Update for 2.8.19

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>

.gitattributes

@@ -0,0 +1 @@
+*.pnm -diff -text

.gitignore

@@ -28,6 +28,7 @@
 /ffserver
 /config.*
 /coverage.info
+/avversion.h
 /doc/*.1
 /doc/*.3
 /doc/*.html
@@ -36,9 +37,11 @@
 /doc/avoptions_codec.texi
 /doc/avoptions_format.texi
 /doc/doxy/html/
+/doc/examples/avio_dir_cmd
 /doc/examples/avio_reading
 /doc/examples/decoding_encoding
 /doc/examples/demuxing_decoding
+/doc/examples/extract_mvs
 /doc/examples/filter_audio
 /doc/examples/filtering_audio
 /doc/examples/filtering_video
@@ -60,8 +63,11 @@
 /libavutil/ffversion.h
 /tests/audiogen
 /tests/base64
+/tests/checkasm/checkasm
 /tests/data/
+/tests/pixfmts.mak
 /tests/rotozoom
+/tests/test_copy.ffmeta
 /tests/tiny_psnr
 /tests/tiny_ssim
 /tests/videogen
@@ -80,6 +86,7 @@
 /tools/pktdumper
 /tools/probetest
 /tools/qt-faststart
+/tools/sidxindex
 /tools/trasher
 /tools/seek_print
 /tools/uncoded_frame

LICENSE.md

@@ -1,67 +1,76 @@
 #FFmpeg:
 
 Most files in FFmpeg are under the GNU Lesser General Public License version 2.1
-or later (LGPL v2.1+). Read the file COPYING.LGPLv2.1 for details. Some other
+or later (LGPL v2.1+). Read the file `COPYING.LGPLv2.1` for details. Some other
 files have MIT/X11/BSD-style licenses. In combination the LGPL v2.1+ applies to
 FFmpeg.
 
 Some optional parts of FFmpeg are licensed under the GNU General Public License
-version 2 or later (GPL v2+). See the file COPYING.GPLv2 for details. None of
-these parts are used by default, you have to explicitly pass --enable-gpl to
+version 2 or later (GPL v2+). See the file `COPYING.GPLv2` for details. None of
+these parts are used by default, you have to explicitly pass `--enable-gpl` to
 configure to activate them. In this case, FFmpeg's license changes to GPL v2+.
 
 Specifically, the GPL parts of FFmpeg are:
 
 - libpostproc
-- libmpcodecs
 - optional x86 optimizations in the files
-  libavcodec/x86/idct_mmx.c
+  - `libavcodec/x86/flac_dsp_gpl.asm`
+  - `libavcodec/x86/idct_mmx.c`
+  - `libavfilter/x86/vf_removegrain.asm`
 - libutvideo encoding/decoding wrappers in
-  libavcodec/libutvideo*.cpp
-- the X11 grabber in libavdevice/x11grab.c
+  `libavcodec/libutvideo*.cpp`
+- the X11 grabber in `libavdevice/x11grab.c`
 - the swresample test app in
-  libswresample/swresample-test.c
-- the texi2pod.pl tool
+  `libswresample/swresample-test.c`
+- the `texi2pod.pl` tool
 - the following filters in libavfilter:
-    - f_ebur128.c
-    - vf_blackframe.c
-    - vf_boxblur.c
-    - vf_colormatrix.c
-    - vf_cropdetect.c
-    - vf_decimate.c
-    - vf_delogo.c
-    - vf_geq.c
-    - vf_histeq.c
-    - vf_hqdn3d.c
-    - vf_kerndeint.c
-    - vf_mcdeint.c
-    - vf_mp.c
-    - vf_owdenoise.c
-    - vf_perspective.c
-    - vf_phase.c
-    - vf_pp.c
-    - vf_pullup.c
-    - vf_sab.c
-    - vf_smartblur.c
-    - vf_spp.c
-    - vf_stereo3d.c
-    - vf_super2xsai.c
-    - vf_tinterlace.c
-    - vsrc_mptestsrc.c
+    - `f_ebur128.c`
+    - `vf_blackframe.c`
+    - `vf_boxblur.c`
+    - `vf_colormatrix.c`
+    - `vf_cover_rect.c`
+    - `vf_cropdetect.c`
+    - `vf_delogo.c`
+    - `vf_eq.c`
+    - `vf_find_rect.c`
+    - `vf_fspp.c`
+    - `vf_geq.c`
+    - `vf_histeq.c`
+    - `vf_hqdn3d.c`
+    - `vf_interlace.c`
+    - `vf_kerndeint.c`
+    - `vf_mcdeint.c`
+    - `vf_mpdecimate.c`
+    - `vf_owdenoise.c`
+    - `vf_perspective.c`
+    - `vf_phase.c`
+    - `vf_pp.c`
+    - `vf_pp7.c`
+    - `vf_pullup.c`
+    - `vf_sab.c`
+    - `vf_smartblur.c`
+    - `vf_repeatfields.c`
+    - `vf_spp.c`
+    - `vf_stereo3d.c`
+    - `vf_super2xsai.c`
+    - `vf_tinterlace.c`
+    - `vf_uspp.c`
+    - `vsrc_mptestsrc.c`
 
 Should you, for whatever reason, prefer to use version 3 of the (L)GPL, then
-the configure parameter --enable-version3 will activate this licensing option
-for you. Read the file COPYING.LGPLv3 or, if you have enabled GPL parts,
-COPYING.GPLv3 to learn the exact legal terms that apply in this case.
+the configure parameter `--enable-version3` will activate this licensing option
+for you. Read the file `COPYING.LGPLv3` or, if you have enabled GPL parts,
+`COPYING.GPLv3` to learn the exact legal terms that apply in this case.
 
