if (che_presence[elem_type][elem_id]) {
+                int error = che_presence[elem_type][elem_id] > 1;
+                av_log(ac->avctx, error ? AV_LOG_ERROR : AV_LOG_DEBUG, "channel element %d.%d duplicate\n",
+                       elem_type, elem_id);
+                if (error) {
+                    err = AVERROR_INVALIDDATA;
+                    goto fail;
+                }
+            }
+            che_presence[elem_type][elem_id]++;
+
             if (!(che=get_che(ac, elem_type, elem_id))) {
                 av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
                        elem_type, elem_id);
@@ -3324,20 +3341,14 @@ static int aac_decode_frame(AVCodecContext *avctx, void *data,
                                        AV_PKT_DATA_JP_DUALMONO,
                                        &jp_dualmono_size);
 
-    if (new_extradata && 0) {
-        av_free(avctx->extradata);
-        avctx->extradata = av_mallocz(new_extradata_size +
-                                      AV_INPUT_BUFFER_PADDING_SIZE);
-        if (!avctx->extradata)
-            return AVERROR(ENOMEM);
-        avctx->extradata_size = new_extradata_size;
-        memcpy(avctx->extradata, new_extradata, new_extradata_size);
-        push_output_configuration(ac);
-        if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
-                                         avctx->extradata,
-                                         avctx->extradata_size*8LL, 1) < 0) {
-            pop_output_configuration(ac);
-            return AVERROR_INVALIDDATA;
+    if (new_extradata) {
+        /* discard previous configuration */
+        ac->oc[1].status = OC_NONE;
+        err = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
+                                           new_extradata,
+                                           new_extradata_size * 8LL, 1);
+        if (err < 0) {
+            return err;
         }
     }
 

libavcodec/aacenc.c

@@ -28,6 +28,7 @@
  *              TODOs:
  * add sane pulse detection
  ***********************************/
+#include <float.h>
 
 #include "libavutil/libm.h"
 #include "libavutil/thread.h"
@@ -855,7 +856,7 @@ static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                 /* Not so fast though */
                 ratio = sqrtf(ratio);
             }
-            s->lambda = FFMIN(s->lambda * ratio, 65536.f);
+            s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);
 
             /* Keep iterating if we must reduce and lambda is in the sky */
             if (ratio > 0.9f && ratio < 1.1f) {
@@ -900,7 +901,7 @@ static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
 
-    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
+    av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);
 
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
@@ -982,11 +983,13 @@ static av_cold int aac_encode_init(AVCodecContext *avctx)
     }
 
     if (s->needs_pce) {
+        char buf[64];
         for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
             if (avctx->channel_layout == aac_pce_configs[i].layout)
                 break;
-        ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout\n");
-        av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout\n");
+        av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout);
+        ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf);
+        av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
         s->pce = aac_pce_configs[i];
         s->reorder_map = s->pce.reorder_map;
         s->chan_map = s->pce.config_map;

libavcodec/aacps.c

@@ -414,33 +414,33 @@ static void hybrid_synthesis(PSDSPContext *dsp, INTFLOAT out[2][38][64],
             memset(out[0][n], 0, 5*sizeof(out[0][n][0]));
             memset(out[1][n], 0, 5*sizeof(out[1][n][0]));
             for (i = 0; i < 12; i++) {
-                out[0][n][0] += in[   i][n][0];
-                out[1][n][0] += in[   i][n][1];
+                out[0][n][0] += (UINTFLOAT)in[   i][n][0];
+                out[1][n][0] += (UINTFLOAT)in[   i][n][1];
             }
             for (i = 0; i < 8; i++) {
-                out[0][n][1] += in[12+i][n][0];
-                out[1][n][1] += in[12+i][n][1];
+                out[0][n][1] += (UINTFLOAT)in[12+i][n][0];
+                out[1][n][1] += (UINTFLOAT)in[12+i][n][1];
             }
             for (i = 0; i < 4; i++) {
-                out[0][n][2] += in[20+i][n][0];
-                out[1][n][2] += in[20+i][n][1];
-                out[0][n][3] += in[24+i][n][0];
-                out[1][n][3] += in[24+i][n][1];
-                out[0][n][4] += in[28+i][n][0];
-                out[1][n][4] += in[28+i][n][1];
+                out[0][n][2] += (UINTFLOAT)in[20+i][n][0];
+                out[1][n][2] += (UINTFLOAT)in[20+i][n][1];
+                out[0][n][3] += (UINTFLOAT)in[24+i][n][0];
+                out[1][n][3] += (UINTFLOAT)in[24+i][n][1];
+                out[0][n][4] += (UINTFLOAT)in[28+i][n][0];
+                out[1][n][4] += (UINTFLOAT)in[28+i][n][1];
             }
         }
         dsp->hybrid_synthesis_deint(out, in + 27, 5, len);
     } else {
         for (n = 0; n < len; n++) {
-            out[0][n][0] = in[0][n][0] + in[1][n][0] + in[2][n][0] +
-                           in[3][n][0] + in[4][n][0] + in[5][n][0];
-            out[1][n][0] = in[0][n][1] + in[1][n][1] + in[2][n][1] +
-                           in[3][n][1] + in[4][n][1] + in[5][n][1];
-            out[0][n][1] = in[6][n][0] + in[7][n][0];
-            out[1][n][1] = in[6][n][1] + in[7][n][1];
-            out[0][n][2] = in[8][n][0] + in[9][n][0];
-            out[1][n][2] = in[8][n][1] + in[9][n][1];
+            out[0][n][0] = (UINTFLOAT)in[0][n][0] + in[1][n][0] + in[2][n][0] +
+                           (UINTFLOAT)in[3][n][0] + in[4][n][0] + in[5][n][0];
+            out[1][n][0] = (UINTFLOAT)in[0][n][1] + in[1][n][1] + in[2][n][1] +
+                           (UINTFLOAT)in[3][n][1] + in[4][n][1] + in[5][n][1];
+            out[0][n][1] = (UINTFLOAT)in[6][n][0] + in[7][n][0];
+            out[1][n][1] = (UINTFLOAT)in[6][n][1] + in[7][n][1];
+            out[0][n][2] = (UINTFLOAT)in[8][n][0] + in[9][n][0];
+            out[1][n][2] = (UINTFLOAT)in[8][n][1] + in[9][n][1];
         }
         dsp->hybrid_synthesis_deint(out, in + 7, 3, len);
     }

libavcodec/aacpsdsp_template.c

@@ -54,10 +54,10 @@ static void ps_hybrid_analysis_c(INTFLOAT (*out)[2], INTFLOAT (*in)[2],
         INT64FLOAT sum_im = (INT64FLOAT)filter[i][6][0] * in[6][1];
 
         for (j = 0; j < 6; j++) {
-            INTFLOAT in0_re = in[j][0];
-            INTFLOAT in0_im = in[j][1];
-            INTFLOAT in1_re = in[12-j][0];
-            INTFLOAT in1_im = in[12-j][1];
+            INT64FLOAT in0_re = in[j][0];
+            INT64FLOAT in0_im = in[j][1];
+            INT64FLOAT in1_re = in[12-j][0];
+            INT64FLOAT in1_im = in[12-j][1];
             sum_re += (INT64FLOAT)filter[i][j][0] * (in0_re + in1_re) -
                       (INT64FLOAT)filter[i][j][1] * (in0_im - in1_im);
             sum_im += (INT64FLOAT)filter[i][j][0] * (in0_im + in1_im) +
@@ -150,10 +150,10 @@ static void ps_stereo_interpolate_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
     INTFLOAT h1 = h[0][1];
     INTFLOAT h2 = h[0][2];
     INTFLOAT h3 = h[0][3];
-    INTFLOAT hs0 = h_step[0][0];
-    INTFLOAT hs1 = h_step[0][1];
-    INTFLOAT hs2 = h_step[0][2];
-    INTFLOAT hs3 = h_step[0][3];
+    UINTFLOAT hs0 = h_step[0][0];
+    UINTFLOAT hs1 = h_step[0][1];
+    UINTFLOAT hs2 = h_step[0][2];
+    UINTFLOAT hs3 = h_step[0][3];
     int n;
 
     for (n = 0; n < len; n++) {
@@ -181,10 +181,10 @@ static void ps_stereo_interpolate_ipdopd_c(INTFLOAT (*l)[2], INTFLOAT (*r)[2],
     INTFLOAT h01  = h[0][1],      h11  = h[1][1];
     INTFLOAT h02  = h[0][2],      h12  = h[1][2];
     INTFLOAT h03  = h[0][3],      h13  = h[1][3];
-    INTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
-    INTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
-    INTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
-    INTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
+    UINTFLOAT hs00 = h_step[0][0], hs10 = h_step[1][0];
+    UINTFLOAT hs01 = h_step[0][1], hs11 = h_step[1][1];
+    UINTFLOAT hs02 = h_step[0][2], hs12 = h_step[1][2];
+    UINTFLOAT hs03 = h_step[0][3], hs13 = h_step[1][3];
     int n;
 
     for (n = 0; n < len; n++) {

libavcodec/aacpsy.c

@@ -308,6 +308,9 @@ static av_cold int psy_3gpp_init(FFPsyContext *ctx) {
     const int bandwidth    = ctx->cutoff ? ctx->cutoff : AAC_CUTOFF(ctx->avctx);
     const float num_bark   = calc_bark((float)bandwidth);
 
+    if (bandwidth <= 0)
+        return AVERROR(EINVAL);
+
     ctx->model_priv_data = av_mallocz(sizeof(AacPsyContext));
     if (!ctx->model_priv_data)
         return AVERROR(ENOMEM);
@@ -794,7 +797,7 @@ static void psy_3gpp_analyze_channel(FFPsyContext *ctx, int channel,
 
         if (pe < 1.15f * desired_pe) {
             /* 6.6.1.3.6 "Final threshold modification by linearization" */
-            norm_fac = 1.0f / norm_fac;
+            norm_fac = norm_fac ? 1.0f / norm_fac : 0;
             for (w = 0; w < wi->num_windows*16; w += 16) {
                 for (g = 0; g < num_bands; g++) {
                     AacPsyBand *band = &pch->band[w+g];

libavcodec/aacsbr_tablegen_common.h

@@ -111,16 +111,4 @@ static DECLARE_ALIGNED(32, INTFLOAT, sbr_qmf_window_us)[640] = {
     Q31( 0.8537385600f),
 };
 
-static av_cold void aacsbr_tableinit(void)
-{
-    int n;
-    for (n = 1; n < 320; n++)
-        sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
-    sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
-    sbr_qmf_window_us[512] = -sbr_qmf_window_us[512];
-
-    for (n = 0; n < 320; n++)
-        sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
-}
-
 #endif /* AVCODEC_AACSBR_TABLEGEN_COMMON_H */

libavcodec/aacsbr_template.c

@@ -34,6 +34,18 @@
 
 #include "libavutil/qsort.h"
 
+static av_cold void aacsbr_tableinit(void)
+{
+    int n;
+    for (n = 1; n < 320; n++)
+        sbr_qmf_window_us[320 + n] = sbr_qmf_window_us[320 - n];
+    sbr_qmf_window_us[384] = -sbr_qmf_window_us[384];
+    sbr_qmf_window_us[512] = -sbr_qmf_window_us[512];
+
+    for (n = 0; n < 320; n++)
+        sbr_qmf_window_ds[n] = sbr_qmf_window_us[2*n];
+}
+
 av_cold void AAC_RENAME(ff_aac_sbr_init)(void)
 {
     static const struct {

libavcodec/aarch64/h264dsp_init_aarch64.c

@@ -34,20 +34,20 @@ void ff_h264_v_loop_filter_chroma_neon(uint8_t *pix, int stride, int alpha,
 void ff_h264_h_loop_filter_chroma_neon(uint8_t *pix, int stride, int alpha,
                                        int beta, int8_t *tc0);
 
-void ff_weight_h264_pixels_16_neon(uint8_t *dst, int stride, int height,
+void ff_weight_h264_pixels_16_neon(uint8_t *dst, ptrdiff_t stride, int height,
                                    int log2_den, int weight, int offset);
-void ff_weight_h264_pixels_8_neon(uint8_t *dst, int stride, int height,
+void ff_weight_h264_pixels_8_neon(uint8_t *dst, ptrdiff_t stride, int height,
                                   int log2_den, int weight, int offset);
-void ff_weight_h264_pixels_4_neon(uint8_t *dst, int stride, int height,
+void ff_weight_h264_pixels_4_neon(uint8_t *dst, ptrdiff_t stride, int height,
                                   int log2_den, int weight, int offset);
 
-void ff_biweight_h264_pixels_16_neon(uint8_t *dst, uint8_t *src, int stride,
+void ff_biweight_h264_pixels_16_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
                                      int height, int log2_den, int weightd,
                                      int weights, int offset);
-void ff_biweight_h264_pixels_8_neon(uint8_t *dst, uint8_t *src, int stride,
+void ff_biweight_h264_pixels_8_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
                                     int height, int log2_den, int weightd,
                                     int weights, int offset);
-void ff_biweight_h264_pixels_4_neon(uint8_t *dst, uint8_t *src, int stride,
+void ff_biweight_h264_pixels_4_neon(uint8_t *dst, uint8_t *src, ptrdiff_t stride,
                                     int height, int log2_den, int weightd,
                                     int weights, int offset);
 

libavcodec/aasc.c

@@ -104,26 +104,26 @@ static int aasc_decode_frame(AVCodecContext *avctx,
         ff_msrle_decode(avctx, s->frame, 8, &s->gb);
         break;
     case MKTAG('A', 'A', 'S', 'C'):
-    switch (compr) {
-    case 0:
-        stride = (avctx->width * psize + psize) & ~psize;
-        if (buf_size < stride * avctx->height)
+        switch (compr) {
+        case 0:
+            stride = (avctx->width * psize + psize) & ~psize;
+            if (buf_size < stride * avctx->height)
+                return AVERROR_INVALIDDATA;
+            for (i = avctx->height - 1; i >= 0; i--) {
+                memcpy(s->frame->data[0] + i * s->frame->linesize[0], buf, avctx->width * psize);
+                buf += stride;
+                buf_size -= stride;
+            }
+            break;
+        case 1:
+            bytestream2_init(&s->gb, buf, buf_size);
+            ff_msrle_decode(avctx, s->frame, 8, &s->gb);
+            break;
+        default:
+            av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
             return AVERROR_INVALIDDATA;
-        for (i = avctx->height - 1; i >= 0; i--) {
-            memcpy(s->frame->data[0] + i * s->frame->linesize[0], buf, avctx->width * psize);
-            buf += stride;
-            buf_size -= stride;
         }
         break;
-    case 1:
-        bytestream2_init(&s->gb, buf, buf_size);
-        ff_msrle_decode(avctx, s->frame, 8, &s->gb);
-        break;
-    default:
-        av_log(avctx, AV_LOG_ERROR, "Unknown compression type %d\n", compr);
-        return AVERROR_INVALIDDATA;
-    }
-        break;
     default:
         av_log(avctx, AV_LOG_ERROR, "Unknown FourCC: %X\n", avctx->codec_tag);
         return -1;

libavcodec/ac3.h

@@ -75,6 +75,7 @@
 #define AC3_DYNAMIC_RANGE1      0
 
 typedef int                     INTFLOAT;
+typedef unsigned int            UINTFLOAT;
 typedef int16_t                 SHORTFLOAT;
 
 #else /* USE_FIXED */
@@ -94,6 +95,7 @@ typedef int16_t                 SHORTFLOAT;
 #define AC3_DYNAMIC_RANGE1      1.0f
 
 typedef float                   INTFLOAT;
+typedef float                   UINTFLOAT;
 typedef float                   SHORTFLOAT;
 
 #endif /* USE_FIXED */

libavcodec/ac3_parser.c

@@ -162,7 +162,9 @@ int avpriv_ac3_parse_header(AC3HeaderInfo **phdr, const uint8_t *buf,
         return AVERROR(ENOMEM);
     hdr = *phdr;
 
-    init_get_bits8(&gb, buf, size);
+    err = init_get_bits8(&gb, buf, size);
+    if (err < 0)
+        return AVERROR_INVALIDDATA;
     err = ff_ac3_parse_header(&gb, hdr);
     if (err < 0)
         return AVERROR_INVALIDDATA;

libavcodec/ac3dec.c

@@ -106,25 +106,6 @@ static const uint8_t ac3_default_coeffs[8][5][2] = {
     { { 2, 7 }, { 5, 5 }, { 7, 2 }, { 6, 7 }, { 7, 6 }, },
 };
 
-static const uint64_t custom_channel_map_locations[16][2] = {
-    { 1, AV_CH_FRONT_LEFT },
-    { 1, AV_CH_FRONT_CENTER },
-    { 1, AV_CH_FRONT_RIGHT },
-    { 1, AV_CH_SIDE_LEFT },
-    { 1, AV_CH_SIDE_RIGHT },
-    { 0, AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER },
-    { 0, AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT },
-    { 0, AV_CH_BACK_CENTER },
-    { 0, AV_CH_TOP_CENTER },
-    { 0, AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT },
-    { 0, AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT },
-    { 0, AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT},
-    { 0, AV_CH_TOP_FRONT_CENTER },
-    { 0, AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT },
-    { 0, AV_CH_LOW_FREQUENCY_2 },
-    { 1, AV_CH_LOW_FREQUENCY },
-};
-
 /**
  * Symmetrical Dequantization
  * reference: Section 7.3.3 Expansion of Mantissas for Symmetrical Quantization
@@ -1690,6 +1671,7 @@ dependent_frame:
 
     if (s->frame_type == EAC3_FRAME_TYPE_DEPENDENT) {
         uint64_t ich_layout = avpriv_ac3_channel_layout_tab[s->prev_output_mode & ~AC3_OUTPUT_LFEON];
+        int channel_map_size = ff_ac3_channels_tab[s->output_mode & ~AC3_OUTPUT_LFEON] + s->lfe_on;
         uint64_t channel_layout;
         int extend = 0;
 
@@ -1699,30 +1681,41 @@ dependent_frame:
         channel_layout = ich_layout;
         for (ch = 0; ch < 16; ch++) {
             if (s->channel_map & (1 << (EAC3_MAX_CHANNELS - ch - 1))) {
-                channel_layout |= custom_channel_map_locations[ch][1];
+                channel_layout |= ff_eac3_custom_channel_map_locations[ch][1];
             }
         }
+        if (av_get_channel_layout_nb_channels(channel_layout) > EAC3_MAX_CHANNELS) {
+            av_log(avctx, AV_LOG_ERROR, "Too many channels (%d) coded\n",
+                   av_get_channel_layout_nb_channels(channel_layout));
+            return AVERROR_INVALIDDATA;
+        }
 
         avctx->channel_layout = channel_layout;
         avctx->channels = av_get_channel_layout_nb_channels(channel_layout);
 
         for (ch = 0; ch < EAC3_MAX_CHANNELS; ch++) {
             if (s->channel_map & (1 << (EAC3_MAX_CHANNELS - ch - 1))) {
-                if (custom_channel_map_locations[ch][0]) {
+                if (ff_eac3_custom_channel_map_locations[ch][0]) {
                     int index = av_get_channel_layout_channel_index(channel_layout,
-                                                                    custom_channel_map_locations[ch][1]);
+                                                                    ff_eac3_custom_channel_map_locations[ch][1]);
                     if (index < 0)
                         return AVERROR_INVALIDDATA;
+                    if (extend >= channel_map_size)
+                        return AVERROR_INVALIDDATA;
+
                     extended_channel_map[index] = offset + channel_map[extend++];
                 } else {
                     int i;
 
                     for (i = 0; i < 64; i++) {
-                        if ((1LL << i) & custom_channel_map_locations[ch][1]) {
+                        if ((1ULL << i) & ff_eac3_custom_channel_map_locations[ch][1]) {
                             int index = av_get_channel_layout_channel_index(channel_layout,
-                                                                            1LL << i);
+                                                                            1ULL << i);
                             if (index < 0)
                                 return AVERROR_INVALIDDATA;
+                            if (extend >= channel_map_size)
+                                return AVERROR_INVALIDDATA;
+
                             extended_channel_map[index] = offset + channel_map[extend++];
                         }
                     }
@@ -1738,7 +1731,9 @@ dependent_frame:
 
     for (ch = 0; ch < avctx->channels; ch++) {
         int map = extended_channel_map[ch];
-        memcpy((SHORTFLOAT *)frame->data[ch], s->output_buffer[map],
+        av_assert0(ch>=AV_NUM_DATA_POINTERS || frame->extended_data[ch] == frame->data[ch]);
+        memcpy((SHORTFLOAT *)frame->extended_data[ch],
+               s->output_buffer[map],
                s->num_blocks * AC3_BLOCK_SIZE * sizeof(SHORTFLOAT));
     }
 