 There are a handful of files under other licensing terms, namely:
 
-* The files libavcodec/jfdctfst.c, libavcodec/jfdctint_template.c and
-  libavcodec/jrevdct.c are taken from libjpeg, see the top of the files for
+* The files `libavcodec/jfdctfst.c`, `libavcodec/jfdctint_template.c` and
+  `libavcodec/jrevdct.c` are taken from libjpeg, see the top of the files for
   licensing details. Specifically note that you must credit the IJG in the
   documentation accompanying your program if you only distribute executables.
   You must also indicate any changes including additions and deletions to
   those three files in the documentation.
+* `tests/reference.pnm` is under the expat license.
 
 
 external libraries
@@ -74,21 +83,22 @@ compatible libraries
 --------------------
 
 The following libraries are under GPL:
-    - frei0r
-    - libcdio
-    - libutvideo
-    - libvidstab
-    - libx264
-    - libx265
-    - libxavs
-    - libxvid
+- frei0r
+- libcdio
+- libutvideo
+- libvidstab
+- libx264
+- libx265
+- libxavs
+- libxvid
+
 When combining them with FFmpeg, FFmpeg needs to be licensed as GPL as well by
-passing --enable-gpl to configure.
+passing `--enable-gpl` to configure.
 
 The OpenCORE and VisualOn libraries are under the Apache License 2.0. That
 license is incompatible with the LGPL v2.1 and the GPL v2, but not with
 version 3 of those licenses. So to combine these libraries with FFmpeg, the
-license version needs to be upgraded by passing --enable-version3 to configure.
+license version needs to be upgraded by passing `--enable-version3` to configure.
 
 incompatible libraries
 ----------------------
@@ -96,7 +106,7 @@ incompatible libraries
 The Fraunhofer AAC library, FAAC and aacplus are under licenses which
 are incompatible with the GPLv2 and v3. We do not know for certain if their
 licenses are compatible with the LGPL.
-If you wish to enable these libraries, pass --enable-nonfree to configure.
+If you wish to enable these libraries, pass `--enable-nonfree` to configure.
 But note that if you enable any of these libraries the resulting binary will
 be under a complex license mix that is more restrictive than the LGPL and that
 may result in additional obligations. It is possible that these

MAINTAINERS

@@ -14,7 +14,6 @@ patches and related discussions.
 Project Leader
 ==============
 
-Michael Niedermayer
   final design decisions
 
 
@@ -43,9 +42,9 @@ QuickTime faststart:
 Miscellaneous Areas
 ===================
 
-documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu
+documentation                           Stefano Sabatini, Mike Melanson, Timothy Gu, Lou Logan
 build system (configure, makefiles)     Diego Biurrun, Mans Rullgard
-project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser
+project server                          Árpád Gereöffy, Michael Niedermayer, Reimar Doeffinger, Alexander Strasser, Lou Logan
 presets                                 Robert Swain
 metadata subsystem                      Aurelien Jacobs
 release management                      Michael Niedermayer
@@ -54,12 +53,12 @@ release management                      Michael Niedermayer
 Communication
 =============
 
-website                                 Robert Swain
+website                                 Deby Barbara Lepage
 fate.ffmpeg.org                         Timothy Gu
 Trac bug tracker                        Alexander Strasser, Michael Niedermayer, Carl Eugen Hoyos, Lou Logan
-mailing lists                           Michael Niedermayer, Baptiste Coudurier, Lou Logan
+mailing lists                           Baptiste Coudurier, Lou Logan
 Google+                                 Paul B Mahol, Michael Niedermayer, Alexander Strasser
-Twitter                                 Lou Logan
+Twitter                                 Lou Logan, Reynaldo H. Verdejo Pinochet
 Launchpad                               Timothy Gu
 