@@ -1800,6 +1795,9 @@ dependent_frame:
 
     *got_frame_ptr = 1;
 
+    if (!s->superframe_size)
+        return FFMIN(full_buf_size, s->frame_size);
+
     return FFMIN(full_buf_size, s->superframe_size);
 }
 

libavcodec/ac3dec_fixed.c

@@ -107,29 +107,30 @@ static void scale_coefs (
       }
     } else {
       shift = -shift;
+      mul <<= shift;
       for (i=0; i<len; i+=8) {
 
           temp = src[i] * mul;
           temp1 = src[i+1] * mul;
           temp2 = src[i+2] * mul;
 
-          dst[i] = temp << shift;
+          dst[i] = temp;
           temp3 = src[i+3] * mul;
 
-          dst[i+1] = temp1 << shift;
+          dst[i+1] = temp1;
           temp4 = src[i + 4] * mul;
-          dst[i+2] = temp2 << shift;
+          dst[i+2] = temp2;
 
           temp5 = src[i+5] * mul;
-          dst[i+3] = temp3 << shift;
+          dst[i+3] = temp3;
           temp6 = src[i+6] * mul;
 
-          dst[i+4] = temp4 << shift;
+          dst[i+4] = temp4;
           temp7 = src[i+7] * mul;
 
-          dst[i+5] = temp5 << shift;
-          dst[i+6] = temp6 << shift;
-          dst[i+7] = temp7 << shift;
+          dst[i+5] = temp5;
+          dst[i+6] = temp6;
+          dst[i+7] = temp7;
 
       }
     }

libavcodec/ac3enc.c

@@ -1065,7 +1065,7 @@ static int bit_alloc(AC3EncodeContext *s, int snr_offset)
 {
     int blk, ch;
 
-    snr_offset = (snr_offset - 240) << 2;
+    snr_offset = (snr_offset - 240) * 4;
 
     reset_block_bap(s);
     for (blk = 0; blk < s->num_blocks; blk++) {
@@ -2051,7 +2051,8 @@ av_cold int ff_ac3_encode_close(AVCodecContext *avctx)
         av_freep(&block->cpl_coord_mant);
     }
 
-    s->mdct_end(s);
+    if (s->mdct_end)
+        s->mdct_end(s);
 
     return 0;
 }
@@ -2433,7 +2434,7 @@ av_cold int ff_ac3_encode_init(AVCodecContext *avctx)
 
     ret = validate_options(s);
     if (ret)
-        return ret;
+        goto init_fail;
 
     avctx->frame_size = AC3_BLOCK_SIZE * s->num_blocks;
     avctx->initial_padding = AC3_BLOCK_SIZE;

libavcodec/ac3tab.c

@@ -314,3 +314,21 @@ const uint16_t ff_eac3_default_chmap[8] = {
     AC3_CHMAP_L |               AC3_CHMAP_R | AC3_CHMAP_L_SUR |                  AC3_CHMAP_R_SUR,
     AC3_CHMAP_L | AC3_CHMAP_C | AC3_CHMAP_R | AC3_CHMAP_L_SUR |                  AC3_CHMAP_R_SUR
 };
+const uint64_t ff_eac3_custom_channel_map_locations[16][2] = {
+    { 1, AV_CH_FRONT_LEFT },
+    { 1, AV_CH_FRONT_CENTER },
+    { 1, AV_CH_FRONT_RIGHT },
+    { 1, AV_CH_SIDE_LEFT },
+    { 1, AV_CH_SIDE_RIGHT },
+    { 0, AV_CH_FRONT_LEFT_OF_CENTER | AV_CH_FRONT_RIGHT_OF_CENTER },
+    { 0, AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT },
+    { 0, AV_CH_BACK_CENTER },
+    { 0, AV_CH_TOP_CENTER },
+    { 0, AV_CH_SURROUND_DIRECT_LEFT | AV_CH_SURROUND_DIRECT_RIGHT },
+    { 0, AV_CH_WIDE_LEFT | AV_CH_WIDE_RIGHT },
+    { 0, AV_CH_TOP_FRONT_LEFT | AV_CH_TOP_FRONT_RIGHT},
+    { 0, AV_CH_TOP_FRONT_CENTER },
+    { 0, AV_CH_TOP_BACK_LEFT | AV_CH_TOP_BACK_RIGHT },
+    { 0, AV_CH_LOW_FREQUENCY_2 },
+    { 1, AV_CH_LOW_FREQUENCY },
+};

libavcodec/ac3tab.h

@@ -50,6 +50,8 @@ extern const uint16_t ff_ac3_fast_gain_tab[8];
 extern const uint16_t ff_eac3_default_chmap[8];
 extern const uint8_t  ff_ac3_band_start_tab[AC3_CRITICAL_BANDS+1];
 extern const uint8_t  ff_ac3_bin_to_band_tab[253];
+extern const uint64_t ff_eac3_custom_channel_map_locations[16][2];
+
 
 /** Custom channel map locations bitmask
  *  Other channels described in documentation:

libavcodec/adpcm.c

@@ -110,6 +110,10 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
     case AV_CODEC_ID_ADPCM_MTAF:
         min_channels = 2;
         max_channels = 8;
+        if (avctx->channels & 1) {
+            avpriv_request_sample(avctx, "channel count %d\n", avctx->channels);
+            return AVERROR_PATCHWELCOME;
+        }
         break;
     case AV_CODEC_ID_ADPCM_PSX:
         max_channels = 8;
@@ -135,8 +139,8 @@ static av_cold int adpcm_decode_init(AVCodecContext * avctx)
         break;
     case AV_CODEC_ID_ADPCM_IMA_APC:
         if (avctx->extradata && avctx->extradata_size >= 8) {
-            c->status[0].predictor = AV_RL32(avctx->extradata);
-            c->status[1].predictor = AV_RL32(avctx->extradata + 4);
+            c->status[0].predictor = av_clip_intp2(AV_RL32(avctx->extradata    ), 18);
+            c->status[1].predictor = av_clip_intp2(AV_RL32(avctx->extradata + 4), 18);
         }
         break;
     case AV_CODEC_ID_ADPCM_IMA_WS:
@@ -289,7 +293,7 @@ static inline int16_t adpcm_ima_oki_expand_nibble(ADPCMChannelStatus *c, int nib
     c->predictor = av_clip_intp2(predictor, 11);
     c->step_index = step_index;
 
-    return c->predictor << 4;
+    return c->predictor * 16;
 }
 
 static inline int16_t adpcm_ct_expand_nibble(ADPCMChannelStatus *c, int8_t nibble)
@@ -378,6 +382,10 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
             avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
             filter=0;
         }
+        if (shift < 0) {
+            avpriv_request_sample(avctx, "unknown XA-ADPCM shift %d", shift);
+            shift = 0;
+        }
         f0 = xa_adpcm_table[filter][0];
         f1 = xa_adpcm_table[filter][1];
 
@@ -388,7 +396,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
             d = in[16+i+j*4];
 
             t = sign_extend(d, 4);
-            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
+            s = t*(1<<shift) + ((s_1*f0 + s_2*f1+32)>>6);
             s_2 = s_1;
             s_1 = av_clip_int16(s);
             out0[j] = s_1;
@@ -403,10 +411,14 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
 
         shift  = 12 - (in[5+i*2] & 15);
         filter = in[5+i*2] >> 4;
-        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table)) {
+        if (filter >= FF_ARRAY_ELEMS(xa_adpcm_table) || shift < 0) {
             avpriv_request_sample(avctx, "unknown XA-ADPCM filter %d", filter);
             filter=0;
         }
+        if (shift < 0) {
+            avpriv_request_sample(avctx, "unknown XA-ADPCM shift %d", shift);
+            shift = 0;
+        }
 
         f0 = xa_adpcm_table[filter][0];
         f1 = xa_adpcm_table[filter][1];
@@ -415,7 +427,7 @@ static int xa_decode(AVCodecContext *avctx, int16_t *out0, int16_t *out1,
             d = in[16+i+j*4];
 
             t = sign_extend(d >> 4, 4);
-            s = ( t<<shift ) + ((s_1*f0 + s_2*f1+32)>>6);
+            s = t*(1<<shift) + ((s_1*f0 + s_2*f1+32)>>6);
             s_2 = s_1;
             s_1 = av_clip_int16(s);
             out1[j] = s_1;
@@ -1141,8 +1153,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                 return AVERROR_INVALIDDATA;
             }
         }
-        for (i=0; i<=st; i++)
+        for (i=0; i<=st; i++) {
             c->status[i].predictor  = bytestream2_get_le32u(&gb);
+            if (FFABS((int64_t)c->status[i].predictor) > (1<<16))
+                return AVERROR_INVALIDDATA;
+        }
 
         for (n = nb_samples >> (1 - st); n > 0; n--) {
             int byte   = bytestream2_get_byteu(&gb);
@@ -1189,8 +1204,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
 
             for (count2 = 0; count2 < 28; count2++) {
                 byte = bytestream2_get_byteu(&gb);
-                next_left_sample  = sign_extend(byte >> 4, 4) << shift_left;
-                next_right_sample = sign_extend(byte,      4) << shift_right;
+                next_left_sample  = sign_extend(byte >> 4, 4) * (1 << shift_left);
+                next_right_sample = sign_extend(byte,      4) * (1 << shift_right);
 
                 next_left_sample = (next_left_sample +
                     (current_left_sample * coeff1l) +
@@ -1229,7 +1244,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
             if (st) byte[1] = bytestream2_get_byteu(&gb);
             for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
                 for(channel = 0; channel < avctx->channels; channel++) {
-                    int sample = sign_extend(byte[channel] >> i, 4) << shift[channel];
+                    int sample = sign_extend(byte[channel] >> i, 4) * (1 << shift[channel]);
                     sample = (sample +
                              c->status[channel].sample1 * coeff[channel][0] +
                              c->status[channel].sample2 * coeff[channel][1] + 0x80) >> 8;
@@ -1290,10 +1305,10 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
 
                     for (count2=0; count2<28; count2++) {
                         if (count2 & 1)
-                            next_sample = sign_extend(byte,    4) << shift;
+                            next_sample = (unsigned)sign_extend(byte,    4) << shift;
                         else {
                             byte = bytestream2_get_byte(&gb);
-                            next_sample = sign_extend(byte >> 4, 4) << shift;
+                            next_sample = (unsigned)sign_extend(byte >> 4, 4) << shift;
                         }
 
                         next_sample += (current_sample  * coeff1) +
@@ -1344,11 +1359,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                     int level, pred;
                     int byte = bytestream2_get_byteu(&gb);
 
-                    level = sign_extend(byte >> 4, 4) << shift[n];
+                    level = sign_extend(byte >> 4, 4) * (1 << shift[n]);
                     pred  = s[-1] * coeff[0][n] + s[-2] * coeff[1][n];
                     s[0]  = av_clip_int16((level + pred + 0x80) >> 8);
 
-                    level = sign_extend(byte, 4) << shift[n];
+                    level = sign_extend(byte, 4) * (1 << shift[n]);
                     pred  = s[0] * coeff[0][n] + s[-1] * coeff[1][n];
                     s[1]  = av_clip_int16((level + pred + 0x80) >> 8);
                 }
@@ -1505,8 +1520,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                         sampledat = sign_extend(byte >> 4, 4);
                     }
 
-                    sampledat = ((prev1 * factor1 + prev2 * factor2) +
-                                 ((sampledat * scale) << 11)) >> 11;
+                    sampledat = ((prev1 * factor1 + prev2 * factor2) >> 11) +
+                                sampledat * scale;
                     *samples = av_clip_int16(sampledat);
                     prev2 = prev1;
                     prev1 = *samples++;
@@ -1568,8 +1583,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                 int byte = bytestream2_get_byteu(&gb);
                 int index = (byte >> 4) & 7;
                 unsigned int exp = byte & 0x0F;
-                int factor1 = table[ch][index * 2];
-                int factor2 = table[ch][index * 2 + 1];
+                int64_t factor1 = table[ch][index * 2];
+                int64_t factor2 = table[ch][index * 2 + 1];
 
                 /* Decode 14 samples.  */
                 for (n = 0; n < 14 && (i * 14 + n < nb_samples); n++) {
@@ -1583,7 +1598,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                     }
 
                     sampledat = ((c->status[ch].sample1 * factor1
-                                + c->status[ch].sample2 * factor2) >> 11) + (sampledat << exp);
+                                + c->status[ch].sample2 * factor2) >> 11) + sampledat * (1 << exp);
                     *samples = av_clip_int16(sampledat);
                     c->status[ch].sample2 = c->status[ch].sample1;
                     c->status[ch].sample1 = *samples++;
@@ -1630,7 +1645,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                     else
                         sampledat = sign_extend(byte >> 4, 4);
 
-                    sampledat = (((sampledat << 12) >> (header & 0xf)) << 6) + prev;
+                    sampledat = ((sampledat * (1 << 12)) >> (header & 0xf)) * (1 << 6) + prev;
                     *samples++ = av_clip_int16(sampledat >> 6);
                     c->status[channel].sample2 = c->status[channel].sample1;
                     c->status[channel].sample1 = sampledat;
@@ -1667,7 +1682,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx, void *data,
                             scale = sign_extend(byte, 4);
                         }
 
-                        scale  = scale << 12;
+                        scale  = scale * (1 << 12);
                         sample = (int)((scale >> shift) + (c->status[channel].sample1 * xa_adpcm_table[filter][0] + c->status[channel].sample2 * xa_adpcm_table[filter][1]) / 64);
                     }
                     *samples++ = av_clip_int16(sample);

libavcodec/adxenc.c

@@ -48,7 +48,7 @@ static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
     s2 = prev->s2;
     for (i = 0, j = 0; j < 32; i += channels, j++) {
         s0 = wav[i];
-        d = ((s0 << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
+        d = s0 + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
         if (max < d)
             max = d;
         if (min > d)
@@ -79,13 +79,13 @@ static void adx_encode(ADXContext *c, uint8_t *adx, const int16_t *wav,
     s1 = prev->s1;
     s2 = prev->s2;
     for (i = 0, j = 0; j < 32; i += channels, j++) {
-        d = ((wav[i] << COEFF_BITS) - c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS;
+        d = wav[i] + ((-c->coeff[0] * s1 - c->coeff[1] * s2) >> COEFF_BITS);
 
         d = av_clip_intp2(ROUNDED_DIV(d, scale), 3);
 
         put_sbits(&pb, 4, d);
 
-        s0 = ((d << COEFF_BITS) * scale + c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS;
+        s0 = d * scale + ((c->coeff[0] * s1 + c->coeff[1] * s2) >> COEFF_BITS);
         s2 = s1;
         s1 = s0;
     }

libavcodec/aic.c

@@ -208,6 +208,9 @@ static int aic_decode_coeffs(GetBitContext *gb, int16_t *dst,
     int mb, idx;
     unsigned val;
 
+    if (get_bits_left(gb) < 5)
+        return AVERROR_INVALIDDATA;
+
     has_skips  = get_bits1(gb);
     coeff_type = get_bits1(gb);
     coeff_bits = get_bits(gb, 3);

libavcodec/alac.c

@@ -171,12 +171,12 @@ static inline int sign_only(int v)
     return v ? FFSIGN(v) : 0;
 }
 
-static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
+static void lpc_prediction(int32_t *error_buffer, uint32_t *buffer_out,
                            int nb_samples, int bps, int16_t *lpc_coefs,
                            int lpc_order, int lpc_quant)
 {
     int i;
-    int32_t *pred = buffer_out;
+    uint32_t *pred = buffer_out;
 
     /* first sample always copies */
     *buffer_out = *error_buffer;
@@ -208,27 +208,27 @@ static void lpc_prediction(int32_t *error_buffer, int32_t *buffer_out,
     for (; i < nb_samples; i++) {
         int j;
         int val = 0;
-        int error_val = error_buffer[i];
+        unsigned error_val = error_buffer[i];
         int error_sign;
         int d = *pred++;
 
         /* LPC prediction */
         for (j = 0; j < lpc_order; j++)
             val += (pred[j] - d) * lpc_coefs[j];
-        val = (val + (1 << (lpc_quant - 1))) >> lpc_quant;
+        val = (val + (1LL << (lpc_quant - 1))) >> lpc_quant;
         val += d + error_val;
         buffer_out[i] = sign_extend(val, bps);
 
         /* adapt LPC coefficients */
         error_sign = sign_only(error_val);
         if (error_sign) {
-            for (j = 0; j < lpc_order && error_val * error_sign > 0; j++) {
+            for (j = 0; j < lpc_order && (int)(error_val * error_sign) > 0; j++) {
                 int sign;
                 val  = d - pred[j];
                 sign = sign_only(val) * error_sign;
                 lpc_coefs[j] -= sign;
-                val *= sign;
-                error_val -= (val >> lpc_quant) * (j + 1);
+                val *= (unsigned)sign;
+                error_val -= (val >> lpc_quant) * (j + 1U);
             }
         }
     }
@@ -250,10 +250,12 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
 
     alac->extra_bits = get_bits(&alac->gb, 2) << 3;
     bps = alac->sample_size - alac->extra_bits + channels - 1;
-    if (bps > 32U) {
+    if (bps > 32) {
         avpriv_report_missing_feature(avctx, "bps %d", bps);
         return AVERROR_PATCHWELCOME;
     }
+    if (bps < 1)
+        return AVERROR_INVALIDDATA;
 
     /* whether the frame is compressed */
     is_compressed = !get_bits1(&alac->gb);
@@ -300,13 +302,16 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
         decorr_shift       = get_bits(&alac->gb, 8);
         decorr_left_weight = get_bits(&alac->gb, 8);
 
+        if (channels == 2 && decorr_left_weight && decorr_shift > 31)
+            return AVERROR_INVALIDDATA;
+
         for (ch = 0; ch < channels; ch++) {
             prediction_type[ch]   = get_bits(&alac->gb, 4);
             lpc_quant[ch]         = get_bits(&alac->gb, 4);
             rice_history_mult[ch] = get_bits(&alac->gb, 3);
             lpc_order[ch]         = get_bits(&alac->gb, 5);
 
-            if (lpc_order[ch] >= alac->max_samples_per_frame)
+            if (lpc_order[ch] >= alac->max_samples_per_frame || !lpc_quant[ch])
                 return AVERROR_INVALIDDATA;
 
             /* read the predictor table */
@@ -395,13 +400,13 @@ static int decode_element(AVCodecContext *avctx, AVFrame *frame, int ch_index,
     case 20: {
         for (ch = 0; ch < channels; ch++) {
             for (i = 0; i < alac->nb_samples; i++)
-                alac->output_samples_buffer[ch][i] <<= 12;
+                alac->output_samples_buffer[ch][i] *= 1U << 12;
         }}
         break;
     case 24: {
         for (ch = 0; ch < channels; ch++) {
             for (i = 0; i < alac->nb_samples; i++)
-                alac->output_samples_buffer[ch][i] <<= 8;
+                alac->output_samples_buffer[ch][i] *= 1U << 8;
         }}
         break;
     }

libavcodec/alacdsp.c

@@ -29,12 +29,12 @@ static void decorrelate_stereo(int32_t *buffer[2], int nb_samples,
     int i;
 
     for (i = 0; i < nb_samples; i++) {
-        int32_t a, b;
+        uint32_t a, b;
 
         a = buffer[0][i];
         b = buffer[1][i];
 
-        a -= (b * decorr_left_weight) >> decorr_shift;
+        a -= (int)(b * decorr_left_weight) >> decorr_shift;
         b += a;
 
         buffer[0][i] = b;
@@ -49,7 +49,7 @@ static void append_extra_bits(int32_t *buffer[2], int32_t *extra_bits_buffer[2],
 
     for (ch = 0; ch < channels; ch++)
         for (i = 0; i < nb_samples; i++)
-            buffer[ch][i] = (buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
+            buffer[ch][i] = ((unsigned)buffer[ch][i] << extra_bits) | extra_bits_buffer[ch][i];
 }
 
 av_cold void ff_alacdsp_init(ALACDSPContext *c)

libavcodec/aliaspixdec.c

@@ -62,6 +62,9 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame,
     if (ret < 0)
         return ret;
 
+    if (bytestream2_get_bytes_left(&gb) < width*height / 255)
+        return AVERROR_INVALIDDATA;
+
     ret = ff_get_buffer(avctx, f, 0);
     if (ret < 0)
         return ret;

libavcodec/allcodecs.c

@@ -58,6 +58,7 @@ extern AVCodec ff_ayuv_decoder;
 extern AVCodec ff_bethsoftvid_decoder;
 extern AVCodec ff_bfi_decoder;
 extern AVCodec ff_bink_decoder;
+extern AVCodec ff_bitpacked_decoder;
 extern AVCodec ff_bmp_encoder;
 extern AVCodec ff_bmp_decoder;
 extern AVCodec ff_bmv_video_decoder;
@@ -153,6 +154,7 @@ extern AVCodec ff_huffyuv_encoder;
 extern AVCodec ff_huffyuv_decoder;
 extern AVCodec ff_idcin_decoder;
 extern AVCodec ff_iff_ilbm_decoder;
+extern AVCodec ff_imm4_decoder;
 extern AVCodec ff_indeo2_decoder;
 extern AVCodec ff_indeo3_decoder;
 extern AVCodec ff_indeo4_decoder;
@@ -211,6 +213,7 @@ extern AVCodec ff_mszh_decoder;
 extern AVCodec ff_mts2_decoder;
 extern AVCodec ff_mvc1_decoder;
 extern AVCodec ff_mvc2_decoder;
+extern AVCodec ff_mwsc_decoder;
 extern AVCodec ff_mxpeg_decoder;
 extern AVCodec ff_nuv_decoder;
 extern AVCodec ff_paf_video_decoder;
@@ -234,7 +237,7 @@ extern AVCodec ff_prores_encoder;
 extern AVCodec ff_prores_decoder;
 extern AVCodec ff_prores_aw_encoder;
 extern AVCodec ff_prores_ks_encoder;
-extern AVCodec ff_prores_lgpl_decoder;
+extern AVCodec ff_prosumer_decoder;
 extern AVCodec ff_psd_decoder;
 extern AVCodec ff_ptx_decoder;
 extern AVCodec ff_qdraw_decoder;
@@ -245,6 +248,7 @@ extern AVCodec ff_r10k_encoder;
 extern AVCodec ff_r10k_decoder;
 extern AVCodec ff_r210_encoder;
 extern AVCodec ff_r210_decoder;
+extern AVCodec ff_rasc_decoder;
 extern AVCodec ff_rawvideo_encoder;
 extern AVCodec ff_rawvideo_decoder;
 extern AVCodec ff_rl2_decoder;
@@ -334,8 +338,8 @@ extern AVCodec ff_vp9_decoder;
 extern AVCodec ff_vp9_rkmpp_decoder;
 extern AVCodec ff_vp9_v4l2m2m_decoder;
 extern AVCodec ff_vqa_decoder;
-extern AVCodec ff_bitpacked_decoder;
 extern AVCodec ff_webp_decoder;
+extern AVCodec ff_wcmv_decoder;
 extern AVCodec ff_wrapped_avframe_encoder;
 extern AVCodec ff_wrapped_avframe_decoder;
 extern AVCodec ff_wmv1_encoder;
@@ -393,6 +397,7 @@ extern AVCodec ff_atrac3_decoder;
 extern AVCodec ff_atrac3al_decoder;
 extern AVCodec ff_atrac3p_decoder;
 extern AVCodec ff_atrac3pal_decoder;
+extern AVCodec ff_atrac9_decoder;
 extern AVCodec ff_binkaudio_dct_decoder;
 extern AVCodec ff_binkaudio_rdft_decoder;
 extern AVCodec ff_bmv_audio_decoder;
@@ -419,6 +424,7 @@ extern AVCodec ff_g729_decoder;
 extern AVCodec ff_gsm_decoder;
 extern AVCodec ff_gsm_ms_decoder;
 extern AVCodec ff_iac_decoder;
+extern AVCodec ff_ilbc_decoder;
 extern AVCodec ff_imc_decoder;
 extern AVCodec ff_interplay_acm_decoder;
 extern AVCodec ff_mace3_decoder;
@@ -546,6 +552,8 @@ extern AVCodec ff_pcm_u32be_encoder;
 extern AVCodec ff_pcm_u32be_decoder;
 extern AVCodec ff_pcm_u32le_encoder;
 extern AVCodec ff_pcm_u32le_decoder;
+extern AVCodec ff_pcm_vidc_encoder;
+extern AVCodec ff_pcm_vidc_decoder;
 extern AVCodec ff_pcm_zork_decoder;
 
 /* DPCM codecs */
@@ -668,6 +676,7 @@ extern AVCodec ff_libaom_av1_encoder;
 extern AVCodec ff_libcelt_decoder;
 extern AVCodec ff_libcodec2_encoder;
 extern AVCodec ff_libcodec2_decoder;
+extern AVCodec ff_libdavs2_decoder;
 extern AVCodec ff_libfdk_aac_encoder;
 extern AVCodec ff_libfdk_aac_decoder;
 extern AVCodec ff_libgsm_encoder;
@@ -706,6 +715,7 @@ extern AVCodec ff_libx264_encoder;
 extern AVCodec ff_libx264rgb_encoder;
 extern AVCodec ff_libx265_encoder;
 extern AVCodec ff_libxavs_encoder;
+extern AVCodec ff_libxavs2_encoder;
 extern AVCodec ff_libxvid_encoder;
 extern AVCodec ff_libzvbi_teletext_decoder;
 
@@ -761,7 +771,15 @@ extern AVCodec ff_vp9_cuvid_decoder;
 extern AVCodec ff_vp9_mediacodec_decoder;
 extern AVCodec ff_vp9_vaapi_encoder;
 
+// The iterate API is not usable with ossfuzz due to the excessive size of binaries created
+#if CONFIG_OSSFUZZ
+AVCodec * codec_list[] = {
+    NULL,
+    NULL
+};
+#else
 #include "libavcodec/codec_list.c"
+#endif
 
 static AVOnce av_codec_static_init = AV_ONCE_INIT;
 static void av_codec_init_static(void)

libavcodec/alsdec.c

@@ -348,6 +348,11 @@ static av_cold int read_specific_config(ALSDecContext *ctx)
     if (als_id != MKBETAG('A','L','S','\0'))
         return AVERROR_INVALIDDATA;
 
+    if (avctx->channels > FF_SANE_NB_CHANNELS) {
+        avpriv_request_sample(avctx, "Huge number of channels\n");
+        return AVERROR_PATCHWELCOME;
+    }
+
     ctx->cur_frame_length = sconf->frame_length;
 
     // read channel config
@@ -487,7 +492,7 @@ static void parse_bs_info(const uint32_t bs_info, unsigned int n,
 static int32_t decode_rice(GetBitContext *gb, unsigned int k)
 {
     int max = get_bits_left(gb) - k;
-    int q   = get_unary(gb, 0, max);
+    unsigned q = get_unary(gb, 0, max);
     int r   = k ? get_bits1(gb) : !(q & 1);
 
     if (k > 1) {
@@ -507,7 +512,7 @@ static void parcor_to_lpc(unsigned int k, const int32_t *par, int32_t *cof)
     int i, j;
 
     for (i = 0, j = k - 1; i < j; i++, j--) {
-        int tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
+        unsigned tmp1 = ((MUL64(par[k], cof[j]) + (1 << 19)) >> 20);
         cof[j]  += ((MUL64(par[k], cof[i]) + (1 << 19)) >> 20);
         cof[i]  += tmp1;
     }
@@ -657,7 +662,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
 
     // do not continue in case of a damaged stream since
     // block_length must be evenly divisible by sub_blocks
-    if (bd->block_length & (sub_blocks - 1)) {
+    if (bd->block_length & (sub_blocks - 1) || bd->block_length <= 0) {
         av_log(avctx, AV_LOG_WARNING,
                "Block length is not evenly divisible by the number of subblocks.\n");
         return AVERROR_INVALIDDATA;
@@ -756,7 +761,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
             }
 
             for (k = 2; k < opt_order; k++)
-                quant_cof[k] = (quant_cof[k] * (1 << 14)) + (add_base << 13);
+                quant_cof[k] = (quant_cof[k] * (1U << 14)) + (add_base << 13);
         }
     }
 
@@ -767,8 +772,8 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
         if (*bd->use_ltp) {
             int r, c;
 
-            bd->ltp_gain[0]   = decode_rice(gb, 1) << 3;
-            bd->ltp_gain[1]   = decode_rice(gb, 2) << 3;
+            bd->ltp_gain[0]   = decode_rice(gb, 1) * 8;
+            bd->ltp_gain[1]   = decode_rice(gb, 2) * 8;
 
             r                 = get_unary(gb, 0, 4);
             c                 = get_bits(gb, 2);
@@ -779,8 +784,8 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
 
             bd->ltp_gain[2]   = ltp_gain_values[r][c];
 
-            bd->ltp_gain[3]   = decode_rice(gb, 2) << 3;
-            bd->ltp_gain[4]   = decode_rice(gb, 1) << 3;
+            bd->ltp_gain[3]   = decode_rice(gb, 2) * 8;
+            bd->ltp_gain[4]   = decode_rice(gb, 1) * 8;
 
             *bd->ltp_lag      = get_bits(gb, ctx->ltp_lag_length);
             *bd->ltp_lag     += FFMAX(4, opt_order + 1);
@@ -789,14 +794,20 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
 
     // read first value and residuals in case of a random access block
     if (bd->ra_block) {
+        start = FFMIN(opt_order, 3);
+        av_assert0(sb_length <= sconf->frame_length);
+        if (sb_length <= start) {
+            // opt_order or sb_length may be corrupted, either way this is unsupported and not well defined in the specification
+            av_log(avctx, AV_LOG_ERROR, "Sub block length smaller or equal start\n");
+            return AVERROR_PATCHWELCOME;
+        }
+
         if (opt_order)
             bd->raw_samples[0] = decode_rice(gb, avctx->bits_per_raw_sample - 4);
         if (opt_order > 1)
             bd->raw_samples[1] = decode_rice(gb, FFMIN(s[0] + 3, ctx->s_max));
         if (opt_order > 2)
             bd->raw_samples[2] = decode_rice(gb, FFMIN(s[0] + 1, ctx->s_max));
-
-        start = FFMIN(opt_order, 3);
     }
 
     // read all residuals
@@ -810,7 +821,9 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
         unsigned int low;
         unsigned int value;
 
-        ff_bgmc_decode_init(gb, &high, &low, &value);
+        int ret = ff_bgmc_decode_init(gb, &high, &low, &value);
+        if (ret < 0)
+            return ret;
 
         current_res = bd->raw_samples + start;
 
@@ -820,6 +833,9 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
             k    [sb] = s[sb] > b ? s[sb] - b : 0;
             delta[sb] = 5 - s[sb] + k[sb];
 
+            if (k[sb] >= 32)
+                return AVERROR_INVALIDDATA;
+
             ff_bgmc_decode(gb, sb_len, current_res,
                         delta[sb], sx[sb], &high, &low, &value, ctx->bgmc_lut, ctx->bgmc_lut_status);
 
@@ -861,7 +877,7 @@ static int read_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
                     res >>= 1;
 
                     if (cur_k) {
-                        res  *= 1 << cur_k;
+                        res  *= 1U << cur_k;
                         res  |= get_bits_long(gb, cur_k);
                     }
                 }
@@ -912,7 +928,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
             y = 1 << 6;
 
             for (base = begin; base < end; base++, tab++)
-                y += MUL64(bd->ltp_gain[tab], raw_samples[base]);
+                y += (uint64_t)MUL64(bd->ltp_gain[tab], raw_samples[base]);
 
             raw_samples[ltp_smp] += y >> 7;
         }
@@ -924,7 +940,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
             y = 1 << 19;
 
             for (sb = 0; sb < smp; sb++)
-                y += MUL64(lpc_cof[sb], raw_samples[-(sb + 1)]);
+                y += (uint64_t)MUL64(lpc_cof[sb], raw_samples[-(sb + 1)]);
 
             *raw_samples++ -= y >> 20;
             parcor_to_lpc(smp, quant_cof, lpc_cof);
@@ -940,7 +956,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
 
         // reconstruct difference signal for prediction (joint-stereo)
         if (bd->js_blocks && bd->raw_other) {
-            int32_t *left, *right;
+            uint32_t *left, *right;
 
             if (bd->raw_other > raw_samples) {  // D = R - L
                 left  = raw_samples;
@@ -974,7 +990,7 @@ static int decode_var_block_data(ALSDecContext *ctx, ALSBlockData *bd)
         y = 1 << 19;
 
         for (sb = -opt_order; sb < 0; sb++)
-            y += MUL64(lpc_cof[sb], raw_samples[sb]);
+            y += (uint64_t)MUL64(lpc_cof[sb], raw_samples[sb]);
 
         *raw_samples -= y >> 20;
     }
@@ -999,6 +1015,10 @@ static int read_block(ALSDecContext *ctx, ALSBlockData *bd)
     ALSSpecificConfig *sconf = &ctx->sconf;
 
     *bd->shift_lsbs = 0;
+
+    if (get_bits_left(gb) < 7)
+        return AVERROR_INVALIDDATA;
+
     // read block type flag and read the samples accordingly
     if (get_bits1(gb)) {
         ret = read_var_block_data(ctx, bd);
@@ -1033,7 +1053,7 @@ static int decode_block(ALSDecContext *ctx, ALSBlockData *bd)
 
     if (*bd->shift_lsbs)
         for (smp = 0; smp < bd->block_length; smp++)
-            bd->raw_samples[smp] <<= *bd->shift_lsbs;
+            bd->raw_samples[smp] = (unsigned)bd->raw_samples[smp] << *bd->shift_lsbs;
 
     return 0;
 }
@@ -1169,10 +1189,10 @@ static int decode_blocks(ALSDecContext *ctx, unsigned int ra_frame,
                 av_log(ctx->avctx, AV_LOG_WARNING, "Invalid channel pair.\n");
 
             for (s = 0; s < div_blocks[b]; s++)
-                bd[0].raw_samples[s] = bd[1].raw_samples[s] - bd[0].raw_samples[s];
+                bd[0].raw_samples[s] = bd[1].raw_samples[s] - (unsigned)bd[0].raw_samples[s];
         } else if (bd[1].js_blocks) {
             for (s = 0; s < div_blocks[b]; s++)
-                bd[1].raw_samples[s] = bd[1].raw_samples[s] + bd[0].raw_samples[s];
+                bd[1].raw_samples[s] = bd[1].raw_samples[s] + (unsigned)bd[0].raw_samples[s];
         }
 
         offset  += div_blocks[b];
@@ -1379,6 +1399,9 @@ static SoftFloat_IEEE754 multiply(SoftFloat_IEEE754 a, SoftFloat_IEEE754 b) {
     mantissa_temp = (uint64_t)a.mant * (uint64_t)b.mant;
     mask_64       = (uint64_t)0x1 << 47;
 
+    if (!mantissa_temp)
+        return FLOAT_0;
+
     // Count the valid bit count
     while (!(mantissa_temp & mask_64) && mask_64) {
         bit_count--;
@@ -1395,7 +1418,11 @@ static SoftFloat_IEEE754 multiply(SoftFloat_IEEE754 a, SoftFloat_IEEE754 b) {
         }
     }
 
-    mantissa = (unsigned int)(mantissa_temp >> cutoff_bit_count);
+    if (cutoff_bit_count >= 0) {
+        mantissa = (unsigned int)(mantissa_temp >> cutoff_bit_count);
+    } else {
+        mantissa = (unsigned int)(mantissa_temp <<-cutoff_bit_count);
+    }
 
     // Need one more shift?
     if (mantissa & 0x01000000ul) {
@@ -1407,7 +1434,7 @@ static SoftFloat_IEEE754 multiply(SoftFloat_IEEE754 a, SoftFloat_IEEE754 b) {
         return_val = 0x80000000U;
     }
 
-    return_val |= (a.exp + b.exp + bit_count - 47) << 23;
+    return_val |= ((unsigned)av_clip(a.exp + b.exp + bit_count - 47, -126, 127) << 23) & 0x7F800000;
     return_val |= mantissa;
     return av_bits2sf_ieee754(return_val);
 }
@@ -1452,6 +1479,9 @@ static int read_diff_float_data(ALSDecContext *ctx, unsigned int ra_frame) {
         ff_mlz_flush_dict(ctx->mlz);
     }
 
+    if (avctx->channels * 8 > get_bits_left(gb))
+        return AVERROR_INVALIDDATA;
+
     for (c = 0; c < avctx->channels; ++c) {
         if (use_acf) {
             //acf_flag
@@ -1792,15 +1822,17 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr,
     #define INTERLEAVE_OUTPUT(bps)                                                   \
     {                                                                                \
         int##bps##_t *dest = (int##bps##_t*)frame->data[0];                          \
+        int channels = avctx->channels;                                              \
+        int32_t **raw_samples = ctx->raw_samples;                                    \
         shift = bps - ctx->avctx->bits_per_raw_sample;                               \
         if (!ctx->cs_switch) {                                                       \
             for (sample = 0; sample < ctx->cur_frame_length; sample++)               \
-                for (c = 0; c < avctx->channels; c++)                                \
-                    *dest++ = ctx->raw_samples[c][sample] << shift;                  \
+                for (c = 0; c < channels; c++)                                       \
+                    *dest++ = raw_samples[c][sample] * (1U << shift);                \
         } else {                                                                     \
             for (sample = 0; sample < ctx->cur_frame_length; sample++)               \
-                for (c = 0; c < avctx->channels; c++)                                \
-                    *dest++ = ctx->raw_samples[sconf->chan_pos[c]][sample] << shift; \
+                for (c = 0; c < channels; c++)                                       \
+                    *dest++ = raw_samples[sconf->chan_pos[c]][sample] * (1U << shift);\
         }                                                                            \
     }
 
@@ -1984,6 +2016,8 @@ static av_cold int decode_init(AVCodecContext *avctx)
 
     // allocate quantized parcor coefficient buffer
     num_buffers = sconf->mc_coding ? avctx->channels : 1;
+    if (num_buffers * (uint64_t)num_buffers > INT_MAX) // protect chan_data_buffer allocation
+        return AVERROR_INVALIDDATA;
 
     ctx->quant_cof        = av_malloc_array(num_buffers, sizeof(*ctx->quant_cof));
     ctx->lpc_cof          = av_malloc_array(num_buffers, sizeof(*ctx->lpc_cof));
@@ -2116,7 +2150,6 @@ static av_cold int decode_init(AVCodecContext *avctx)
     return 0;
 
 fail:
-    decode_end(avctx);
     return ret;
 }
 
@@ -2142,4 +2175,5 @@ AVCodec ff_als_decoder = {
     .decode         = decode_frame,
     .flush          = flush,
     .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
+    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
 };

libavcodec/amfenc.c

@@ -438,7 +438,7 @@ static int amf_copy_buffer(AVCodecContext *avctx, AVPacket *pkt, AMFBuffer *buff
     int64_t          timestamp = AV_NOPTS_VALUE;
     int64_t          size = buffer->pVtbl->GetSize(buffer);
 
-    if ((ret = ff_alloc_packet2(avctx, pkt, size, 0)) < 0) {
+    if ((ret = av_new_packet(pkt, size)) < 0) {
         return ret;
     }
     memcpy(pkt->data, buffer->pVtbl->GetNative(buffer), size);

libavcodec/amrwbdec.c

@@ -862,15 +862,20 @@ static float find_hb_gain(AMRWBContext *ctx, const float *synth,
 {
     int wsp = (vad > 0);
     float tilt;
+    float tmp;
 
     if (ctx->fr_cur_mode == MODE_23k85)
         return qua_hb_gain[hb_idx] * (1.0f / (1 << 14));
 
-    tilt = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1) /
-           ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
+    tmp = ctx->celpm_ctx.dot_productf(synth, synth + 1, AMRWB_SFR_SIZE - 1);
+
+    if (tmp > 0) {
+        tilt = tmp / ctx->celpm_ctx.dot_productf(synth, synth, AMRWB_SFR_SIZE);
+    } else
+        tilt = 0;
 
     /* return gain bounded by [0.1, 1.0] */
-    return av_clipf((1.0 - FFMAX(0.0, tilt)) * (1.25 - 0.25 * wsp), 0.1, 1.0);
+    return av_clipf((1.0 - tilt) * (1.25 - 0.25 * wsp), 0.1, 1.0);
 }
 
 /**

libavcodec/anm.c

@@ -54,7 +54,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
 
     bytestream2_skipu(&s->gb, 16 * 8);
     for (i = 0; i < 256; i++)
-        s->palette[i] = bytestream2_get_le32u(&s->gb);
+        s->palette[i] = (0xFFU << 24) | bytestream2_get_le32u(&s->gb);
 
     return 0;
 }
@@ -119,6 +119,9 @@ static int decode_frame(AVCodecContext *avctx,
     uint8_t *dst, *dst_end;
     int count, ret;
 