 
@@ -138,6 +137,7 @@ Codecs:
   4xm.c                                 Michael Niedermayer
   8bps.c                                Roberto Togni
   8svx.c                                Jaikrishnan Menon
+  aacenc*, aaccoder.c                   Rostislav Pehlivanov
   aasc.c                                Kostya Shishkov
   ac3*                                  Justin Ruggles
   alacenc.c                             Jaikrishnan Menon
@@ -156,6 +156,7 @@ Codecs:
   celp_filters.*                        Vitor Sessak
   cinepak.c                             Roberto Togni
   cinepakenc.c                          Rl / Aetey G.T. AB
+  ccaption_dec.c                        Anshul Maheshwari
   cljr                                  Alex Beregszaszi
   cllc.c                                Derek Buitenhuis
   cook.c, cookdata.h                    Benjamin Larsson
@@ -165,10 +166,12 @@ Codecs:
   dca.c                                 Kostya Shishkov, Benjamin Larsson
   dnxhd*                                Baptiste Coudurier
   dpcm.c                                Mike Melanson
+  dss_sp.c                              Oleksij Rempel
   dv.c                                  Roman Shaposhnik
   dvbsubdec.c                           Anshul Maheshwari
   dxa.c                                 Kostya Shishkov
   eacmv*, eaidct*, eat*                 Peter Ross
+  evrc*                                 Paul B Mahol
   exif.c, exif.h                        Thilo Borgmann
   ffv1*                                 Michael Niedermayer
   ffwavesynth.c                         Nicolas George
@@ -198,6 +201,7 @@ Codecs:
   libcelt_dec.c                         Nicolas George
   libdirac*                             David Conrad
   libgsm.c                              Michel Bardiaux
+  libkvazaar.c                          Arttu Ylä-Outinen
   libopenjpeg.c                         Jaikrishnan Menon
   libopenjpegenc.c                      Michael Bradshaw
   libschroedinger*                      David Conrad
@@ -226,6 +230,7 @@ Codecs:
   msvideo1.c                            Mike Melanson
   nellymoserdec.c                       Benjamin Larsson
   nuv.c                                 Reimar Doeffinger
+  nvenc.c                               Timo Rothenpieler
   paf.*                                 Paul B Mahol
   pcx.c                                 Ivo van Poorten
   pgssubdec.c                           Reimar Doeffinger
@@ -234,6 +239,7 @@ Codecs:
   qdm2.c, qdm2data.h                    Roberto Togni, Benjamin Larsson
   qdrw.c                                Kostya Shishkov
   qpeg.c                                Kostya Shishkov
+  qsv*                                  Ivan Uskov
   qtrle.c                               Mike Melanson
   ra144.c, ra144.h, ra288.c, ra288.h    Roberto Togni
   resample2.c                           Michael Niedermayer
@@ -295,11 +301,12 @@ Codecs:
 
 Hardware acceleration:
   crystalhd.c                           Philip Langdale
-  dxva2*                                Laurent Aimar
+  dxva2*                                Hendrik Leppkes, Laurent Aimar
   libstagefright.cpp                    Mohamed Naufal
   vaapi*                                Gwenole Beauchesne
   vda*                                  Sebastien Zwickert
-  vdpau*                                Carl Eugen Hoyos
+  vdpau*                                Philip Langdale, Carl Eugen Hoyos
+  videotoolbox*                         Sebastien Zwickert
 
 
 libavdevice
@@ -309,6 +316,7 @@ libavdevice
 
 
   avfoundation.m                        Thilo Borgmann
+  decklink*                             Deti Fliegl
   dshow.c                               Roger Pack (CC rogerdpack@gmail.com)
   fbdev_enc.c                           Lukasz Marek
   gdigrab.c                             Roger Pack (CC rogerdpack@gmail.com)
@@ -319,7 +327,7 @@ libavdevice
   pulse_audio_enc.c                     Lukasz Marek
   qtkit.m                               Thilo Borgmann
   sdl                                   Stefano Sabatini
-  v4l2.c                                Luca Abeni
+  v4l2.c                                Giorgio Vazzana
   vfwcap.c                              Ramiro Polla
   xv.c                                  Lukasz Marek
 
@@ -330,6 +338,7 @@ Generic parts:
   graphdump.c                           Nicolas George
 
 Filters:
+  f_drawgraph.c                         Paul B Mahol
   af_adelay.c                           Paul B Mahol
   af_aecho.c                            Paul B Mahol
   af_afade.c                            Paul B Mahol
@@ -340,27 +349,41 @@ Filters:
   af_astreamsync.c                      Nicolas George
   af_atempo.c                           Pavel Koshevoy
   af_biquads.c                          Paul B Mahol
+  af_chorus.c                           Paul B Mahol
   af_compand.c                          Paul B Mahol
   af_ladspa.c                           Paul B Mahol
   af_pan.c                              Nicolas George
+  af_sidechaincompress.c                Paul B Mahol
+  af_silenceremove.c                    Paul B Mahol
+  avf_aphasemeter.c                     Paul B Mahol
   avf_avectorscope.c                    Paul B Mahol
   avf_showcqt.c                         Muhammad Faiz
   vf_blend.c                            Paul B Mahol
+  vf_colorchannelmixer.c                Paul B Mahol
   vf_colorbalance.c                     Paul B Mahol
+  vf_colorkey.c                         Timo Rothenpieler
+  vf_colorlevels.c                      Paul B Mahol
+  vf_deband.c                           Paul B Mahol
   vf_dejudder.c                         Nicholas Robbins
   vf_delogo.c                           Jean Delvare (CC <khali@linux-fr.org>)
   vf_drawbox.c/drawgrid                 Andrey Utkin
   vf_extractplanes.c                    Paul B Mahol
   vf_histogram.c                        Paul B Mahol
   vf_hqx.c                              Clément Bœsch
+  vf_idet.c                             Pascal Massimino
   vf_il.c                               Paul B Mahol
+  vf_lenscorrection.c                   Daniel Oberhoff
   vf_mergeplanes.c                      Paul B Mahol
+  vf_neighbor.c                         Paul B Mahol
   vf_psnr.c                             Paul B Mahol
+  vf_random.c                           Paul B Mahol
   vf_scale.c                            Michael Niedermayer
   vf_separatefields.c                   Paul B Mahol
+  vf_ssim.c                             Paul B Mahol
   vf_stereo3d.c                         Paul B Mahol
   vf_telecine.c                         Paul B Mahol
   vf_yadif.c                            Michael Niedermayer
+  vf_zoompan.c                          Paul B Mahol
 