+    if (buf_size < 7)
+        return AVERROR_INVALIDDATA;
+
     if ((ret = ff_reget_buffer(avctx, s->frame)) < 0)
         return ret;
     dst     = s->frame->data[0];

libavcodec/ansi.c

@@ -430,7 +430,8 @@ static int decode_frame(AVCodecContext *avctx,
                     s->args[s->nb_args] = FFMAX(s->args[s->nb_args], 0) * 10 + buf[0] - '0';
                 break;
             case ';':
-                s->nb_args++;
+                if (s->nb_args < MAX_NB_ARGS)
+                    s->nb_args++;
                 if (s->nb_args < MAX_NB_ARGS)
                     s->args[s->nb_args] = 0;
                 break;
@@ -473,6 +474,11 @@ static av_cold int decode_close(AVCodecContext *avctx)
     return 0;
 }
 
+static const AVCodecDefault ansi_defaults[] = {
+    { "max_pixels", "640*480" },
+    { NULL },
+};
+
 AVCodec ff_ansi_decoder = {
     .name           = "ansi",
     .long_name      = NULL_IF_CONFIG_SMALL("ASCII/ANSI art"),
@@ -484,4 +490,5 @@ AVCodec ff_ansi_decoder = {
     .decode         = decode_frame,
     .capabilities   = AV_CODEC_CAP_DR1,
     .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE,
+    .defaults       = ansi_defaults,
 };

libavcodec/apedec.c

@@ -101,7 +101,7 @@ typedef struct APEFilter {
     int16_t *historybuffer; ///< filter memory
     int16_t *delay;         ///< filtered values
 
-    int avg;
+    uint32_t avg;
 } APEFilter;
 
 typedef struct APERice {
@@ -460,7 +460,7 @@ static inline void update_rice(APERice *rice, unsigned int x)
 
     if (rice->ksum < lim)
         rice->k--;
-    else if (rice->ksum >= (1 << (rice->k + 5)))
+    else if (rice->ksum >= (1 << (rice->k + 5)) && rice->k < 24)
         rice->k++;
 }
 
@@ -554,7 +554,7 @@ static inline int ape_decode_value_3990(APEContext *ctx, APERice *rice)
     overflow = range_get_symbol(ctx, counts_3980, counts_diff_3980);
 
     if (overflow == (MODEL_ELEMENTS - 1)) {
-        overflow  = range_decode_bits(ctx, 16) << 16;
+        overflow  = (unsigned)range_decode_bits(ctx, 16) << 16;
         overflow |= range_decode_bits(ctx, 16);
     }
 
@@ -589,7 +589,7 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
                               int32_t *out, APERice *rice, int blockstodecode)
 {
     int i;
-    int ksummax, ksummin;
+    unsigned ksummax, ksummin;
 
     rice->ksum = 0;
     for (i = 0; i < FFMIN(blockstodecode, 5); i++) {
@@ -610,7 +610,7 @@ static void decode_array_0000(APEContext *ctx, GetBitContext *gb,
     ksummin = rice->k ? (1 << rice->k + 6) : 0;
     for (; i < blockstodecode; i++) {
         out[i] = get_rice_ook(&ctx->gb, rice->k);
-        rice->ksum += out[i] - out[i - 64];
+        rice->ksum += out[i] - (unsigned)out[i - 64];
         while (rice->ksum < ksummin) {
             rice->k--;
             ksummin = rice->k ? ksummin >> 1 : 0;
@@ -836,7 +836,7 @@ static av_always_inline int filter_fast_3320(APEPredictor *p,
     else
         p->coeffsA[filter][0]--;
 
-    p->filterA[filter] += p->lastA[filter];
+    p->filterA[filter] += (unsigned)p->lastA[filter];
 
     return p->filterA[filter];
 }
@@ -859,9 +859,9 @@ static av_always_inline int filter_3800(APEPredictor *p,
         return predictionA;
     }
     d2 =  p->buf[delayA];
-    d1 = (p->buf[delayA] - p->buf[delayA - 1]) << 1;
-    d0 =  p->buf[delayA] + ((p->buf[delayA - 2] - p->buf[delayA - 1]) << 3);
-    d3 =  p->buf[delayB] * 2 - p->buf[delayB - 1];
+    d1 = (p->buf[delayA] - (unsigned)p->buf[delayA - 1]) * 2;
+    d0 =  p->buf[delayA] + ((p->buf[delayA - 2] - (unsigned)p->buf[delayA - 1]) * 8);
+    d3 =  p->buf[delayB] * 2U - p->buf[delayB - 1];
     d4 =  p->buf[delayB];
 
     predictionA = d0 * p->coeffsA[filter][0] +
@@ -880,8 +880,8 @@ static av_always_inline int filter_3800(APEPredictor *p,
     p->coeffsB[filter][0] += (((d3 >> 29) & 4) - 2) * sign;
     p->coeffsB[filter][1] -= (((d4 >> 30) & 2) - 1) * sign;
 
-    p->filterB[filter] = p->lastA[filter] + (predictionB >> shift);
-    p->filterA[filter] = p->filterB[filter] + ((p->filterA[filter] * 31) >> 5);
+    p->filterB[filter] = p->lastA[filter] + (unsigned)(predictionB >> shift);
+    p->filterA[filter] = p->filterB[filter] + (unsigned)((int)(p->filterA[filter] * 31U) >> 5);
 
     return p->filterA[filter];
 }
@@ -902,10 +902,10 @@ static void long_filter_high_3800(int32_t *buffer, int order, int shift, int len
         dotprod = 0;
         sign = APESIGN(buffer[i]);
         for (j = 0; j < order; j++) {
-            dotprod += delay[j] * coeffs[j];
+            dotprod += delay[j] * (unsigned)coeffs[j];
             coeffs[j] += ((delay[j] >> 31) | 1) * sign;
         }
-        buffer[i] -= dotprod >> shift;
+        buffer[i] -= (unsigned)(dotprod >> shift);
         for (j = 0; j < order - 1; j++)
             delay[j] = delay[j + 1];
         delay[order - 1] = buffer[i];
@@ -916,7 +916,8 @@ static void long_filter_ehigh_3830(int32_t *buffer, int length)
 {
     int i, j;
     int32_t dotprod, sign;
-    int32_t coeffs[8] = { 0 }, delay[8] = { 0 };
+    int32_t delay[8] = { 0 };
+    uint32_t coeffs[8] = { 0 };
 
     for (i = 0; i < length; i++) {
         dotprod = 0;
@@ -928,7 +929,7 @@ static void long_filter_ehigh_3830(int32_t *buffer, int length)
         for (j = 7; j > 0; j--)
             delay[j] = delay[j - 1];
         delay[0] = buffer[i];
-        buffer[i] -= dotprod >> 9;
+        buffer[i] -= (unsigned)(dotprod >> 9);
     }
 }
 
@@ -1037,13 +1038,13 @@ static av_always_inline int predictor_update_3930(APEPredictor *p,
                                                   const int delayA)
 {
     int32_t predictionA, sign;
-    int32_t d0, d1, d2, d3;
+    uint32_t d0, d1, d2, d3;
 
     p->buf[delayA]     = p->lastA[filter];
     d0 = p->buf[delayA    ];
-    d1 = p->buf[delayA    ] - p->buf[delayA - 1];
-    d2 = p->buf[delayA - 1] - p->buf[delayA - 2];
-    d3 = p->buf[delayA - 2] - p->buf[delayA - 3];
+    d1 = p->buf[delayA    ] - (unsigned)p->buf[delayA - 1];
+    d2 = p->buf[delayA - 1] - (unsigned)p->buf[delayA - 2];
+    d3 = p->buf[delayA - 2] - (unsigned)p->buf[delayA - 3];
 
     predictionA = d0 * p->coeffsA[filter][0] +
                   d1 * p->coeffsA[filter][1] +
@@ -1051,13 +1052,13 @@ static av_always_inline int predictor_update_3930(APEPredictor *p,
                   d3 * p->coeffsA[filter][3];
 
     p->lastA[filter] = decoded + (predictionA >> 9);
-    p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
+    p->filterA[filter] = p->lastA[filter] + ((int)(p->filterA[filter] * 31U) >> 5);
 
     sign = APESIGN(decoded);
-    p->coeffsA[filter][0] += ((d0 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][1] += ((d1 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][2] += ((d2 < 0) * 2 - 1) * sign;
-    p->coeffsA[filter][3] += ((d3 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][0] += (((int32_t)d0 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][1] += (((int32_t)d1 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][2] += (((int32_t)d2 < 0) * 2 - 1) * sign;
+    p->coeffsA[filter][3] += (((int32_t)d3 < 0) * 2 - 1) * sign;
 
     return p->filterA[filter];
 }
@@ -1121,7 +1122,7 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,
 
     p->buf[delayA]     = p->lastA[filter];
     p->buf[adaptA]     = APESIGN(p->buf[delayA]);
-    p->buf[delayA - 1] = p->buf[delayA] - p->buf[delayA - 1];
+    p->buf[delayA - 1] = p->buf[delayA] - (unsigned)p->buf[delayA - 1];
     p->buf[adaptA - 1] = APESIGN(p->buf[delayA - 1]);
 
     predictionA = p->buf[delayA    ] * p->coeffsA[filter][0] +
@@ -1130,9 +1131,9 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,
                   p->buf[delayA - 3] * p->coeffsA[filter][3];
 
     /*  Apply a scaled first-order filter compression */
-    p->buf[delayB]     = p->filterA[filter ^ 1] - ((p->filterB[filter] * 31) >> 5);
+    p->buf[delayB]     = p->filterA[filter ^ 1] - ((int)(p->filterB[filter] * 31U) >> 5);
     p->buf[adaptB]     = APESIGN(p->buf[delayB]);
-    p->buf[delayB - 1] = p->buf[delayB] - p->buf[delayB - 1];
+    p->buf[delayB - 1] = p->buf[delayB] - (unsigned)p->buf[delayB - 1];
     p->buf[adaptB - 1] = APESIGN(p->buf[delayB - 1]);
     p->filterB[filter] = p->filterA[filter ^ 1];
 
@@ -1142,8 +1143,8 @@ static av_always_inline int predictor_update_filter(APEPredictor *p,
                   p->buf[delayB - 3] * p->coeffsB[filter][3] +
                   p->buf[delayB - 4] * p->coeffsB[filter][4];
 
-    p->lastA[filter] = decoded + ((predictionA + (predictionB >> 1)) >> 10);
-    p->filterA[filter] = p->lastA[filter] + ((p->filterA[filter] * 31) >> 5);
+    p->lastA[filter] = decoded + ((int)((unsigned)predictionA + (predictionB >> 1)) >> 10);
+    p->filterA[filter] = p->lastA[filter] + ((int)(p->filterA[filter] * 31U) >> 5);
 
     sign = APESIGN(decoded);
     p->coeffsA[filter][0] += p->buf[adaptA    ] * sign;
@@ -1202,14 +1203,14 @@ static void predictor_decode_mono_3950(APEContext *ctx, int count)
         A = *decoded0;
 
         p->buf[YDELAYA] = currentA;
-        p->buf[YDELAYA - 1] = p->buf[YDELAYA] - p->buf[YDELAYA - 1];
+        p->buf[YDELAYA - 1] = p->buf[YDELAYA] - (unsigned)p->buf[YDELAYA - 1];
 
         predictionA = p->buf[YDELAYA    ] * p->coeffsA[0][0] +
                       p->buf[YDELAYA - 1] * p->coeffsA[0][1] +
                       p->buf[YDELAYA - 2] * p->coeffsA[0][2] +
                       p->buf[YDELAYA - 3] * p->coeffsA[0][3];
 
-        currentA = A + (predictionA >> 10);
+        currentA = A + (unsigned)(predictionA >> 10);
 
         p->buf[YADAPTCOEFFSA]     = APESIGN(p->buf[YDELAYA    ]);
         p->buf[YADAPTCOEFFSA - 1] = APESIGN(p->buf[YDELAYA - 1]);
@@ -1229,7 +1230,7 @@ static void predictor_decode_mono_3950(APEContext *ctx, int count)
             p->buf = p->historybuffer;
         }
 
-        p->filterA[0] = currentA + ((p->filterA[0] * 31) >> 5);
+        p->filterA[0] = currentA + (unsigned)((int)(p->filterA[0] * 31U) >> 5);
         *(decoded0++) = p->filterA[0];
     }
 
@@ -1266,8 +1267,8 @@ static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
                                                      f->delay - order,
                                                      f->adaptcoeffs - order,
                                                      order, APESIGN(*data));
-        res = (res + (1 << (fracbits - 1))) >> fracbits;
-        res += *data;
+        res = (int)(res + (1U << (fracbits - 1))) >> fracbits;
+        res += (unsigned)*data;
         *data++ = res;
 
         /* Update the output history */
@@ -1282,10 +1283,10 @@ static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
             /* Version 3.98 and later files */
 
             /* Update the adaption coefficients */
-            absres = FFABS(res);
+            absres = res < 0 ? -(unsigned)res : res;
             if (absres)
                 *f->adaptcoeffs = APESIGN(res) *
-                                  (8 << ((absres > f->avg * 3) + (absres > f->avg * 4 / 3)));
+                                  (8 << ((absres > f->avg * 3LL) + (absres > (f->avg + f->avg / 3))));
                 /* equivalent to the following code
                     if (absres <= f->avg * 4 / 3)
                         *f->adaptcoeffs = APESIGN(res) * 8;
@@ -1297,7 +1298,7 @@ static void do_apply_filter(APEContext *ctx, int version, APEFilter *f,
             else
                 *f->adaptcoeffs = 0;
 
-            f->avg += (absres - f->avg) / 16;
+            f->avg += (int)(absres - (unsigned)f->avg) / 16;
 
             f->adaptcoeffs[-1] >>= 1;
             f->adaptcoeffs[-2] >>= 1;
@@ -1376,7 +1377,7 @@ static void ape_unpack_mono(APEContext *ctx, int count)
 
 static void ape_unpack_stereo(APEContext *ctx, int count)
 {
-    int32_t left, right;
+    unsigned left, right;
     int32_t *decoded0 = ctx->decoded[0];
     int32_t *decoded1 = ctx->decoded[1];
 
@@ -1393,7 +1394,7 @@ static void ape_unpack_stereo(APEContext *ctx, int count)
 
     /* Decorrelate and scale to output depth */
     while (count--) {
-        left = *decoded1 - (*decoded0 / 2);
+        left = *decoded1 - (unsigned)(*decoded0 / 2);
         right = left + *decoded0;
 
         *(decoded0++) = left;
@@ -1451,7 +1452,8 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
         if (s->fileversion >= 3900) {
             if (offset > 3) {
                 av_log(avctx, AV_LOG_ERROR, "Incorrect offset passed\n");
-                s->data = NULL;
+                av_freep(&s->data);
+                s->data_size = 0;
                 return AVERROR_INVALIDDATA;
             }
             if (s->data_end - s->ptr < offset) {
@@ -1499,7 +1501,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
     av_fast_malloc(&s->decoded_buffer, &s->decoded_size, decoded_buffer_size);
     if (!s->decoded_buffer)
         return AVERROR(ENOMEM);
-    memset(s->decoded_buffer, 0, s->decoded_size);
+    memset(s->decoded_buffer, 0, decoded_buffer_size);
     s->decoded[0] = s->decoded_buffer;
     s->decoded[1] = s->decoded_buffer + FFALIGN(blockstodecode, 8);
 
@@ -1527,7 +1529,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
         for (ch = 0; ch < s->channels; ch++) {
             sample8 = (uint8_t *)frame->data[ch];
             for (i = 0; i < blockstodecode; i++)
-                *sample8++ = (s->decoded[ch][i] + 0x80) & 0xff;
+                *sample8++ = (s->decoded[ch][i] + 0x80U) & 0xff;
         }
         break;
     case 16:
@@ -1541,7 +1543,7 @@ static int ape_decode_frame(AVCodecContext *avctx, void *data,
         for (ch = 0; ch < s->channels; ch++) {
             sample24 = (int32_t *)frame->data[ch];
             for (i = 0; i < blockstodecode; i++)
-                *sample24++ = s->decoded[ch][i] << 8;
+                *sample24++ = s->decoded[ch][i] * 256U;
         }
         break;
     }

libavcodec/aptx.c

@@ -480,7 +480,7 @@ static void aptx_update_codeword_history(Channel *channel)
     int32_t cw = ((channel->quantize[0].quantized_sample & 3) << 0) +
                  ((channel->quantize[1].quantized_sample & 2) << 1) +
                  ((channel->quantize[2].quantized_sample & 1) << 3);
-    channel->codeword_history = (cw << 8) + (channel->codeword_history << 4);
+    channel->codeword_history = (cw << 8) + ((unsigned)channel->codeword_history << 4);
 }
 
 static void aptx_generate_dither(Channel *channel)
@@ -492,9 +492,9 @@ static void aptx_generate_dither(Channel *channel)
     aptx_update_codeword_history(channel);
 
     m = (int64_t)5184443 * (channel->codeword_history >> 7);
-    d = (m << 2) + (m >> 22);
+    d = (m * 4) + (m >> 22);
     for (subband = 0; subband < NB_SUBBANDS; subband++)
-        channel->dither[subband] = d << (23 - 5*subband);
+        channel->dither[subband] = (unsigned)d << (23 - 5*subband);
     channel->dither_parity = (d >> 25) & 1;
 }
 
@@ -759,12 +759,12 @@ static void aptx_invert_quantization(InvertQuantize *invert_quantize,
     if (quantized_sample < 0)
         qr = -qr;
 
-    qr = rshift64_clip24(((int64_t)qr<<32) + MUL64(dither, tables->invert_quantize_dither_factors[idx]), 32);
+    qr = rshift64_clip24((qr * (1LL<<32)) + MUL64(dither, tables->invert_quantize_dither_factors[idx]), 32);
     invert_quantize->reconstructed_difference = MUL64(invert_quantize->quantization_factor, qr) >> 19;
 
     /* update factor_select */
     factor_select = 32620 * invert_quantize->factor_select;
-    factor_select = rshift32(factor_select + (tables->quantize_factor_select_offset[idx] << 15), 15);
+    factor_select = rshift32(factor_select + (tables->quantize_factor_select_offset[idx] * (1 << 15)), 15);
     invert_quantize->factor_select = av_clip(factor_select, 0, tables->factor_max);
 
     /* update quantization factor */
@@ -801,7 +801,7 @@ static void aptx_prediction_filtering(Prediction *prediction,
     prediction->previous_reconstructed_sample = reconstructed_sample;
 
     reconstructed_differences = aptx_reconstructed_differences_update(prediction, reconstructed_difference, order);
-    srd0 = FFDIFFSIGN(reconstructed_difference, 0) << 23;
+    srd0 = FFDIFFSIGN(reconstructed_difference, 0) * (1 << 23);
     for (i = 0; i < order; i++) {
         int32_t srd = FF_SIGNBIT(reconstructed_differences[-i-1]) | 1;
         prediction->d_weight[i] -= rshift32(prediction->d_weight[i] - srd*srd0, 8);
@@ -830,7 +830,7 @@ static void aptx_process_subband(InvertQuantize *invert_quantize,
 
     range = 0x100000;
     sw1 = rshift32(-same_sign[1] * prediction->s_weight[1], 1);
-    sw1 = (av_clip(sw1, -range, range) & ~0xF) << 4;
+    sw1 = (av_clip(sw1, -range, range) & ~0xF) * 16;
 
     range = 0x300000;
     weight[0] = 254 * prediction->s_weight[0] + 0x800000*same_sign[0] + sw1;
@@ -989,6 +989,9 @@ static av_cold int aptx_init(AVCodecContext *avctx)
     AptXContext *s = avctx->priv_data;
     int chan, subband;
 
+    if (avctx->channels != 2)
+        return AVERROR_INVALIDDATA;
+
     s->hd = avctx->codec->id == AV_CODEC_ID_APTX_HD;
     s->block_size = s->hd ? 6 : 4;
 
@@ -1044,7 +1047,7 @@ static int aptx_decode_frame(AVCodecContext *avctx, void *data,
         for (channel = 0; channel < NB_CHANNELS; channel++)
             for (sample = 0; sample < 4; sample++)
                 AV_WN32A(&frame->data[channel][4*(opos+sample)],
-                         samples[channel][sample] << 8);
+                         samples[channel][sample] * 256);
     }
 
     *got_frame_ptr = 1;

libavcodec/assdec.c

@@ -31,7 +31,8 @@ static av_cold int ass_decode_init(AVCodecContext *avctx)
     avctx->subtitle_header = av_malloc(avctx->extradata_size + 1);
     if (!avctx->subtitle_header)
         return AVERROR(ENOMEM);
-    memcpy(avctx->subtitle_header, avctx->extradata, avctx->extradata_size);
+    if (avctx->extradata_size)
+        memcpy(avctx->subtitle_header, avctx->extradata, avctx->extradata_size);
     avctx->subtitle_header[avctx->extradata_size] = 0;
     avctx->subtitle_header_size = avctx->extradata_size;
     return 0;

libavcodec/atrac3.c

@@ -964,7 +964,7 @@ static av_cold int atrac3_decode_init(AVCodecContext *avctx)
         return AVERROR_INVALIDDATA;
     }
 
-    if (avctx->block_align >= UINT_MAX / 2)
+    if (avctx->block_align > 1024 || avctx->block_align <= 0)
         return AVERROR(EINVAL);
 
     q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +

libavcodec/atrac3plus.c

@@ -456,6 +456,10 @@ static int decode_channel_wordlen(GetBitContext *gb, Atrac3pChanUnitCtx *ctx,
     } else if (chan->fill_mode == 3) {
         pos = ch_num ? chan->num_coded_vals + chan->split_point
                      : ctx->num_quant_units - chan->split_point;
+        if (pos > FF_ARRAY_ELEMS(chan->qu_wordlen)) {
+            av_log(avctx, AV_LOG_ERROR, "Split point beyond array\n");
+            pos = FF_ARRAY_ELEMS(chan->qu_wordlen);
+        }
         for (i = chan->num_coded_vals; i < pos; i++)
             chan->qu_wordlen[i] = 1;
     }

libavcodec/atrac9dec.c

@@ -0,0 +1,979 @@
+/*
+ * ATRAC9 decoder
+ * Copyright (c) 2018 Rostislav Pehlivanov <atomnuker@gmail.com>
+ *
+ * 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
+ */
+
+#include "internal.h"
+#include "get_bits.h"
+#include "fft.h"
+#include "atrac9tab.h"
+#include "libavutil/lfg.h"
+#include "libavutil/float_dsp.h"
+
+typedef struct ATRAC9ChannelData {
+    int band_ext;
+    int q_unit_cnt;
+    int band_ext_data[4];
+    int32_t scalefactors[31];
+    int32_t scalefactors_prev[31];
+
+    int precision_coarse[30];
+    int precision_fine[30];
+    int precision_mask[30];
+
+    int codebookset[30];
+
+    int32_t q_coeffs_coarse[256];
+    int32_t q_coeffs_fine[256];
+
+    DECLARE_ALIGNED(32, float, coeffs  )[256];
+    DECLARE_ALIGNED(32, float, prev_win)[128];
+} ATRAC9ChannelData;
+
+typedef struct ATRAC9BlockData {
+    ATRAC9ChannelData channel[2];
+
+    /* Base */
+    int band_count;
+    int q_unit_cnt;
+    int q_unit_cnt_prev;
+
+    /* Stereo block only */
+    int stereo_q_unit;
+
+    /* Band extension only */
+    int has_band_ext;
+    int has_band_ext_data;
+    int band_ext_q_unit;
+
+    /* Gradient */
+    int grad_mode;
+    int grad_boundary;
+    int gradient[31];
+
+    /* Stereo */
+    int cpe_base_channel;
+    int is_signs[30];
+
+    int reuseable;
+
+} ATRAC9BlockData;
+
+typedef struct ATRAC9Context {
+    AVCodecContext *avctx;
+    AVFloatDSPContext *fdsp;
+    FFTContext imdct;
+    ATRAC9BlockData block[5];
+    AVLFG lfg;
+
+    /* Set on init */
+    int frame_log2;
+    int avg_frame_size;
+    int frame_count;
+    int samplerate_idx;
+    const ATRAC9BlockConfig *block_config;
+
+    /* Generated on init */
+    VLC sf_vlc[2][8];            /* Signed/unsigned, length */
+    VLC coeff_vlc[2][8][4];      /* Cookbook, precision, cookbook index */
+    uint8_t alloc_curve[48][48];
+    DECLARE_ALIGNED(32, float, imdct_win)[256];
+
+    DECLARE_ALIGNED(32, float, temp)[256];
+} ATRAC9Context;
+
+static inline int parse_gradient(ATRAC9Context *s, ATRAC9BlockData *b,
+                                 GetBitContext *gb)
+{
+    int grad_range[2];
+    int grad_value[2];
+    int values, sign, base;
+    uint8_t *curve;
+    float scale;
+
+    b->grad_mode = get_bits(gb, 2);
+    if (b->grad_mode) {
+        grad_range[0] = get_bits(gb, 5);
+        grad_range[1] = 31;
+        grad_value[0] = get_bits(gb, 5);
+        grad_value[1] = 31;
+    } else {
+        grad_range[0] = get_bits(gb, 6);
+        grad_range[1] = get_bits(gb, 6) + 1;
+        grad_value[0] = get_bits(gb, 5);
+        grad_value[1] = get_bits(gb, 5);
+    }
+    b->grad_boundary = get_bits(gb, 4);
+
+    if (grad_range[0] >= grad_range[1] || grad_range[1] > 31)
+        return AVERROR_INVALIDDATA;
+
+    if (grad_value[0] > 31 || grad_value[1] > 31)
+        return AVERROR_INVALIDDATA;
+
+    if (b->grad_boundary > b->q_unit_cnt)
+        return AVERROR_INVALIDDATA;
+
+    values    = grad_value[1] - grad_value[0];
+    sign      = 1 - 2*(values < 0);
+    base      = grad_value[0] + sign;
+    scale     = (FFABS(values) - 1) / 31.0f;
+    curve     = s->alloc_curve[grad_range[1] - grad_range[0] - 1];
+
+    for (int i = 0; i <= b->q_unit_cnt; i++)
+        b->gradient[i] = grad_value[i >= grad_range[0]];
+
+    for (int i = grad_range[0]; i < grad_range[1]; i++)
+        b->gradient[i] = base + sign*((int)(scale*curve[i - grad_range[0]]));
+
+    return 0;
+}
+
+static inline void calc_precision(ATRAC9Context *s, ATRAC9BlockData *b,
+                                  ATRAC9ChannelData *c)
+{
+    memset(c->precision_mask, 0, sizeof(c->precision_mask));
+    for (int i = 1; i < b->q_unit_cnt; i++) {
+        const int delta = FFABS(c->scalefactors[i] - c->scalefactors[i - 1]) - 1;
+        if (delta > 0) {
+            const int neg = c->scalefactors[i - 1] > c->scalefactors[i];
+            c->precision_mask[i - neg] += FFMIN(delta, 5);
+        }
+    }
+
+    if (b->grad_mode) {
+        for (int i = 0; i < b->q_unit_cnt; i++) {
+            c->precision_coarse[i] = c->scalefactors[i];
+            c->precision_coarse[i] += c->precision_mask[i] - b->gradient[i];
+            if (c->precision_coarse[i] < 0)
+                continue;
+            switch (b->grad_mode) {
+            case 1:
+                c->precision_coarse[i] >>= 1;
+                break;
+            case 2:
+                c->precision_coarse[i] = (3 * c->precision_coarse[i]) >> 3;
+                break;
+            case 3:
+                c->precision_coarse[i] >>= 2;
+                break;
+            }
+        }
+    } else {
+        for (int i = 0; i < b->q_unit_cnt; i++)
+            c->precision_coarse[i] = c->scalefactors[i] - b->gradient[i];
+    }
+
+
+    for (int i = 0; i < b->q_unit_cnt; i++)
+        c->precision_coarse[i] = FFMAX(c->precision_coarse[i], 1);
+
+    for (int i = 0; i < b->grad_boundary; i++)
+        c->precision_coarse[i]++;
+
+    for (int i = 0; i < b->q_unit_cnt; i++) {
+        c->precision_fine[i] = 0;
+        if (c->precision_coarse[i] > 15) {
+            c->precision_fine[i] = FFMIN(c->precision_coarse[i], 30) - 15;
+            c->precision_coarse[i] = 15;
+        }
+    }
+}
+
+static inline int parse_band_ext(ATRAC9Context *s, ATRAC9BlockData *b,
+                                 GetBitContext *gb, int stereo)
+{
+    int ext_band = 0;
+
+    if (b->has_band_ext) {
+        if (b->q_unit_cnt < 13 || b->q_unit_cnt > 20)
+            return AVERROR_INVALIDDATA;
+        ext_band = at9_tab_band_ext_group[b->q_unit_cnt - 13][2];
+        if (stereo) {
+            b->channel[1].band_ext = get_bits(gb, 2);
+            b->channel[1].band_ext = ext_band > 2 ? b->channel[1].band_ext : 4;
+        } else {
+            skip_bits1(gb);
+        }
+    }
+
+    b->has_band_ext_data = get_bits1(gb);
+    if (!b->has_band_ext_data)
+        return 0;
+
+    if (!b->has_band_ext) {
+        skip_bits(gb, 2);
+        skip_bits_long(gb, get_bits(gb, 5));
+        return 0;
+    }
+
+    b->channel[0].band_ext = get_bits(gb, 2);
+    b->channel[0].band_ext = ext_band > 2 ? b->channel[0].band_ext : 4;
+
+    if (!get_bits(gb, 5)) {
+        for (int i = 0; i <= stereo; i++) {
+            ATRAC9ChannelData *c = &b->channel[i];
+            const int count = at9_tab_band_ext_cnt[c->band_ext][ext_band];
+            for (int j = 0; j < count; j++) {
+                int len = at9_tab_band_ext_lengths[c->band_ext][ext_band][j];
+                c->band_ext_data[j] = av_clip_uintp2_c(c->band_ext_data[j], len);
+            }
+        }
+
+        return 0;
+    }
+
+    for (int i = 0; i <= stereo; i++) {
+        ATRAC9ChannelData *c = &b->channel[i];
+        const int count = at9_tab_band_ext_cnt[c->band_ext][ext_band];
+        for (int j = 0; j < count; j++) {
+            int len = at9_tab_band_ext_lengths[c->band_ext][ext_band][j];
+            c->band_ext_data[j] = get_bits(gb, len);
+        }
+    }
+
+    return 0;
+}
+
+static inline int read_scalefactors(ATRAC9Context *s, ATRAC9BlockData *b,
+                                    ATRAC9ChannelData *c, GetBitContext *gb,
+                                    int channel_idx, int first_in_pkt)
+{
+    static const int mode_map[2][4] = { { 0, 1, 2, 3 }, { 0, 2, 3, 4 } };
+    const int mode = mode_map[channel_idx][get_bits(gb, 2)];
+
+    memset(c->scalefactors, 0, sizeof(c->scalefactors));
+
+    if (first_in_pkt && (mode == 4 || ((mode == 3) && !channel_idx))) {
+        av_log(s->avctx, AV_LOG_ERROR, "Invalid scalefactor coding mode!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    switch (mode) {
+    case 0: { /* VLC delta offset */
+        const uint8_t *sf_weights = at9_tab_sf_weights[get_bits(gb, 3)];
+        const int base = get_bits(gb, 5);
+        const int len = get_bits(gb, 2) + 3;
+        const VLC *tab = &s->sf_vlc[0][len];
+
+        c->scalefactors[0] = get_bits(gb, len);
+
+        for (int i = 1; i < b->band_ext_q_unit; i++) {
+            int val = c->scalefactors[i - 1] + get_vlc2(gb, tab->table, 9, 2);
+            c->scalefactors[i] = val & ((1 << len) - 1);
+        }
+
+        for (int i = 0; i < b->band_ext_q_unit; i++)
+            c->scalefactors[i] += base - sf_weights[i];
+
+        break;
+    }
+    case 1: { /* CLC offset */
+        const int len = get_bits(gb, 2) + 2;
+        const int base = len < 5 ? get_bits(gb, 5) : 0;
+        for (int i = 0; i < b->band_ext_q_unit; i++)
+            c->scalefactors[i] = base + get_bits(gb, len);
+        break;
+    }
+    case 2:
+    case 4: { /* VLC dist to baseline */
+        const int *baseline = mode == 4 ? c->scalefactors_prev :
+                              channel_idx ? b->channel[0].scalefactors :
+                              c->scalefactors_prev;
+        const int baseline_len = mode == 4 ? b->q_unit_cnt_prev :
+                                 channel_idx ? b->band_ext_q_unit :
+                                 b->q_unit_cnt_prev;
+
+        const int len = get_bits(gb, 2) + 2;
+        const int unit_cnt = FFMIN(b->band_ext_q_unit, baseline_len);
+        const VLC *tab = &s->sf_vlc[1][len];
+
+        for (int i = 0; i < unit_cnt; i++) {
+            int dist = get_vlc2(gb, tab->table, 9, 2);
+            c->scalefactors[i] = baseline[i] + dist;
+        }
+
+        for (int i = unit_cnt; i < b->band_ext_q_unit; i++)
+            c->scalefactors[i] = get_bits(gb, 5);
+
+        break;
+    }
+    case 3: { /* VLC offset with baseline */
+        const int *baseline = channel_idx ? b->channel[0].scalefactors :
+                              c->scalefactors_prev;
+        const int baseline_len = channel_idx ? b->band_ext_q_unit :
+                                 b->q_unit_cnt_prev;
+
+        const int base = get_bits(gb, 5) - (1 << (5 - 1));
+        const int len = get_bits(gb, 2) + 1;
+        const int unit_cnt = FFMIN(b->band_ext_q_unit, baseline_len);
+        const VLC *tab = &s->sf_vlc[0][len];
+
+        c->scalefactors[0] = get_bits(gb, len);
+
+        for (int i = 1; i < unit_cnt; i++) {
+            int val = c->scalefactors[i - 1] + get_vlc2(gb, tab->table, 9, 2);
+            c->scalefactors[i] = val & ((1 << len) - 1);
+        }
+
+        for (int i = 0; i < unit_cnt; i++)
+            c->scalefactors[i] += base + baseline[i];
+
+        for (int i = unit_cnt; i < b->band_ext_q_unit; i++)
+            c->scalefactors[i] = get_bits(gb, 5);
+        break;
+    }
+    }
+
+    for (int i = 0; i < b->band_ext_q_unit; i++)
+        if (c->scalefactors[i] < 0 || c->scalefactors[i] > 31)
+            return AVERROR_INVALIDDATA;
+
+    memcpy(c->scalefactors_prev, c->scalefactors, sizeof(c->scalefactors));
+
+    return 0;
+}
+
+static inline void calc_codebook_idx(ATRAC9Context *s, ATRAC9BlockData *b,
+                                     ATRAC9ChannelData *c)
+{
+    int avg = 0;
+    const int last_sf = c->scalefactors[c->q_unit_cnt];
+
+    memset(c->codebookset, 0, sizeof(c->codebookset));
+
+    if (c->q_unit_cnt <= 1)
+        return;
+    if (s->samplerate_idx > 7)
+        return;
+
+    c->scalefactors[c->q_unit_cnt] = c->scalefactors[c->q_unit_cnt - 1];
+
+    if (c->q_unit_cnt > 12) {
+        for (int i = 0; i < 12; i++)
+            avg += c->scalefactors[i];
+        avg = (avg + 6) / 12;
+    }
+
+    for (int i = 8; i < c->q_unit_cnt; i++) {
+        const int prev = c->scalefactors[i - 1];
+        const int cur  = c->scalefactors[i    ];
+        const int next = c->scalefactors[i + 1];
+        const int min  = FFMIN(prev, next);
+        if ((cur - min >= 3 || 2*cur - prev - next >= 3))
+            c->codebookset[i] = 1;
+    }
+
+
+    for (int i = 12; i < c->q_unit_cnt; i++) {
+        const int cur = c->scalefactors[i];
+        const int cnd = at9_q_unit_to_coeff_cnt[i] == 16;
+        const int min = FFMIN(c->scalefactors[i + 1], c->scalefactors[i - 1]);
+        if (c->codebookset[i])
+            continue;
+
+        c->codebookset[i] = (((cur - min) >= 2) && (cur >= (avg - cnd)));
+    }
+
+    c->scalefactors[c->q_unit_cnt] = last_sf;
+}
+
+static inline void read_coeffs_coarse(ATRAC9Context *s, ATRAC9BlockData *b,
+                                      ATRAC9ChannelData *c, GetBitContext *gb)
+{
+    const int max_prec = s->samplerate_idx > 7 ? 1 : 7;
+
+    memset(c->q_coeffs_coarse, 0, sizeof(c->q_coeffs_coarse));
+
+    for (int i = 0; i < c->q_unit_cnt; i++) {
+        int *coeffs = &c->q_coeffs_coarse[at9_q_unit_to_coeff_idx[i]];
+        const int bands = at9_q_unit_to_coeff_cnt[i];
+        const int prec = c->precision_coarse[i] + 1;
+
+        if (prec <= max_prec) {
+            const int cb = c->codebookset[i];
+            const int cbi = at9_q_unit_to_codebookidx[i];
+            const VLC *tab = &s->coeff_vlc[cb][prec][cbi];
+            const HuffmanCodebook *huff = &at9_huffman_coeffs[cb][prec][cbi];
+            const int groups = bands >> huff->value_cnt_pow;
+
+            for (int j = 0; j < groups; j++) {
+                uint16_t val = get_vlc2(gb, tab->table, 9, huff->max_bit_size);
+
+                for (int k = 0; k < huff->value_cnt; k++) {
+                    coeffs[k] = sign_extend(val, huff->value_bits);
+                    val >>= huff->value_bits;
+                }
+
+                coeffs += huff->value_cnt;
+            }
+        } else {
+            for (int j = 0; j < bands; j++)
+                coeffs[j] = sign_extend(get_bits(gb, prec), prec);
+        }
+    }
+}
+
+static inline void read_coeffs_fine(ATRAC9Context *s, ATRAC9BlockData *b,
+                                    ATRAC9ChannelData *c, GetBitContext *gb)
+{
+    memset(c->q_coeffs_fine, 0, sizeof(c->q_coeffs_fine));
+
+    for (int i = 0; i < c->q_unit_cnt; i++) {
+        const int start = at9_q_unit_to_coeff_idx[i + 0];
+        const int end   = at9_q_unit_to_coeff_idx[i + 1];
+        const int len   = c->precision_fine[i] + 1;
+
+        if (c->precision_fine[i] <= 0)
+            continue;
+
+        for (int j = start; j < end; j++)
+            c->q_coeffs_fine[j] = sign_extend(get_bits(gb, len), len);
+    }
+}
+
+static inline void dequantize(ATRAC9Context *s, ATRAC9BlockData *b,
+                              ATRAC9ChannelData *c)
+{
+    memset(c->coeffs, 0, sizeof(c->coeffs));
+
+    for (int i = 0; i < c->q_unit_cnt; i++) {
+        const int start = at9_q_unit_to_coeff_idx[i + 0];
+        const int end   = at9_q_unit_to_coeff_idx[i + 1];
+
+        const float coarse_c = at9_quant_step_coarse[c->precision_coarse[i]];
+        const float fine_c   = at9_quant_step_fine[c->precision_fine[i]];
+
+        for (int j = start; j < end; j++) {
+            const float vc = c->q_coeffs_coarse[j] * coarse_c;
+            const float vf = c->q_coeffs_fine[j]   * fine_c;
+            c->coeffs[j] = vc + vf;
+        }
+    }
+}
+
+static inline void apply_intensity_stereo(ATRAC9Context *s, ATRAC9BlockData *b,
+                                          const int stereo)
+{
+    float *src = b->channel[ b->cpe_base_channel].coeffs;
+    float *dst = b->channel[!b->cpe_base_channel].coeffs;
+
+    if (!stereo)
+        return;
+
+    if (b->q_unit_cnt <= b->stereo_q_unit)
+        return;
+
+    for (int i = b->stereo_q_unit; i < b->q_unit_cnt; i++) {
+        const int sign  = b->is_signs[i];
+        const int start = at9_q_unit_to_coeff_idx[i + 0];
+        const int end   = at9_q_unit_to_coeff_idx[i + 1];
+        for (int j = start; j < end; j++)
+            dst[j] = sign*src[j];
+    }
+}
+
+static inline void apply_scalefactors(ATRAC9Context *s, ATRAC9BlockData *b,
+                                      const int stereo)
+{
+    for (int i = 0; i <= stereo; i++) {
+        float *coeffs = b->channel[i].coeffs;
+        for (int j = 0; j < b->q_unit_cnt; j++) {
+            const int start = at9_q_unit_to_coeff_idx[j + 0];
+            const int end   = at9_q_unit_to_coeff_idx[j + 1];
+            const int scalefactor = b->channel[i].scalefactors[j];
+            const float scale = at9_scalefactor_c[scalefactor];
+            for (int k = start; k < end; k++)