 Sources:
   vsrc_mandelbrot.c                     Michael Niedermayer
@@ -377,11 +400,13 @@ Generic parts:
 
 Muxers/Demuxers:
   4xm.c                                 Mike Melanson
+  aadec.c                               Vesselin Bontchev (vesselin.bontchev at yandex dot com)
   adtsenc.c                             Robert Swain
   afc.c                                 Paul B Mahol
   aiffdec.c                             Baptiste Coudurier, Matthieu Bouron
   aiffenc.c                             Baptiste Coudurier, Matthieu Bouron
   ape.c                                 Kostya Shishkov
+  apngdec.c                             Benoit Fouet
   ass*                                  Aurelien Jacobs
   astdec.c                              Paul B Mahol
   astenc.c                              James Almer
@@ -394,6 +419,7 @@ Muxers/Demuxers:
   cdxl.c                                Paul B Mahol
   crc.c                                 Michael Niedermayer
   daud.c                                Reimar Doeffinger
+  dss.c                                 Oleksij Rempel
   dtshddec.c                            Paul B Mahol
   dv.c                                  Roman Shaposhnik
   dxa.c                                 Kostya Shishkov
@@ -406,6 +432,7 @@ Muxers/Demuxers:
   gxf.c                                 Reimar Doeffinger
   gxfenc.c                              Baptiste Coudurier
   hls.c                                 Anssi Hannula
+  hls encryption (hlsenc.c)             Christian Suloway
   idcin.c                               Mike Melanson
   idroqdec.c                            Mike Melanson
   iff.c                                 Jaikrishnan Menon
@@ -428,7 +455,7 @@ Muxers/Demuxers:
   mgsts.c                               Paul B Mahol
   microdvd*                             Aurelien Jacobs
   mm.c                                  Peter Ross
-  mov.c                                 Michael Niedermayer, Baptiste Coudurier
+  mov.c                                 Baptiste Coudurier
   movenc.c                              Baptiste Coudurier, Matthieu Bouron
   mpc.c                                 Kostya Shishkov
   mpeg.c                                Michael Niedermayer
@@ -458,12 +485,19 @@ Muxers/Demuxers:
   rmdec.c, rmenc.c                      Ronald S. Bultje, Kostya Shishkov
   rtmp*                                 Kostya Shishkov
   rtp.c, rtpenc.c                       Martin Storsjo
+  rtpdec_ac3.*                          Gilles Chanteperdrix
+  rtpdec_dv.*                           Thomas Volkert
+  rtpdec_h261.*, rtpenc_h261.*          Thomas Volkert
+  rtpdec_hevc.*, rtpenc_hevc.*          Thomas Volkert
+  rtpdec_mpa_robust.*                   Gilles Chanteperdrix
   rtpdec_asf.*                          Ronald S. Bultje
+  rtpdec_vp9.c                          Thomas Volkert
   rtpenc_mpv.*, rtpenc_aac.*            Martin Storsjo
   rtsp.c                                Luca Barbato
   sbgdec.c                              Nicolas George
   sdp.c                                 Martin Storsjo
   segafilm.c                            Mike Melanson
+  segment.c                             Stefano Sabatini
   siff.c                                Kostya Shishkov
   smacker.c                             Kostya Shishkov
   smjpeg*                               Paul B Mahol
@@ -484,12 +518,14 @@ Muxers/Demuxers:
   wvenc.c                               Paul B Mahol
 
 Protocols:
+  async.c                               Zhang Rui
   bluray.c                              Petri Hintukainen
   ftp.c                                 Lukasz Marek
   http.c                                Ronald S. Bultje
   libssh.c                              Lukasz Marek
   mms*.c                                Ronald S. Bultje
   udp.c                                 Luca Abeni
+  icecast.c                             Marvin Scholz
 
 
 libswresample
@@ -518,19 +554,21 @@ Amiga / PowerPC                         Colin Ward
 Linux / PowerPC                         Luca Barbato
 Windows MinGW                           Alex Beregszaszi, Ramiro Polla
 Windows Cygwin                          Victor Paesa
-Windows MSVC                            Matthew Oliver
+Windows MSVC                            Matthew Oliver, Hendrik Leppkes
 Windows ICL                             Matthew Oliver
 ADI/Blackfin DSP                        Marc Hoffman
 Sparc                                   Roman Shaposhnik
-x86                                     Michael Niedermayer
+OS/2                                    KO Myung-Hun
 