+                coeffs[k] *= scale;
+        }
+    }
+}
+
+static inline void fill_with_noise(ATRAC9Context *s, ATRAC9ChannelData *c,
+                                   int start, int count)
+{
+    float maxval = 0.0f;
+    for (int i = 0; i < count; i += 2) {
+        double tmp[2];
+        av_bmg_get(&s->lfg, tmp);
+        c->coeffs[start + i + 0] = tmp[0];
+        c->coeffs[start + i + 1] = tmp[1];
+        maxval = FFMAX(FFMAX(FFABS(tmp[0]), FFABS(tmp[1])), maxval);
+    }
+    /* Normalize */
+    for (int i = 0; i < count; i++)
+        c->coeffs[start + i] /= maxval;
+}
+
+static inline void scale_band_ext_coeffs(ATRAC9ChannelData *c, float sf[6],
+                                         const int s_unit, const int e_unit)
+{
+    for (int i = s_unit; i < e_unit; i++) {
+        const int start = at9_q_unit_to_coeff_idx[i + 0];
+        const int end   = at9_q_unit_to_coeff_idx[i + 1];
+        for (int j = start; j < end; j++)
+            c->coeffs[j] *= sf[i - s_unit];
+    }
+}
+
+static inline void apply_band_extension(ATRAC9Context *s, ATRAC9BlockData *b,
+                                       const int stereo)
+{
+    const int g_units[4] = { /* A, B, C, total units */
+        b->q_unit_cnt,
+        at9_tab_band_ext_group[b->q_unit_cnt - 13][0],
+        at9_tab_band_ext_group[b->q_unit_cnt - 13][1],
+        FFMAX(g_units[2], 22),
+    };
+
+    const int g_bins[4] = { /* A, B, C, total bins */
+        at9_q_unit_to_coeff_idx[g_units[0]],
+        at9_q_unit_to_coeff_idx[g_units[1]],
+        at9_q_unit_to_coeff_idx[g_units[2]],
+        at9_q_unit_to_coeff_idx[g_units[3]],
+    };
+
+    for (int ch = 0; ch <= stereo; ch++) {
+        ATRAC9ChannelData *c = &b->channel[ch];
+
+        /* Mirror the spectrum */
+        for (int i = 0; i < 3; i++)
+            for (int j = 0; j < (g_bins[i + 1] - g_bins[i + 0]); j++)
+                c->coeffs[g_bins[i] + j] = c->coeffs[g_bins[i] - j - 1];
+
+        switch (c->band_ext) {
+        case 0: {
+            float sf[6] = { 0.0f };
+            const int l = g_units[3] - g_units[0] - 1;
+            const int n_start = at9_q_unit_to_coeff_idx[g_units[3] - 1];
+            const int n_cnt   = at9_q_unit_to_coeff_cnt[g_units[3] - 1];
+            switch (at9_tab_band_ext_group[b->q_unit_cnt - 13][2]) {
+            case 3:
+                sf[0] = at9_band_ext_scales_m0[0][0][c->band_ext_data[0]];
+                sf[1] = at9_band_ext_scales_m0[0][1][c->band_ext_data[0]];
+                sf[2] = at9_band_ext_scales_m0[0][2][c->band_ext_data[1]];
+                sf[3] = at9_band_ext_scales_m0[0][3][c->band_ext_data[2]];
+                sf[4] = at9_band_ext_scales_m0[0][4][c->band_ext_data[3]];
+                break;
+            case 4:
+                sf[0] = at9_band_ext_scales_m0[1][0][c->band_ext_data[0]];
+                sf[1] = at9_band_ext_scales_m0[1][1][c->band_ext_data[0]];
+                sf[2] = at9_band_ext_scales_m0[1][2][c->band_ext_data[1]];
+                sf[3] = at9_band_ext_scales_m0[1][3][c->band_ext_data[2]];
+                sf[4] = at9_band_ext_scales_m0[1][4][c->band_ext_data[3]];
+                break;
+            case 5:
+                sf[0] = at9_band_ext_scales_m0[2][0][c->band_ext_data[0]];
+                sf[1] = at9_band_ext_scales_m0[2][1][c->band_ext_data[1]];
+                sf[2] = at9_band_ext_scales_m0[2][2][c->band_ext_data[1]];
+                break;
+            }
+
+            sf[l] = at9_scalefactor_c[c->scalefactors[g_units[0]]];
+
+            fill_with_noise(s, c, n_start, n_cnt);
+            scale_band_ext_coeffs(c, sf, g_units[0], g_units[3]);
+            break;
+        }
+        case 1: {
+            float sf[6];
+            for (int i = g_units[0]; i < g_units[3]; i++)
+                sf[i - g_units[0]] = at9_scalefactor_c[c->scalefactors[i]];
+
+            fill_with_noise(s, c, g_bins[0], g_bins[3] - g_bins[0]);
+            scale_band_ext_coeffs(c, sf, g_units[0], g_units[3]);
+            break;
+        }
+        case 2: {
+            const float g_sf[2] = {
+                at9_band_ext_scales_m2[c->band_ext_data[0]],
+                at9_band_ext_scales_m2[c->band_ext_data[1]],
+            };
+
+            for (int i = 0; i < 2; i++)
+                for (int j = g_bins[i + 0]; j < g_bins[i + 1]; j++)
+                    c->coeffs[j] *= g_sf[i];
+            break;
+        }
+        case 3: {
+            float scale = at9_band_ext_scales_m3[c->band_ext_data[0]][0];
+            float rate  = at9_band_ext_scales_m3[c->band_ext_data[1]][1];
+            rate = pow(2, rate);
+            for (int i = g_bins[0]; i < g_bins[3]; i++) {
+                scale *= rate;
+                c->coeffs[i] *= scale;
+            }
+            break;
+        }
+        case 4: {
+            const float m = at9_band_ext_scales_m4[c->band_ext_data[0]];
+            const float g_sf[3] = { 0.7079468f*m, 0.5011902f*m, 0.3548279f*m };
+
+            for (int i = 0; i < 3; i++)
+                for (int j = g_bins[i + 0]; j < g_bins[i + 1]; j++)
+                    c->coeffs[j] *= g_sf[i];
+            break;
+        }
+        }
+    }
+}
+
+static int atrac9_decode_block(ATRAC9Context *s, GetBitContext *gb,
+                               ATRAC9BlockData *b, AVFrame *frame,
+                               int frame_idx, int block_idx)
+{
+    const int first_in_pkt = !get_bits1(gb);
+    const int reuse_params =  get_bits1(gb);
+    const int stereo = s->block_config->type[block_idx] == ATRAC9_BLOCK_TYPE_CPE;
+
+    if (s->block_config->type[block_idx] == ATRAC9_BLOCK_TYPE_LFE) {
+        ATRAC9ChannelData *c = &b->channel[0];
+        const int precision = reuse_params ? 8 : 4;
+        c->q_unit_cnt = b->q_unit_cnt = 2;
+
+        memset(c->scalefactors, 0, sizeof(c->scalefactors));
+        memset(c->q_coeffs_fine, 0, sizeof(c->q_coeffs_fine));
+        memset(c->q_coeffs_coarse, 0, sizeof(c->q_coeffs_coarse));
+
+        for (int i = 0; i < b->q_unit_cnt; i++) {
+            c->scalefactors[i] = get_bits(gb, 5);
+            c->precision_coarse[i] = precision;
+            c->precision_fine[i] = 0;
+        }
+
+        for (int i = 0; i < c->q_unit_cnt; i++) {
+            const int start = at9_q_unit_to_coeff_idx[i + 0];
+            const int end   = at9_q_unit_to_coeff_idx[i + 1];
+            for (int j = start; j < end; j++)
+                c->q_coeffs_coarse[j] = get_bits(gb, c->precision_coarse[i] + 1);
+        }
+
+        dequantize        (s, b, c);
+        apply_scalefactors(s, b, 0);
+
+        goto imdct;
+    }
+
+    if (first_in_pkt && reuse_params) {
+        av_log(s->avctx, AV_LOG_ERROR, "Invalid block flags!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* Band parameters */
+    if (!reuse_params) {
+        int stereo_band, ext_band;
+        const int min_band_count = s->samplerate_idx > 7 ? 1 : 3;
+        b->reuseable = 0;
+        b->band_count = get_bits(gb, 4) + min_band_count;
+        b->q_unit_cnt = at9_tab_band_q_unit_map[b->band_count];
+
+        b->band_ext_q_unit = b->stereo_q_unit = b->q_unit_cnt;
+
+        if (b->band_count > at9_tab_sri_max_bands[s->samplerate_idx]) {
+            av_log(s->avctx, AV_LOG_ERROR, "Invalid band count %i!\n",
+                   b->band_count);
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (stereo) {
+            stereo_band = get_bits(gb, 4) + min_band_count;
+            if (stereo_band > b->band_count) {
+                av_log(s->avctx, AV_LOG_ERROR, "Invalid stereo band %i!\n",
+                       stereo_band);
+                return AVERROR_INVALIDDATA;
+            }
+            b->stereo_q_unit = at9_tab_band_q_unit_map[stereo_band];
+        }
+
+        b->has_band_ext = get_bits1(gb);
+        if (b->has_band_ext) {
+            ext_band = get_bits(gb, 4) + min_band_count;
+            if (ext_band < b->band_count) {
+                av_log(s->avctx, AV_LOG_ERROR, "Invalid extension band %i!\n",
+                       ext_band);
+                return AVERROR_INVALIDDATA;
+            }
+            b->band_ext_q_unit = at9_tab_band_q_unit_map[ext_band];
+        }
+        b->reuseable = 1;
+    }
+    if (!b->reuseable) {
+        av_log(s->avctx, AV_LOG_ERROR, "invalid block reused!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* Calculate bit alloc gradient */
+    if (parse_gradient(s, b, gb))
+        return AVERROR_INVALIDDATA;
+
+    /* IS data */
+    b->cpe_base_channel = 0;
+    if (stereo) {
+        b->cpe_base_channel = get_bits1(gb);
+        if (get_bits1(gb)) {
+            for (int i = b->stereo_q_unit; i < b->q_unit_cnt; i++)
+                b->is_signs[i] = 1 - 2*get_bits1(gb);
+        } else {
+            for (int i = 0; i < FF_ARRAY_ELEMS(b->is_signs); i++)
+                b->is_signs[i] = 1;
+        }
+    }
+
+    /* Band extension */
+    if (parse_band_ext(s, b, gb, stereo))
+        return AVERROR_INVALIDDATA;
+
+    /* Scalefactors */
+    for (int i = 0; i <= stereo; i++) {
+        ATRAC9ChannelData *c = &b->channel[i];
+        c->q_unit_cnt = i == b->cpe_base_channel ? b->q_unit_cnt :
+                                                   b->stereo_q_unit;
+        if (read_scalefactors(s, b, c, gb, i, first_in_pkt))
+            return AVERROR_INVALIDDATA;
+
+        calc_precision    (s, b, c);
+        calc_codebook_idx (s, b, c);
+        read_coeffs_coarse(s, b, c, gb);
+        read_coeffs_fine  (s, b, c, gb);
+        dequantize        (s, b, c);
+    }
+
+    b->q_unit_cnt_prev = b->has_band_ext ? b->band_ext_q_unit : b->q_unit_cnt;
+
+    apply_intensity_stereo(s, b, stereo);
+    apply_scalefactors    (s, b, stereo);
+
+    if (b->has_band_ext && b->has_band_ext_data)
+        apply_band_extension  (s, b, stereo);
+
+imdct:
+    for (int i = 0; i <= stereo; i++) {
+        ATRAC9ChannelData *c = &b->channel[i];
+        const int dst_idx = s->block_config->plane_map[block_idx][i];
+        const int wsize = 1 << s->frame_log2;
+        const ptrdiff_t offset = wsize*frame_idx*sizeof(float);
+        float *dst = (float *)(frame->extended_data[dst_idx] + offset);
+
+        s->imdct.imdct_half(&s->imdct, s->temp, c->coeffs);
+        s->fdsp->vector_fmul_window(dst, c->prev_win, s->temp,
+                                    s->imdct_win, wsize >> 1);
+        memcpy(c->prev_win, s->temp + (wsize >> 1), sizeof(float)*wsize >> 1);
+    }
+
+    return 0;
+}
+
+static int atrac9_decode_frame(AVCodecContext *avctx, void *data,
+                               int *got_frame_ptr, AVPacket *avpkt)
+{
+    int ret;
+    GetBitContext gb;
+    AVFrame *frame = data;
+    ATRAC9Context *s = avctx->priv_data;
+    const int frames = FFMIN(avpkt->size / s->avg_frame_size, s->frame_count);
+
+    frame->nb_samples = (1 << s->frame_log2) * frames;
+    ret = ff_get_buffer(avctx, frame, 0);
+    if (ret < 0)
+        return ret;
+
+    init_get_bits8(&gb, avpkt->data, avpkt->size);
+
+    for (int i = 0; i < frames; i++) {
+        for (int j = 0; j < s->block_config->count; j++) {
+            ret = atrac9_decode_block(s, &gb, &s->block[j], frame, i, j);
+            if (ret)
+                return ret;
+            align_get_bits(&gb);
+        }
+    }
+
+    *got_frame_ptr = 1;
+
+    return avctx->block_align;
+}
+
+static void atrac9_decode_flush(AVCodecContext *avctx)
+{
+    ATRAC9Context *s = avctx->priv_data;
+
+    for (int j = 0; j < s->block_config->count; j++) {
+        ATRAC9BlockData *b = &s->block[j];
+        const int stereo = s->block_config->type[j] == ATRAC9_BLOCK_TYPE_CPE;
+        for (int i = 0; i <= stereo; i++) {
+            ATRAC9ChannelData *c = &b->channel[i];
+            memset(c->prev_win, 0, sizeof(c->prev_win));
+        }
+    }
+}
+
+static av_cold int atrac9_decode_close(AVCodecContext *avctx)
+{
+    ATRAC9Context *s = avctx->priv_data;
+
+    for (int i = 1; i < 7; i++)
+        ff_free_vlc(&s->sf_vlc[0][i]);
+    for (int i = 2; i < 6; i++)
+        ff_free_vlc(&s->sf_vlc[1][i]);
+    for (int i = 0; i < 2; i++)
+        for (int j = 0; j < 8; j++)
+            for (int k = 0; k < 4; k++)
+                ff_free_vlc(&s->coeff_vlc[i][j][k]);
+
+    ff_mdct_end(&s->imdct);
+    av_free(s->fdsp);
+
+    return 0;
+}
+
+static av_cold int atrac9_decode_init(AVCodecContext *avctx)
+{
+    GetBitContext gb;
+    ATRAC9Context *s = avctx->priv_data;
+    int version, block_config_idx, superframe_idx, alloc_c_len;
+
+    s->avctx = avctx;
+
+    av_lfg_init(&s->lfg, 0xFBADF00D);
+
+    if (avctx->block_align <= 0) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid block align\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    if (avctx->extradata_size != 12) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid extradata length!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    version = AV_RL32(avctx->extradata);
+    if (version > 2) {
+        av_log(avctx, AV_LOG_ERROR, "Unsupported version (%i)!\n", version);
+        return AVERROR_INVALIDDATA;
+    }
+
+    init_get_bits8(&gb, avctx->extradata + 4, avctx->extradata_size);
+
+    if (get_bits(&gb, 8) != 0xFE) {
+        av_log(avctx, AV_LOG_ERROR, "Incorrect magic byte!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->samplerate_idx = get_bits(&gb, 4);
+    avctx->sample_rate = at9_tab_samplerates[s->samplerate_idx];
+
+    block_config_idx = get_bits(&gb, 3);
+    if (block_config_idx > 5) {
+        av_log(avctx, AV_LOG_ERROR, "Incorrect block config!\n");
+        return AVERROR_INVALIDDATA;
+    }
+    s->block_config = &at9_block_layout[block_config_idx];
+
+    avctx->channel_layout = s->block_config->channel_layout;
+    avctx->channels       = av_get_channel_layout_nb_channels(avctx->channel_layout);
+    avctx->sample_fmt     = AV_SAMPLE_FMT_FLTP;
+
+    if (get_bits1(&gb)) {
+        av_log(avctx, AV_LOG_ERROR, "Incorrect verification bit!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* Average frame size in bytes */
+    s->avg_frame_size = get_bits(&gb, 11) + 1;
+
+    superframe_idx = get_bits(&gb, 2);
+    if (superframe_idx & 1) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid superframe index!\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    s->frame_count = 1 << superframe_idx;
+    s->frame_log2  = at9_tab_sri_frame_log2[s->samplerate_idx];
+
+    if (ff_mdct_init(&s->imdct, s->frame_log2 + 1, 1, 1.0f / 32768.0f))
+        return AVERROR(ENOMEM);
+
+    s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
+    if (!s->fdsp)
+        return AVERROR(ENOMEM);
+
+    /* iMDCT window */
+    for (int i = 0; i < (1 << s->frame_log2); i++) {
+        const int   len  = 1 << s->frame_log2;
+        const float sidx = (      i + 0.5f) / len;
+        const float eidx = (len - i - 0.5f) / len;
+        const float s_c  = sinf(sidx*M_PI - M_PI_2)*0.5f + 0.5f;
+        const float e_c  = sinf(eidx*M_PI - M_PI_2)*0.5f + 0.5f;
+        s->imdct_win[i]  = s_c / ((s_c * s_c) + (e_c * e_c));
+    }
+
+    /* Allocation curve */
+    alloc_c_len = FF_ARRAY_ELEMS(at9_tab_b_dist);
+    for (int i = 1; i <= alloc_c_len; i++)
+        for (int j = 0; j < i; j++)
+            s->alloc_curve[i - 1][j] = at9_tab_b_dist[(j * alloc_c_len) / i];
+
+    /* Unsigned scalefactor VLCs */
+    for (int i = 1; i < 7; i++) {
+        const HuffmanCodebook *hf = &at9_huffman_sf_unsigned[i];
+
+        init_vlc(&s->sf_vlc[0][i], 9, hf->size, hf->bits, 1, 1, hf->codes,
+                 2, 2, 0);
+    }
+
+    /* Signed scalefactor VLCs */
+    for (int i = 2; i < 6; i++) {
+        const HuffmanCodebook *hf = &at9_huffman_sf_signed[i];
+
+        int nums = hf->size;
+        int16_t sym[32];
+        for (int j = 0; j < nums; j++)
+            sym[j] = sign_extend(j, hf->value_bits);
+
+        ff_init_vlc_sparse(&s->sf_vlc[1][i], 9, hf->size, hf->bits, 1, 1,
+                           hf->codes, 2, 2, sym, sizeof(*sym), sizeof(*sym), 0);
+    }
+
+    /* Coefficient VLCs */
+    for (int i = 0; i < 2; i++) {
+        for (int j = 0; j < 8; j++) {
+            for (int k = 0; k < 4; k++) {
+                const HuffmanCodebook *hf = &at9_huffman_coeffs[i][j][k];
+                init_vlc(&s->coeff_vlc[i][j][k], 9, hf->size, hf->bits, 1, 1,
+                         hf->codes, 2, 2, 0);
+            }
+        }
+    }
+
+    return 0;
+}
+
+AVCodec ff_atrac9_decoder = {
+    .name           = "atrac9",
+    .long_name      = NULL_IF_CONFIG_SMALL("ATRAC9 (Adaptive TRansform Acoustic Coding 9)"),
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_ATRAC9,
+    .priv_data_size = sizeof(ATRAC9Context),
+    .init           = atrac9_decode_init,
+    .close          = atrac9_decode_close,
+    .decode         = atrac9_decode_frame,
+    .flush          = atrac9_decode_flush,
+    .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
+    .capabilities   = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
+};

libavcodec/audiodsp.c

@@ -79,7 +79,7 @@ static void vector_clipf_c(float *dst, const float *src, int len,
 static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2,
                                      int order)
 {
-    int res = 0;
+    unsigned res = 0;
 
     while (order--)
         res += *v1++ **v2++;

libavcodec/av1.h

@@ -0,0 +1,130 @@
+/*
+ * 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
+ * AV1 common definitions
+ */
+
+#ifndef AVCODEC_AV1_H
+#define AVCODEC_AV1_H
+
+// OBU types (section 6.2.2).
+typedef enum {
+    // 0 reserved.
+    AV1_OBU_SEQUENCE_HEADER        = 1,
+    AV1_OBU_TEMPORAL_DELIMITER     = 2,
+    AV1_OBU_FRAME_HEADER           = 3,
+    AV1_OBU_TILE_GROUP             = 4,
+    AV1_OBU_METADATA               = 5,
+    AV1_OBU_FRAME                  = 6,
+    AV1_OBU_REDUNDANT_FRAME_HEADER = 7,
+    AV1_OBU_TILE_LIST              = 8,
+    // 9-14 reserved.
+    AV1_OBU_PADDING                = 15,
+} AV1_OBU_Type;
+
+// Metadata types (section 6.7.1).
+enum {
+    AV1_METADATA_TYPE_HDR_CLL     = 1,
+    AV1_METADATA_TYPE_HDR_MDCV    = 2,
+    AV1_METADATA_TYPE_SCALABILITY = 3,
+    AV1_METADATA_TYPE_ITUT_T35    = 4,
+    AV1_METADATA_TYPE_TIMECODE    = 5,
+};
+
+// Frame types (section 6.8.2).
+enum {
+    AV1_FRAME_KEY        = 0,
+    AV1_FRAME_INTER      = 1,
+    AV1_FRAME_INTRA_ONLY = 2,
+    AV1_FRAME_SWITCH     = 3,
+};
+
+// Reference frames (section 6.10.24).
+enum {
+    AV1_REF_FRAME_INTRA   = 0,
+    AV1_REF_FRAME_LAST    = 1,
+    AV1_REF_FRAME_LAST2   = 2,
+    AV1_REF_FRAME_LAST3   = 3,
+    AV1_REF_FRAME_GOLDEN  = 4,
+    AV1_REF_FRAME_BWDREF  = 5,
+    AV1_REF_FRAME_ALTREF2 = 6,
+    AV1_REF_FRAME_ALTREF  = 7,
+};
+
+// Constants (section 3).
+enum {
+    AV1_MAX_OPERATING_POINTS = 32,
+
+    AV1_MAX_SB_SIZE    = 128,
+    AV1_MI_SIZE        = 4,
+
+    AV1_MAX_TILE_WIDTH = 4096,
+    AV1_MAX_TILE_AREA  = 4096 * 2304,
+    AV1_MAX_TILE_ROWS  = 64,
+    AV1_MAX_TILE_COLS  = 64,
+
+    AV1_NUM_REF_FRAMES       = 8,
+    AV1_REFS_PER_FRAME       = 7,
+    AV1_TOTAL_REFS_PER_FRAME = 8,
+    AV1_PRIMARY_REF_NONE     = 7,
+
+    AV1_MAX_SEGMENTS = 8,
+    AV1_SEG_LVL_MAX  = 8,
+
+    AV1_SEG_LVL_ALT_Q      = 0,
+    AV1_SEG_LVL_ALT_LF_Y_V = 1,
+    AV1_SEG_LVL_REF_FRAME  = 5,
+    AV1_SEG_LVL_SKIP       = 6,
+    AV1_SEG_LVL_GLOBAL_MV  = 7,
+
+    AV1_SELECT_SCREEN_CONTENT_TOOLS = 2,
+    AV1_SELECT_INTEGER_MV           = 2,
+
+    AV1_SUPERRES_NUM       = 8,
+    AV1_SUPERRES_DENOM_MIN = 9,
+
+    AV1_INTERPOLATION_FILTER_SWITCHABLE = 4,
+
+    AV1_GM_ABS_ALPHA_BITS       = 12,
+    AV1_GM_ALPHA_PREC_BITS      = 15,
+    AV1_GM_ABS_TRANS_ONLY_BITS  = 9,
+    AV1_GM_TRANS_ONLY_PREC_BITS = 3,
+    AV1_GM_ABS_TRANS_BITS       = 12,
+    AV1_GM_TRANS_PREC_BITS      = 6,
+    AV1_WARPEDMODEL_PREC_BITS   = 16,
+
+    AV1_WARP_MODEL_IDENTITY    = 0,
+    AV1_WARP_MODEL_TRANSLATION = 1,
+    AV1_WARP_MODEL_ROTZOOM     = 2,
+    AV1_WARP_MODEL_AFFINE      = 3,
+};
+
+
+// The main colour configuration information uses the same ISO/IEC 23001-8
+// (H.273) enums as FFmpeg does, so separate definitions are not required.
+
+// Chroma sample position.
+enum {
+    AV1_CSP_UNKNOWN   = 0,
+    AV1_CSP_VERTICAL  = 1, // -> AVCHROMA_LOC_LEFT.
+    AV1_CSP_COLOCATED = 2, // -> AVCHROMA_LOC_TOPLEFT.
+};
+
+#endif /* AVCODEC_AV1_H */

libavcodec/av1_metadata_bsf.c

@@ -0,0 +1,298 @@
+/*
+ * 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
+ */
+
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+
+#include "bsf.h"
+#include "cbs.h"
+#include "cbs_av1.h"
+
+enum {
+    PASS,
+    INSERT,
+    REMOVE,
+};
+
+typedef struct AV1MetadataContext {
+    const AVClass *class;
+
+    CodedBitstreamContext *cbc;
+    CodedBitstreamFragment access_unit;
+
+    int td;
+
+    int color_primaries;
+    int transfer_characteristics;
+    int matrix_coefficients;
+
+    int color_range;
+    int chroma_sample_position;
+
+    AVRational tick_rate;
+    int num_ticks_per_picture;
+} AV1MetadataContext;
+
+
+static int av1_metadata_update_sequence_header(AVBSFContext *bsf,
+                                               AV1RawSequenceHeader *seq)
+{
+    AV1MetadataContext *ctx = bsf->priv_data;
+    AV1RawColorConfig  *clc = &seq->color_config;
+    AV1RawTimingInfo   *tim = &seq->timing_info;
+
+    if (ctx->color_primaries >= 0          ||
+        ctx->transfer_characteristics >= 0 ||
+        ctx->matrix_coefficients >= 0) {
+        if (!clc->color_description_present_flag) {
+            clc->color_description_present_flag = 1;
+            clc->color_primaries          = AVCOL_PRI_UNSPECIFIED;
+            clc->transfer_characteristics = AVCOL_TRC_UNSPECIFIED;
+            clc->matrix_coefficients      = AVCOL_SPC_UNSPECIFIED;
+        }
+
+        if (ctx->color_primaries >= 0)
+            clc->color_primaries = ctx->color_primaries;
+        if (ctx->transfer_characteristics >= 0)
+            clc->transfer_characteristics = ctx->transfer_characteristics;
+        if (ctx->matrix_coefficients >= 0)
+            clc->matrix_coefficients = ctx->matrix_coefficients;
+    }
+
+    if (ctx->color_range >= 0) {
+        if (clc->color_primaries          == AVCOL_PRI_BT709        &&
+            clc->transfer_characteristics == AVCOL_TRC_IEC61966_2_1 &&
+            clc->matrix_coefficients      == AVCOL_SPC_RGB) {
+            av_log(bsf, AV_LOG_WARNING, "Warning: color_range cannot be set "
+                   "on RGB streams encoded in BT.709 sRGB.\n");
+        } else {
+            clc->color_range = ctx->color_range;
+        }
+    }
+
+    if (ctx->chroma_sample_position >= 0) {
+        if (clc->mono_chrome || !clc->subsampling_x || !clc->subsampling_y) {
+            av_log(bsf, AV_LOG_WARNING, "Warning: chroma_sample_position "
+                   "can only be set for 4:2:0 streams.\n");
+        } else {
+            clc->chroma_sample_position = ctx->chroma_sample_position;
+        }
+    }
+
+    if (ctx->tick_rate.num && ctx->tick_rate.den) {
+        int num, den;
+
+        av_reduce(&num, &den, ctx->tick_rate.num, ctx->tick_rate.den,
+                  UINT32_MAX > INT_MAX ? UINT32_MAX : INT_MAX);
+
+        tim->time_scale                = num;
+        tim->num_units_in_display_tick = den;
+        seq->timing_info_present_flag  = 1;
+
+        if (ctx->num_ticks_per_picture > 0) {
+            tim->equal_picture_interval = 1;
+            tim->num_ticks_per_picture_minus_1 =
+                ctx->num_ticks_per_picture - 1;
+        }
+    }
+
+    return 0;
+}
+
+static int av1_metadata_filter(AVBSFContext *bsf, AVPacket *out)
+{
+    AV1MetadataContext *ctx = bsf->priv_data;
+    AVPacket *in = NULL;
+    CodedBitstreamFragment *frag = &ctx->access_unit;
+    AV1RawOBU td, *obu;
+    int err, i;
+
+    err = ff_bsf_get_packet(bsf, &in);
+    if (err < 0)
+        return err;
+
+    err = ff_cbs_read_packet(ctx->cbc, frag, in);
+    if (err < 0) {
+        av_log(bsf, AV_LOG_ERROR, "Failed to read packet.\n");
+        goto fail;
+    }
+
+    for (i = 0; i < frag->nb_units; i++) {
+        if (frag->units[i].type == AV1_OBU_SEQUENCE_HEADER) {
+            obu = frag->units[i].content;
+            err = av1_metadata_update_sequence_header(bsf, &obu->obu.sequence_header);
+            if (err < 0)
+                goto fail;
+        }
+    }
+
+    // If a Temporal Delimiter is present, it must be the first OBU.
+    if (frag->units[0].type == AV1_OBU_TEMPORAL_DELIMITER) {
+        if (ctx->td == REMOVE)
+            ff_cbs_delete_unit(ctx->cbc, frag, 0);
+    } else if (ctx->td == INSERT) {
+        td = (AV1RawOBU) {
+            .header.obu_type = AV1_OBU_TEMPORAL_DELIMITER,
+        };
+
+        err = ff_cbs_insert_unit_content(ctx->cbc, frag, 0, AV1_OBU_TEMPORAL_DELIMITER,
+                                         &td, NULL);
+        if (err < 0) {
+            av_log(bsf, AV_LOG_ERROR, "Failed to insert Temporal Delimiter.\n");
+            goto fail;
+        }
+    }
+
+    err = ff_cbs_write_packet(ctx->cbc, out, frag);
+    if (err < 0) {
+        av_log(bsf, AV_LOG_ERROR, "Failed to write packet.\n");
+        goto fail;
+    }
+
+    err = av_packet_copy_props(out, in);
+    if (err < 0)
+        goto fail;
+
+    err = 0;
+fail:
+    ff_cbs_fragment_uninit(ctx->cbc, frag);
+
+    if (err < 0)
+        av_packet_unref(out);
+    av_packet_free(&in);
+
+    return err;
+}
+
+static int av1_metadata_init(AVBSFContext *bsf)
+{
+    AV1MetadataContext *ctx = bsf->priv_data;
+    CodedBitstreamFragment *frag = &ctx->access_unit;
+    AV1RawOBU *obu;
+    int err, i;
+
+    err = ff_cbs_init(&ctx->cbc, AV_CODEC_ID_AV1, bsf);
+    if (err < 0)
+        return err;
+
+    if (bsf->par_in->extradata) {
+        err = ff_cbs_read_extradata(ctx->cbc, frag, bsf->par_in);
+        if (err < 0) {
+            av_log(bsf, AV_LOG_ERROR, "Failed to read extradata.\n");
+            goto fail;
+        }
+
+        for (i = 0; i < frag->nb_units; i++) {
+            if (frag->units[i].type == AV1_OBU_SEQUENCE_HEADER) {
+                obu = frag->units[i].content;
+                err = av1_metadata_update_sequence_header(bsf, &obu->obu.sequence_header);
+                if (err < 0)
+                    goto fail;
+            }
+        }
+
+        err = ff_cbs_write_extradata(ctx->cbc, bsf->par_out, frag);
+        if (err < 0) {
+            av_log(bsf, AV_LOG_ERROR, "Failed to write extradata.\n");
+            goto fail;
+        }
+    }
+
+    err = 0;
+fail:
+    ff_cbs_fragment_uninit(ctx->cbc, frag);
+    return err;
+}
+
+static void av1_metadata_close(AVBSFContext *bsf)
+{
+    AV1MetadataContext *ctx = bsf->priv_data;
+    ff_cbs_close(&ctx->cbc);
+}
+
+#define OFFSET(x) offsetof(AV1MetadataContext, x)
+#define FLAGS (AV_OPT_FLAG_VIDEO_PARAM|AV_OPT_FLAG_BSF_PARAM)
+static const AVOption av1_metadata_options[] = {
+    { "td", "Temporal Delimiter OBU",
+        OFFSET(td), AV_OPT_TYPE_INT,
+        { .i64 = PASS }, PASS, REMOVE, FLAGS, "td" },
+    { "pass",   NULL, 0, AV_OPT_TYPE_CONST,
+        { .i64 = PASS   }, .flags = FLAGS, .unit = "td" },
+    { "insert", NULL, 0, AV_OPT_TYPE_CONST,
+        { .i64 = INSERT }, .flags = FLAGS, .unit = "td" },
+    { "remove", NULL, 0, AV_OPT_TYPE_CONST,
+        { .i64 = REMOVE }, .flags = FLAGS, .unit = "td" },
+
+    { "color_primaries", "Set color primaries (section 6.4.2)",
+        OFFSET(color_primaries), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, 255, FLAGS },
+    { "transfer_characteristics", "Set transfer characteristics (section 6.4.2)",
+        OFFSET(transfer_characteristics), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, 255, FLAGS },
+    { "matrix_coefficients", "Set matrix coefficients (section 6.4.2)",
+        OFFSET(matrix_coefficients), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, 255, FLAGS },
+
+    { "color_range", "Set color range flag (section 6.4.2)",
+        OFFSET(color_range), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, 1, FLAGS, "cr" },
+    { "tv", "TV (limited) range", 0, AV_OPT_TYPE_CONST,
+        { .i64 = 0 }, .flags = FLAGS, .unit = "cr" },
+    { "pc", "PC (full) range",    0, AV_OPT_TYPE_CONST,
+        { .i64 = 1 }, .flags = FLAGS, .unit = "cr" },
+
+    { "chroma_sample_position", "Set chroma sample position (section 6.4.2)",
+        OFFSET(chroma_sample_position), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, 3, FLAGS, "csp" },
+    { "unknown",   "Unknown chroma sample position",  0, AV_OPT_TYPE_CONST,
+        { .i64 = AV1_CSP_UNKNOWN },   .flags = FLAGS, .unit = "csp" },
+    { "vertical",  "Left chroma sample position",     0, AV_OPT_TYPE_CONST,
+        { .i64 = AV1_CSP_VERTICAL },  .flags = FLAGS, .unit = "csp" },
+    { "colocated", "Top-left chroma sample position", 0, AV_OPT_TYPE_CONST,
+        { .i64 = AV1_CSP_COLOCATED }, .flags = FLAGS, .unit = "csp" },
+
+    { "tick_rate", "Set display tick rate (num_units_in_display_tick / time_scale)",
+        OFFSET(tick_rate), AV_OPT_TYPE_RATIONAL,
+        { .dbl = 0.0 }, 0, UINT_MAX, FLAGS },
+    { "num_ticks_per_picture", "Set display ticks per picture for CFR streams",
+        OFFSET(num_ticks_per_picture), AV_OPT_TYPE_INT,
+        { .i64 = -1 }, -1, INT_MAX, FLAGS },
+
+    { NULL }
+};
+
+static const AVClass av1_metadata_class = {
+    .class_name = "av1_metadata_bsf",
+    .item_name  = av_default_item_name,
+    .option     = av1_metadata_options,
+    .version    = LIBAVUTIL_VERSION_INT,
+};
+
+static const enum AVCodecID av1_metadata_codec_ids[] = {
+    AV_CODEC_ID_AV1, AV_CODEC_ID_NONE,
+};
+
+const AVBitStreamFilter ff_av1_metadata_bsf = {
+    .name           = "av1_metadata",
+    .priv_data_size = sizeof(AV1MetadataContext),
+    .priv_class     = &av1_metadata_class,
+    .init           = &av1_metadata_init,
+    .close          = &av1_metadata_close,
+    .filter         = &av1_metadata_filter,
+    .codec_ids      = av1_metadata_codec_ids,
+};

libavcodec/av1_parse.c

@@ -0,0 +1,107 @@
+/*
+ * AV1 common parsing code
+ *
+ * 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
+ */
+
+#include "config.h"
+
+#include "libavutil/mem.h"
+
+#include "av1.h"
+#include "av1_parse.h"
+#include "bytestream.h"
+
+int ff_av1_extract_obu(AV1OBU *obu, const uint8_t *buf, int length, void *logctx)
+{
+    int64_t obu_size;
+    int start_pos, type, temporal_id, spatial_id;
+    int len;
+
+    len = parse_obu_header(buf, length, &obu_size, &start_pos,
+                           &type, &temporal_id, &spatial_id);
+    if (len < 0)
+        return len;
+
+    obu->type        = type;
+    obu->temporal_id = temporal_id;
+    obu->spatial_id  = spatial_id;
+
+    obu->data     = buf + start_pos;
+    obu->size     = obu_size;
+    obu->raw_data = buf;
+    obu->raw_size = len;
+
+    av_log(logctx, AV_LOG_DEBUG,
+           "obu_type: %d, temporal_id: %d, spatial_id: %d, payload size: %d\n",
+           obu->type, obu->temporal_id, obu->spatial_id, obu->size);
+
+    return len;
+}
+
+int ff_av1_packet_split(AV1Packet *pkt, const uint8_t *buf, int length, void *logctx)
+{
+    GetByteContext bc;
+    int ret, consumed;
+
+    bytestream2_init(&bc, buf, length);
+    pkt->nb_obus = 0;
+
+    while (bytestream2_get_bytes_left(&bc) > 0) {
+        AV1OBU *obu;
+
+        if (pkt->obus_allocated < pkt->nb_obus + 1) {
+            int new_size = pkt->obus_allocated + 1;
+            AV1OBU *tmp = av_realloc_array(pkt->obus, new_size, sizeof(*tmp));
+            if (!tmp)
+                return AVERROR(ENOMEM);
+
+            pkt->obus = tmp;
+            memset(pkt->obus + pkt->obus_allocated, 0,
+                   (new_size - pkt->obus_allocated) * sizeof(*tmp));
+            pkt->obus_allocated = new_size;
+        }
+        obu = &pkt->obus[pkt->nb_obus];
+
+        consumed = ff_av1_extract_obu(obu, bc.buffer, bytestream2_get_bytes_left(&bc), logctx);
+        if (consumed < 0)
+            return consumed;
+
+        bytestream2_skip(&bc, consumed);
+
+        obu->size_bits = get_obu_bit_length(obu->data, obu->size, obu->type);
+
+        if (obu->size_bits < 0 || (!obu->size_bits && obu->type != AV1_OBU_TEMPORAL_DELIMITER)) {
+            av_log(logctx, AV_LOG_ERROR, "Invalid OBU of type %d, skipping.\n", obu->type);
+            continue;
+        }
+
+        pkt->nb_obus++;
+
+        ret = init_get_bits(&obu->gb, obu->data, obu->size_bits);
+        if (ret < 0)
+            return ret;
+    }
+
+    return 0;
+}
+
+void ff_av1_packet_uninit(AV1Packet *pkt)
+{
+    av_freep(&pkt->obus);
+    pkt->obus_allocated = 0;
+}