 
 Releases
 ========
 
-2.2                                     Michael Niedermayer
-2.1                                     Michael Niedermayer
-1.2                                     Michael Niedermayer
+2.8                                     Michael Niedermayer
+2.7                                     Michael Niedermayer
+2.6                                     Michael Niedermayer
+2.5                                     Michael Niedermayer
+2.4                                     Michael Niedermayer
 
 If you want to maintain an older release, please contact us
 
@@ -561,6 +599,7 @@ Michael Niedermayer           9FF2 128B 147E F673 0BAD F133 611E C787 040B 0FAB
 Nicolas George                24CE 01CE 9ACC 5CEB 74D8 8D9D B063 D997 36E5 4C93
 Panagiotis Issaris            6571 13A3 33D9 3726 F728 AA98 F643 B12E ECF3 E029
 Peter Ross                    A907 E02F A6E5 0CD2 34CD 20D2 6760 79C5 AC40 DD6B
+Philip Langdale               5DC5 8D66 5FBA 3A43 18EC 045E F8D6 B194 6A75 682E
 Reimar Doeffinger             C61D 16E5 9E2C D10C 8958 38A4 0899 A2B9 06D4 D9C7
 Reinhard Tartler              9300 5DC2 7E87 6C37 ED7B CA9A 9808 3544 9453 48A4
 Reynaldo H. Verdejo Pinochet  6E27 CD34 170C C78E 4D4F 5F40 C18E 077F 3114 452A

Makefile

@@ -31,7 +31,11 @@ $(foreach prog,$(AVBASENAMES),$(eval OBJS-$(prog)-$(CONFIG_OPENCL) += cmdutils_o
 OBJS-ffmpeg                   += ffmpeg_opt.o ffmpeg_filter.o
 OBJS-ffmpeg-$(HAVE_VDPAU_X11) += ffmpeg_vdpau.o
 OBJS-ffmpeg-$(HAVE_DXVA2_LIB) += ffmpeg_dxva2.o
-OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_vda.o
+ifndef CONFIG_VIDEOTOOLBOX
+OBJS-ffmpeg-$(CONFIG_VDA)     += ffmpeg_videotoolbox.o
+endif
+OBJS-ffmpeg-$(CONFIG_VIDEOTOOLBOX) += ffmpeg_videotoolbox.o
+OBJS-ffserver                 += ffserver_config.o
 
 TESTTOOLS   = audiogen videogen rotozoom tiny_psnr tiny_ssim base64
 HOSTPROGS  := $(TESTTOOLS:%=tests/%) doc/print_options
@@ -59,11 +63,12 @@ include $(SRC_PATH)/common.mak
 
 FF_EXTRALIBS := $(FFEXTRALIBS)
 FF_DEP_LIBS  := $(DEP_LIBS)
+FF_STATIC_DEP_LIBS := $(STATIC_DEP_LIBS)
 
 all: $(AVPROGS)
 
 $(TOOLS): %$(EXESUF): %.o $(EXEOBJS)
-	$(LD) $(LDFLAGS) $(LD_O) $^ $(ELIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $^ $(ELIBS)
 
 tools/cws2fws$(EXESUF): ELIBS = $(ZLIB)
 tools/uncoded_frame$(EXESUF): $(FF_DEP_LIBS)
@@ -79,8 +84,8 @@ SUBDIR_VARS := CLEANFILES EXAMPLES FFLIBS HOSTPROGS TESTPROGS TOOLS      \
                HEADERS ARCH_HEADERS BUILT_HEADERS SKIPHEADERS            \
                ARMV5TE-OBJS ARMV6-OBJS ARMV8-OBJS VFP-OBJS NEON-OBJS     \
                ALTIVEC-OBJS MMX-OBJS YASM-OBJS                           \
-               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MIPS32R2-OBJS  \
-               OBJS SLIBOBJS HOSTOBJS TESTOBJS
+               MIPSFPU-OBJS MIPSDSPR2-OBJS MIPSDSPR1-OBJS MSA-OBJS       \
+               MMI-OBJS OBJS SLIBOBJS HOSTOBJS TESTOBJS
 
 define RESET
 $(1) :=
@@ -92,6 +97,7 @@ $(foreach V,$(SUBDIR_VARS),$(eval $(call RESET,$(V))))
 SUBDIR := $(1)/
 include $(SRC_PATH)/$(1)/Makefile
 -include $(SRC_PATH)/$(1)/$(ARCH)/Makefile
+-include $(SRC_PATH)/$(1)/$(INTRINSICS)/Makefile
 include $(SRC_PATH)/library.mak
 endef
 
@@ -110,14 +116,14 @@ endef
 
 $(foreach P,$(PROGS),$(eval $(call DOPROG,$(P:$(PROGSSUF)$(EXESUF)=))))
 
-ffprobe.o cmdutils.o : libavutil/ffversion.h
+ffprobe.o cmdutils.o libavcodec/utils.o libavformat/utils.o libavdevice/avdevice.o libavfilter/avfilter.o libavutil/utils.o libpostproc/postprocess.o libswresample/swresample.o libswscale/utils.o : libavutil/ffversion.h
 