libavcodec/av1_parse.h

@@ -0,0 +1,174 @@
+/*
+ * AV1 common parsing code
+ *
+ * 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
+ */
+
+#ifndef AVCODEC_AV1_PARSE_H
+#define AVCODEC_AV1_PARSE_H
+
+#include <stdint.h>
+
+#include "av1.h"
+#include "avcodec.h"
+#include "get_bits.h"
+
+typedef struct AV1OBU {
+    /** Size of payload */
+    int size;
+    const uint8_t *data;
+
+    /**
+     * Size, in bits, of just the data, excluding the trailing_one_bit and
+     * any trailing padding.
+     */
+    int size_bits;
+
+    /** Size of entire OBU, including header */
+    int raw_size;
+    const uint8_t *raw_data;
+
+    /** GetBitContext initialized to the start of the payload */
+    GetBitContext gb;
+
+    int type;
+
+    int temporal_id;
+    int spatial_id;
+} AV1OBU;
+
+/** An input packet split into OBUs */
+typedef struct AV1Packet {
+    AV1OBU *obus;
+    int nb_obus;
+    int obus_allocated;
+} AV1Packet;
+
+/**
+ * Extract an OBU from a raw bitstream.
+ *
+ * @note This function does not copy or store any bitstream data. All
+ *       the pointers in the AV1OBU structure will be valid as long
+ *       as the input buffer also is.
+ */
+int ff_av1_extract_obu(AV1OBU *obu, const uint8_t *buf, int length,
+                       void *logctx);
+
+/**
+ * Split an input packet into OBUs.
+ *
+ * @note This function does not copy or store any bitstream data. All
+ *       the pointers in the AV1Packet structure will be valid as
+ *       long as the input buffer also is.
+ */
+int ff_av1_packet_split(AV1Packet *pkt, const uint8_t *buf, int length,
+                        void *logctx);
+
+/**
+ * Free all the allocated memory in the packet.
+ */
+void ff_av1_packet_uninit(AV1Packet *pkt);
+
+static inline int64_t leb128(GetBitContext *gb) {
+    int64_t ret = 0;
+    int i;
+
+    for (i = 0; i < 8; i++) {
+        int byte = get_bits(gb, 8);
+        ret |= (int64_t)(byte & 0x7f) << (i * 7);
+        if (!(byte & 0x80))
+            break;
+    }
+    return ret;
+}
+
+static inline int parse_obu_header(const uint8_t *buf, int buf_size,
+                                   int64_t *obu_size, int *start_pos, int *type,
+                                   int *temporal_id, int *spatial_id)
+{
+    GetBitContext gb;
+    int ret, extension_flag, has_size_flag;
+    int64_t size;
+
+    ret = init_get_bits8(&gb, buf, FFMIN(buf_size, 2 + 8)); // OBU header fields + max leb128 length
+    if (ret < 0)
+        return ret;
+
+    if (get_bits1(&gb) != 0) // obu_forbidden_bit
+        return AVERROR_INVALIDDATA;
+
+    *type      = get_bits(&gb, 4);
+    extension_flag = get_bits1(&gb);
+    has_size_flag  = get_bits1(&gb);
+    skip_bits1(&gb); // obu_reserved_1bit
+
+    if (extension_flag) {
+        *temporal_id = get_bits(&gb, 3);
+        *spatial_id  = get_bits(&gb, 2);
+        skip_bits(&gb, 3); // extension_header_reserved_3bits
+    } else {
+        *temporal_id = *spatial_id = 0;
+    }
+
+    *obu_size  = has_size_flag ? leb128(&gb)
+                               : buf_size - 1 - extension_flag;
+
+    if (get_bits_left(&gb) < 0)
+        return AVERROR_INVALIDDATA;
+
+    *start_pos = get_bits_count(&gb) / 8;
+
+    size = *obu_size + *start_pos;
+
+    if (size > buf_size)
+        return AVERROR_INVALIDDATA;
+
+    return size;
+}
+
+static inline int get_obu_bit_length(const uint8_t *buf, int size, int type)
+{
+    int v;
+
+    /* There are no trailing bits on these */
+    if (type == AV1_OBU_TILE_GROUP || type == AV1_OBU_FRAME) {
+        if (size > INT_MAX / 8)
+            return AVERROR(ERANGE);
+        else
+            return size * 8;
+    }
+
+    while (size > 0 && buf[size - 1] == 0)
+        size--;
+
+    if (!size)
+        return 0;
+
+    v = buf[size - 1];
+
+    if (size > INT_MAX / 8)
+        return AVERROR(ERANGE);
+    size *= 8;
+
+    /* Remove the trailing_one_bit and following trailing zeros */
+    if (v)
+        size -= ff_ctz(v) + 1;
+
+    return size;
+}
+
+#endif /* AVCODEC_AV1_PARSE_H */

libavcodec/av1_parser.c

@@ -0,0 +1,230 @@
+/*
+ * AV1 parser
+ *
+ * Copyright (C) 2018 James Almer <jamrial@gmail.com>
+ *
+ * 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
+ */
+
+#include "av1_parse.h"
+#include "cbs.h"
+#include "cbs_av1.h"
+#include "parser.h"
+
+typedef struct AV1ParseContext {
+    CodedBitstreamContext *cbc;
+    CodedBitstreamFragment temporal_unit;
+    int parsed_extradata;
+} AV1ParseContext;
+
+static const enum AVPixelFormat pix_fmts_8bit[2][2] = {
+    { AV_PIX_FMT_YUV444P, AV_PIX_FMT_NONE },
+    { AV_PIX_FMT_YUV422P, AV_PIX_FMT_YUV420P },
+};
+static const enum AVPixelFormat pix_fmts_10bit[2][2] = {
+    { AV_PIX_FMT_YUV444P10, AV_PIX_FMT_NONE },
+    { AV_PIX_FMT_YUV422P10, AV_PIX_FMT_YUV420P10 },
+};
+static const enum AVPixelFormat pix_fmts_12bit[2][2] = {
+    { AV_PIX_FMT_YUV444P12, AV_PIX_FMT_NONE },
+    { AV_PIX_FMT_YUV422P12, AV_PIX_FMT_YUV420P12 },
+};
+
+static int av1_parser_parse(AVCodecParserContext *ctx,
+                            AVCodecContext *avctx,
+                            const uint8_t **out_data, int *out_size,
+                            const uint8_t *data, int size)
+{
+    AV1ParseContext *s = ctx->priv_data;
+    CodedBitstreamFragment *td = &s->temporal_unit;
+    CodedBitstreamAV1Context *av1 = s->cbc->priv_data;
+    int ret;
+
+    *out_data = data;
+    *out_size = size;
+
+    ctx->key_frame         = -1;
+    ctx->pict_type         = AV_PICTURE_TYPE_NONE;
+    ctx->picture_structure = AV_PICTURE_STRUCTURE_UNKNOWN;
+
+    s->cbc->log_ctx = avctx;
+
+    if (avctx->extradata_size && !s->parsed_extradata) {
+        s->parsed_extradata = 1;
+
+        ret = ff_cbs_read(s->cbc, td, avctx->extradata, avctx->extradata_size);
+        if (ret < 0) {
+            av_log(avctx, AV_LOG_WARNING, "Failed to parse extradata.\n");
+        }
+
+        ff_cbs_fragment_uninit(s->cbc, td);
+    }
+
+    ret = ff_cbs_read(s->cbc, td, data, size);
+    if (ret < 0) {
+        av_log(avctx, AV_LOG_ERROR, "Failed to parse temporal unit.\n");
+        goto end;
+    }
+
+    if (!av1->sequence_header) {
+        av_log(avctx, AV_LOG_ERROR, "No sequence header available\n");
+        goto end;
+    }
+
+    for (int i = 0; i < td->nb_units; i++) {
+        CodedBitstreamUnit *unit = &td->units[i];
+        AV1RawOBU *obu = unit->content;
+        AV1RawSequenceHeader *seq = av1->sequence_header;
+        AV1RawColorConfig *color = &seq->color_config;
+        AV1RawFrameHeader *frame;
+        int frame_type;
+
+        if (unit->type == AV1_OBU_FRAME)
+            frame = &obu->obu.frame.header;
+        else if (unit->type == AV1_OBU_FRAME_HEADER)
+            frame = &obu->obu.frame_header;
+        else
+            continue;
+
+        if (obu->header.spatial_id > 0)
+            continue;
+
+        if (frame->show_existing_frame) {
+            AV1ReferenceFrameState *ref = &av1->ref[frame->frame_to_show_map_idx];
+
+            if (!ref->valid) {
+                av_log(avctx, AV_LOG_ERROR, "Invalid reference frame\n");
+                goto end;
+            }
+
+            ctx->width  = ref->frame_width;
+            ctx->height = ref->frame_height;
+            frame_type  = ref->frame_type;
+
+            ctx->key_frame = 0;
+        } else if (!frame->show_frame) {
+            continue;
+        } else {
+            ctx->width  = av1->frame_width;
+            ctx->height = av1->frame_height;
+            frame_type  = frame->frame_type;
+
+            ctx->key_frame = frame_type == AV1_FRAME_KEY;
+        }
+
+        avctx->profile = seq->seq_profile;
+        avctx->level   = seq->seq_level_idx[0];
+
+        switch (frame_type) {
+        case AV1_FRAME_KEY:
+        case AV1_FRAME_INTRA_ONLY:
+            ctx->pict_type = AV_PICTURE_TYPE_I;
+            break;
+        case AV1_FRAME_INTER:
+            ctx->pict_type = AV_PICTURE_TYPE_P;
+            break;
+        case AV1_FRAME_SWITCH:
+            ctx->pict_type = AV_PICTURE_TYPE_SP;
+            break;
+        }
+        ctx->picture_structure = AV_PICTURE_STRUCTURE_FRAME;
+
+        switch (av1->bit_depth) {
+        case 8:
+            ctx->format = color->mono_chrome ? AV_PIX_FMT_GRAY8
+                                             : pix_fmts_8bit [color->subsampling_x][color->subsampling_y];
+            break;
+        case 10:
+            ctx->format = color->mono_chrome ? AV_PIX_FMT_GRAY10
+                                             : pix_fmts_10bit[color->subsampling_x][color->subsampling_y];
+            break;
+        case 12:
+            ctx->format = color->mono_chrome ? AV_PIX_FMT_GRAY12
+                                             : pix_fmts_12bit[color->subsampling_x][color->subsampling_y];
+            break;
+        }
+        av_assert2(ctx->format != AV_PIX_FMT_NONE);
+    }
+
+end:
+    ff_cbs_fragment_uninit(s->cbc, td);
+
+    s->cbc->log_ctx = NULL;
+
+    return size;
+}
+
+static const CodedBitstreamUnitType decompose_unit_types[] = {
+    AV1_OBU_TEMPORAL_DELIMITER,
+    AV1_OBU_SEQUENCE_HEADER,
+    AV1_OBU_FRAME_HEADER,
+    AV1_OBU_TILE_GROUP,
+    AV1_OBU_FRAME,
+};
+
+static av_cold int av1_parser_init(AVCodecParserContext *ctx)
+{
+    AV1ParseContext *s = ctx->priv_data;
+    int ret;
+
+    ret = ff_cbs_init(&s->cbc, AV_CODEC_ID_AV1, NULL);
+    if (ret < 0)
+        return ret;
+
+    s->cbc->decompose_unit_types    = (CodedBitstreamUnitType *)decompose_unit_types;
+    s->cbc->nb_decompose_unit_types = FF_ARRAY_ELEMS(decompose_unit_types);
+
+    return 0;
+}
+
+static void av1_parser_close(AVCodecParserContext *ctx)
+{
+    AV1ParseContext *s = ctx->priv_data;
+
+    ff_cbs_close(&s->cbc);
+}
+
+static int av1_parser_split(AVCodecContext *avctx,
+                            const uint8_t *buf, int buf_size)
+{
+    AV1OBU obu;
+    const uint8_t *ptr = buf, *end = buf + buf_size;
+
+    while (ptr < end) {
+        int len = ff_av1_extract_obu(&obu, ptr, buf_size, avctx);
+        if (len < 0)
+            break;
+
+        if (obu.type == AV1_OBU_FRAME_HEADER ||
+            obu.type == AV1_OBU_FRAME) {
+            return ptr - buf;
+        }
+        ptr      += len;
+        buf_size -= len;
+    }
+
+    return 0;
+}
+
+AVCodecParser ff_av1_parser = {
+    .codec_ids      = { AV_CODEC_ID_AV1 },
+    .priv_data_size = sizeof(AV1ParseContext),
+    .parser_init    = av1_parser_init,
+    .parser_close   = av1_parser_close,
+    .parser_parse   = av1_parser_parse,
+    .split          = av1_parser_split,
+};

libavcodec/avcodec.h

@@ -409,6 +409,7 @@ enum AVCodecID {
     AV_CODEC_ID_DXV,
     AV_CODEC_ID_SCREENPRESSO,
     AV_CODEC_ID_RSCC,
+    AV_CODEC_ID_AVS2,
 
     AV_CODEC_ID_Y41P = 0x8000,
     AV_CODEC_ID_AVRP,
@@ -446,6 +447,11 @@ enum AVCodecID {
     AV_CODEC_ID_SVG,
     AV_CODEC_ID_GDV,
     AV_CODEC_ID_FITS,
+    AV_CODEC_ID_IMM4,
+    AV_CODEC_ID_PROSUMER,
+    AV_CODEC_ID_MWSC,
+    AV_CODEC_ID_WCMV,
+    AV_CODEC_ID_RASC,
 
     /* various PCM "codecs" */
     AV_CODEC_ID_FIRST_AUDIO = 0x10000,     ///< A dummy id pointing at the start of audio codecs
@@ -485,6 +491,7 @@ enum AVCodecID {
     AV_CODEC_ID_PCM_S64BE,
     AV_CODEC_ID_PCM_F16LE,
     AV_CODEC_ID_PCM_F24LE,
+    AV_CODEC_ID_PCM_VIDC,
 
     /* various ADPCM codecs */
     AV_CODEC_ID_ADPCM_IMA_QT = 0x11000,
@@ -637,6 +644,7 @@ enum AVCodecID {
     AV_CODEC_ID_APTX,
     AV_CODEC_ID_APTX_HD,
     AV_CODEC_ID_SBC,
+    AV_CODEC_ID_ATRAC9,
 
     /* subtitle codecs */
     AV_CODEC_ID_FIRST_SUBTITLE = 0x17000,          ///< A dummy ID pointing at the start of subtitle codecs.
@@ -665,6 +673,7 @@ enum AVCodecID {
     AV_CODEC_ID_PJS,
     AV_CODEC_ID_ASS,
     AV_CODEC_ID_HDMV_TEXT_SUBTITLE,
+    AV_CODEC_ID_TTML,
 
     /* other specific kind of codecs (generally used for attachments) */
     AV_CODEC_ID_FIRST_UNKNOWN = 0x18000,           ///< A dummy ID pointing at the start of various fake codecs.
@@ -1312,7 +1321,7 @@ enum AVPacketSideDataType {
     AV_PKT_DATA_METADATA_UPDATE,
 
     /**
-     * MPEGTS stream ID, this is required to pass the stream ID
+     * MPEGTS stream ID as uint8_t, this is required to pass the stream ID
      * information from the demuxer to the corresponding muxer.
      */
     AV_PKT_DATA_MPEGTS_STREAM_ID,
@@ -1357,6 +1366,12 @@ enum AVPacketSideDataType {
      */
     AV_PKT_DATA_ENCRYPTION_INFO,
 
+    /**
+     * Active Format Description data consisting of a single byte as specified
+     * in ETSI TS 101 154 using AVActiveFormatDescription enum.
+     */
+    AV_PKT_DATA_AFD,
+
     /**
      * The number of side data types.
      * This is not part of the public API/ABI in the sense that it may
@@ -1612,6 +1627,7 @@ typedef struct AVCodecContext {
      * The allocated memory should be AV_INPUT_BUFFER_PADDING_SIZE bytes larger
      * than extradata_size to avoid problems if it is read with the bitstream reader.
      * The bytewise contents of extradata must not depend on the architecture or CPU endianness.
+     * Must be allocated with the av_malloc() family of functions.
      * - encoding: Set/allocated/freed by libavcodec.
      * - decoding: Set/allocated/freed by user.
      */
@@ -5766,6 +5782,7 @@ typedef struct AVBitStreamFilter {
     int (*init)(AVBSFContext *ctx);
     int (*filter)(AVBSFContext *ctx, AVPacket *pkt);
     void (*close)(AVBSFContext *ctx);
+    void (*flush)(AVBSFContext *ctx);
 } AVBitStreamFilter;
 
 #if FF_API_OLD_BSF
@@ -5892,6 +5909,11 @@ int av_bsf_send_packet(AVBSFContext *ctx, AVPacket *pkt);
  */
 int av_bsf_receive_packet(AVBSFContext *ctx, AVPacket *pkt);
 
+/**
+ * Reset the internal bitstream filter state / flush internal buffers.
+ */
+void av_bsf_flush(AVBSFContext *ctx);
+
 /**
  * Free a bitstream filter context and everything associated with it; write NULL
  * into the supplied pointer.

libavcodec/avdct.c

@@ -100,7 +100,7 @@ int avcodec_dct_init(AVDCT *dsp)
 
 #if CONFIG_IDCTDSP
     {
-        IDCTDSPContext idsp;
+        IDCTDSPContext idsp = {0};
         ff_idctdsp_init(&idsp, avctx);
         COPY(idsp, idct);
         COPY(idsp, idct_permutation);

libavcodec/avpacket.c

@@ -375,6 +375,9 @@ const char *av_packet_side_data_name(enum AVPacketSideDataType type)
     case AV_PKT_DATA_DISPLAYMATRIX:              return "Display Matrix";
     case AV_PKT_DATA_STEREO3D:                   return "Stereo 3D";
     case AV_PKT_DATA_AUDIO_SERVICE_TYPE:         return "Audio Service Type";
+    case AV_PKT_DATA_QUALITY_STATS:              return "Quality stats";
+    case AV_PKT_DATA_FALLBACK_TRACK:             return "Fallback track";
+    case AV_PKT_DATA_CPB_PROPERTIES:             return "CPB properties";
     case AV_PKT_DATA_SKIP_SAMPLES:               return "Skip Samples";
     case AV_PKT_DATA_JP_DUALMONO:                return "JP Dual Mono";
     case AV_PKT_DATA_STRINGS_METADATA:           return "Strings Metadata";
@@ -388,6 +391,9 @@ const char *av_packet_side_data_name(enum AVPacketSideDataType type)
     case AV_PKT_DATA_CONTENT_LIGHT_LEVEL:        return "Content light level metadata";
     case AV_PKT_DATA_SPHERICAL:                  return "Spherical Mapping";
     case AV_PKT_DATA_A53_CC:                     return "A53 Closed Captions";
+    case AV_PKT_DATA_ENCRYPTION_INIT_INFO:       return "Encryption initialization data";
+    case AV_PKT_DATA_ENCRYPTION_INFO:            return "Encryption info";
+    case AV_PKT_DATA_AFD:                        return "Active Format Description data";
     }
     return NULL;
 }
@@ -574,10 +580,10 @@ FF_ENABLE_DEPRECATION_WARNINGS
     dst->side_data            = NULL;
     dst->side_data_elems      = 0;
     for (i = 0; i < src->side_data_elems; i++) {
-         enum AVPacketSideDataType type = src->side_data[i].type;
-         int size          = src->side_data[i].size;
-         uint8_t *src_data = src->side_data[i].data;
-         uint8_t *dst_data = av_packet_new_side_data(dst, type, size);
+        enum AVPacketSideDataType type = src->side_data[i].type;
+        int size          = src->side_data[i].size;
+        uint8_t *src_data = src->side_data[i].data;
+        uint8_t *dst_data = av_packet_new_side_data(dst, type, size);
 
         if (!dst_data) {
             av_packet_free_side_data(dst);

libavcodec/avs2_parser.c

@@ -0,0 +1,95 @@
+/*
+ * AVS2-P2/IEEE1857.4 video parser.
+ * Copyright (c) 2018  Huiwen Ren <hwrenx@gmail.com>
+ *
+ * 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
+ */
+
+#include "parser.h"
+
+#define SLICE_MAX_START_CODE    0x000001af
+
+#define ISPIC(x)  ((x) == 0xB3 || (x) == 0xB6)
+#define ISUNIT(x) ((x) == 0xB0 || (x) == 0xB1 || (x) == 0xB2 || ISPIC(x))
+
+static int avs2_find_frame_end(ParseContext *pc, const uint8_t *buf, int buf_size)
+{
+    int pic_found  = pc->frame_start_found;
+    uint32_t state = pc->state;
+    int cur = 0;
+
+    if (!pic_found) {
+        for (; cur < buf_size; ++cur) {
+            state = (state<<8) | buf[cur];
+            if (ISUNIT(buf[cur])){
+                ++cur;
+                pic_found = 1;
+                break;
+            }
+        }
+    }
+
+    if (pic_found) {
+        if (!buf_size)
+            return END_NOT_FOUND;
+        for (; cur < buf_size; ++cur) {
+            state = (state << 8) | buf[cur];
+            if ((state & 0xFFFFFF00) == 0x100 && state > SLICE_MAX_START_CODE) {
+                pc->frame_start_found = 0;
+                pc->state = -1;
+                return cur - 3;
+            }
+        }
+    }
+
+    pc->frame_start_found = pic_found;
+    pc->state = state;
+
+    return END_NOT_FOUND;
+}
+
+static int avs2_parse(AVCodecParserContext *s, AVCodecContext *avctx,
+                      const uint8_t **poutbuf, int *poutbuf_size,
+                      const uint8_t *buf, int buf_size)
+{
+    ParseContext *pc = s->priv_data;
+    int next;
+
+    if (s->flags & PARSER_FLAG_COMPLETE_FRAMES)  {
+        next = buf_size;
+    } else {
+        next = avs2_find_frame_end(pc, buf, buf_size);
+        if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
+            *poutbuf = NULL;
+            *poutbuf_size = 0;
+            return buf_size;
+        }
+    }
+
+    *poutbuf = buf;
+    *poutbuf_size = buf_size;
+
+    return next;
+}
+
+AVCodecParser ff_avs2_parser = {
+    .codec_ids      = { AV_CODEC_ID_AVS2 },
+    .priv_data_size = sizeof(ParseContext),
+    .parser_parse   = avs2_parse,
+    .parser_close   = ff_parse_close,
+    .split          = ff_mpeg4video_split,
+};

libavcodec/bethsoftvideo.c

@@ -109,6 +109,11 @@ static int bethsoftvid_decode_frame(AVCodecContext *avctx,
             if(yoffset >= avctx->height)
                 return AVERROR_INVALIDDATA;
             dst += vid->frame->linesize[0] * yoffset;
+        case VIDEO_P_FRAME:
+        case VIDEO_I_FRAME:
+            break;
+        default:
+            return AVERROR_INVALIDDATA;
     }
 
     // main code