 $(PROGS): %$(PROGSSUF)$(EXESUF): %$(PROGSSUF)_g$(EXESUF)
 	$(CP) $< $@
 	$(STRIP) $@
 
 %$(PROGSSUF)_g$(EXESUF): %.o $(FF_DEP_LIBS)
-	$(LD) $(LDFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
+	$(LD) $(LDFLAGS) $(LDEXEFLAGS) $(LD_O) $(OBJS-$*) $(FF_EXTRALIBS)
 
 OBJDIRS += tools
 
@@ -173,7 +179,7 @@ clean::
 
 distclean::
 	$(RM) $(DISTCLEANSUFFIXES)
-	$(RM) config.* .config libavutil/avconfig.h .version version.h libavutil/ffversion.h libavcodec/codec_names.h
+	$(RM) config.* .config libavutil/avconfig.h .version avversion.h version.h libavutil/ffversion.h libavcodec/codec_names.h
 
 config:
 	$(SRC_PATH)/configure $(value FFMPEG_CONFIGURATION)

README.md

@@ -1,19 +1,42 @@
 FFmpeg README
 =============
 
-1) Documentation
-----------------
+FFmpeg is a collection of libraries and tools to process multimedia content
+such as audio, video, subtitles and related metadata.
 
-* Read the documentation in the doc/ directory in git.
+## Libraries
 
-  You can also view it online at http://ffmpeg.org/documentation.html
+* `libavcodec` provides implementation of a wider range of codecs.
+* `libavformat` implements streaming protocols, container formats and basic I/O access.
+* `libavutil` includes hashers, decompressors and miscellaneous utility functions.
+* `libavfilter` provides a mean to alter decoded Audio and Video through chain of filters.
+* `libavdevice` provides an abstraction to access capture and playback devices.
+* `libswresample` implements audio mixing and resampling routines.
+* `libswscale` implements color conversion and scaling routines.
 
-2) Licensing
-------------
+## Tools
 
-* See the LICENSE file.
+* [ffmpeg](http://ffmpeg.org/ffmpeg.html) is a command line toolbox to
+  manipulate, convert and stream multimedia content.
+* [ffplay](http://ffmpeg.org/ffplay.html) is a minimalistic multimedia player.
+* [ffprobe](http://ffmpeg.org/ffprobe.html) is a simple analysis tool to inspect
+  multimedia content.
+* [ffserver](http://ffmpeg.org/ffserver.html) is a multimedia streaming server
+  for live broadcasts.
+* Additional small tools such as `aviocat`, `ismindex` and `qt-faststart`.
 
-3) Build and Install
---------------------
+## Documentation
 
-* See the INSTALL file.
+The offline documentation is available in the **doc/** directory.
+
+The online documentation is available in the main [website](http://ffmpeg.org)
+and in the [wiki](http://trac.ffmpeg.org).
+
+### Examples
+
+Coding examples are available in the **doc/examples** directory.
+
+## License
+
+FFmpeg codebase is mainly LGPL-licensed with optional components licensed under
+GPL. Please refer to the LICENSE file for detailed information.

RELEASE

@@ -1 +1 @@
-2.2.git
+2.8.19

RELEASE_NOTES

@@ -1,175 +1,15 @@
- ┌───────────────────────────────────────────┐
- │ RELEASE NOTES for FFmpeg 2.3 "Mandelbrot" │
- └───────────────────────────────────────────┘
 
-   The FFmpeg Project proudly presents FFmpeg 2.3 "Mandelbrot", a major
-   release with all the great features committed during the three-month period
-   since the release of FFmpeg 2.2.
+              ┌────────────────────────────────────────┐
+              │ RELEASE NOTES for FFmpeg 2.8 "Feynman" │
+              └────────────────────────────────────────┘
 
-   In this release, there are lots of internal overhauls that make FFmpeg a
-   more accessible project for new developers. Many important new features
-   like QTKit and AVFoundation input devices are committed. Contributions done
-   by Libav such as a new native Opus decoder are also merged.
+   The FFmpeg Project proudly presents FFmpeg 2.8 "Feynman", about 3
+   months after the release of FFmpeg 2.7.
 
-   Because of the increasing difficulty to maintain and lack of maintainers,
-   we are very sorry to say that we have removed all Blackfin and SPARC
-   architecture assembly optimizations with the cleanups done. If you are
-   interested in maintaining optimization for these two architecture, feel
-   free to contact us and we will restore the code!
+   A complete Changelog is available at the root of the project, and the
+   complete Git history on http://source.ffmpeg.org.
 
-   Since this release, the traditional Changelog file is upgraded to this
-   modern-looking release note. Old changelogs are moved to doc/Changelog.old.
-
-   Enjoy!
-
-   ┌────────────────────────────┐
-   │ *  API Information         │
-   └────────────────────────────┘
-
-     FFmpeg 2.3 is completely source-compatible to the FFmpeg 2.2 series. There
-     are however some API deprecations that you need to take care of. Use `git
-     diff n2.2 n2.3 doc/APIchanges` to show the list of added and deprecated
-     APIs. FFmpeg 2.3 includes the following library versions:
-
-       • libavutil      52.92.100
-       • libavcodec     55.69.100
-       • libavformat    55.47.100
-       • libavdevice    55.13.102
-       • libavfilter     4.10.100
-       • libswscale      2. 6.100
-       • libswresample   0.19.100
-       • libpostproc    52. 3.100
-
-     Please refer to the doc/APIChanges file for more information.
-
-   ┌────────────────────────────┐
-   │ New Optimization           │
-   └────────────────────────────┘
-
-     We are excited to announce that we have committed new x86 assembly
-     optimization for HEVC, and FFmpeg's audio resampler libswresample. ARM
-     users will get a boost in MLP/TrueHD decoding thanks to new optimization.
-     Decoding Huffyuv also got a major boost from optimization on the C code.
-
-     Of special interest for Microsoft Visual Studio users, we have also
-     converted some preexisting x86 assembly to NASM/Yasm format compatible
-     with MSVC setup, especially in the area of audio resampling.
-
-     Another major feature in this release is the introduction of AArch64
-     (ARMv8) assembly optimization. AArch64 is another name for the first
-     64-bit ARM architecture, used by Apple A7 SoC inside iPhone 5S. Some
-     32-bit ARM assembly has already been ported to AArch64, but more work is
-     underway.
-
-   ┌────────────────────────────┐
-   │ Native Opus decoder        │
-   └────────────────────────────┘
-
-     Opus is an open audio format jointly developed by Xiph.Org, Mozilla,
-     Skype/Microsoft, and Broadcom. It combines the features of the Skype Cilk
-     speech codec and the Xiph.Org CELT music codec into one low-latency
-     codec. Decoding Opus is already possible since FFmpeg 1.0 using the
-     libopus library, but the new Opus native decoder brings a higher level of
-     stability and speed.
-
-   ┌────────────────────────────┐
-   │ QTKit and AVFoundation     │
-   └────────────────────────────┘
-
-     For OS X users, the new QTKit and AVFoundation devices allow you to use
-     the integrated camera on Macs. AVFoundation is a newer API only available
-     on OS X 10.7 "Lion" or newer. For users with older OS X systems, the
-     QTKit device using the older OS X API is for you.
-
-   ┌────────────────────────────┐
-   │ API Additions              │
-   └────────────────────────────┘
-
-     In this release, stream side data are introduced as AVStream.side_data as
-     a way to store miscellaneous stream-wide information. The format is
-     similar to the previously anonymous structure AVPacket.side_data (now
-     named as AVPacketSideData). With this change, audio ReplayGain
-     information and video rotation matrix are now exported through this API,
-     if available in the demuxer.
-
-     We also have improved libswresample's Doxygen API documentation, so new
-     developers wishing to use FFmpeg's excellent libraries can get started
-     more easily and faster.
-
-   ┌────────────────────────────┐
-   │ Last But Not Least         │
-   └────────────────────────────┘
-
-     Other interesting new features including hqx video filter, a pixel art
-     scaling filter; a fixed-point AC-3 decoder contributed by Imagination
-     Technologies; an On2 TrueMotion VP7 video decoder; an HTML5 WebVTT
-     subtitle decoder that allows creation of WebVTT from any text-based
-     subtitles; and an 1-bit Direct Stream Digital audio decoder.
-
- ┌────────────────────────────┐
- │ ★  List of New Features    │
- └────────────────────────────┘
-
-   ┌────────────────────────────┐
-   │ Command line tools         │
-   └────────────────────────────┘
-
-    • Support for decoding through DXVA2 in ffmpeg
-
-   ┌────────────────────────────┐
-   │ libavcodec                 │
-   └────────────────────────────┘
-
-    • AC3 fixed-point decoding
-    • VP7 video decoder
-    • Alias PIX image encoder and decoder
-    • Improvements to the BRender PIX image decoder
-    • Improvements to the XBM decoder
-    • Improvements to OpenEXR image decoder
-    • Support decoding 16-bit RLE SGI images
-    • Direct Stream Digital (DSD) decoder
-    • On2 AVC (Audio for Video) decoder
-    • Native Opus decoder
-    • WebVTT encoder
-
-   ┌────────────────────────────┐
-   │ libavdevice                │
-   └────────────────────────────┘
-
-    • QTKit input device
-    • GDI screen grabbing for Windows
-    • AVFoundation input device
-
-   ┌────────────────────────────┐
-   │ libavformat                │
-   └────────────────────────────┘
-
-    • subfile protocol
-    • Phantom Cine demuxer
-    • Alternative rendition support for HTTP Live Streaming
-    • Magic Lantern Video (MLV) demuxer
-    • Image format auto-detection
-    • LRC lyric file demuxer and muxer
-    • Samba protocol (via libsmbclient)
-    • WebM DASH Manifest muxer
-
-   ┌────────────────────────────┐
-   │ libavfilter                │
-   └────────────────────────────┘
-
-    • shuffleplanes filter
-    • libbs2b-based stereo-to-binaural audio filter
-    • showcqt multimedia filter
-    • zoompan filter
-    • signalstats filter
-    • hqx filter (hq2x, hq3x, hq4x)
-    • flanger filter
-    • libfribidi support in drawtext
-
- ┌────────────────────────────┐
- │ ⚠  Behaviour changes       │
- └────────────────────────────┘
-
-  • libx264 reference frames count is now limited depending on level chosen
-  • Because of the new image format auto-detection feature, you don't need to
-    specify image format when decoding an image with no extension.
+   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.

arch.mak

@@ -5,11 +5,13 @@ OBJS-$(HAVE_VFP)     += $(VFP-OBJS)     $(VFP-OBJS-yes)
 OBJS-$(HAVE_NEON)    += $(NEON-OBJS)    $(NEON-OBJS-yes)
 
 OBJS-$(HAVE_MIPSFPU)   += $(MIPSFPU-OBJS)    $(MIPSFPU-OBJS-yes)
-OBJS-$(HAVE_MIPS32R2)  += $(MIPS32R2-OBJS)   $(MIPS32R2-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR1) += $(MIPSDSPR1-OBJS)  $(MIPSDSPR1-OBJS-yes)
 OBJS-$(HAVE_MIPSDSPR2) += $(MIPSDSPR2-OBJS)  $(MIPSDSPR2-OBJS-yes)
+OBJS-$(HAVE_MSA)       += $(MSA-OBJS)        $(MSA-OBJS-yes)
+OBJS-$(HAVE_MMI)   += $(MMI-OBJS)   $(MMI-OBJS-yes)
 
 OBJS-$(HAVE_ALTIVEC) += $(ALTIVEC-OBJS) $(ALTIVEC-OBJS-yes)
+OBJS-$(HAVE_VSX)     += $(VSX-OBJS) $(VSX-OBJS-yes)
 
 OBJS-$(HAVE_MMX)     += $(MMX-OBJS)     $(MMX-OBJS-yes)
 OBJS-$(HAVE_YASM)    += $(YASM-OBJS)    $(YASM-OBJS-yes)

cmdutils.c

@@ -41,8 +41,10 @@
 #include "libavutil/avassert.h"
 #include "libavutil/avstring.h"
 #include "libavutil/bprint.h"
+#include "libavutil/display.h"
 #include "libavutil/mathematics.h"
 #include "libavutil/imgutils.h"
+#include "libavutil/libm.h"
 #include "libavutil/parseutils.h"
 #include "libavutil/pixdesc.h"
 #include "libavutil/eval.h"
@@ -58,10 +60,13 @@
 #include <sys/time.h>
 #include <sys/resource.h>
 #endif
+#if HAVE_SETDLLDIRECTORY
+#include <windows.h>
+#endif
 
 static int init_report(const char *env);
 
-struct SwsContext *sws_opts;
+AVDictionary *sws_dict;
 AVDictionary *swr_opts;
 AVDictionary *format_opts, *codec_opts, *resample_opts;
 
@@ -71,20 +76,13 @@ int hide_banner = 0;
 
 void init_opts(void)
 {
-
-    if(CONFIG_SWSCALE)
-        sws_opts = sws_getContext(16, 16, 0, 16, 16, 0, SWS_BICUBIC,
-                              NULL, NULL, NULL);
+    av_dict_set(&sws_dict, "flags", "bicubic", 0);
 }
 
 void uninit_opts(void)
 {
-#if CONFIG_SWSCALE
-    sws_freeContext(sws_opts);
-    sws_opts = NULL;
-#endif
-
     av_dict_free(&swr_opts);
+    av_dict_free(&sws_dict);
     av_dict_free(&format_opts);
     av_dict_free(&codec_opts);
     av_dict_free(&resample_opts);
@@ -111,6 +109,15 @@ static void log_callback_report(void *ptr, int level, const char *fmt, va_list v
     }
 }
 
+void init_dynload(void)
+{
+#if HAVE_SETDLLDIRECTORY
+    /* Calling SetDllDirectory with the empty string (but not NULL) removes the
+     * current working directory from the DLL search path as a security pre-caution. */
+    SetDllDirectory("");
+#endif
+}
+
 static void (*program_exit)(int ret);
 
 void register_exit(void (*cb)(int ret))
@@ -166,7 +173,7 @@ void show_help_options(const OptionDef *options, const char *msg, int req_flags,
     int first;
 
     first = 1;
-    for (po = options; po->name != NULL; po++) {
+    for (po = options; po->name; po++) {
         char buf[64];
 
         if (((po->flags & req_flags) != req_flags) ||
@@ -205,7 +212,7 @@ static const OptionDef *find_option(const OptionDef *po, const char *name)
     const char *p = strchr(name, ':');
     int len = p ? p - name : strlen(name);
 
-    while (po->name != NULL) {
+    while (po->name) {
         if (!strncmp(name, po->name, len) && strlen(po->name) == len)
             break;
         po++;
@@ -254,7 +261,7 @@ static void prepare_app_arguments(int *argc_ptr, char ***argv_ptr)
 
     win32_argv_utf8 = av_mallocz(sizeof(char *) * (win32_argc + 1) + buffsize);
     argstr_flat     = (char *)win32_argv_utf8 + sizeof(char *) * (win32_argc + 1);
-    if (win32_argv_utf8 == NULL) {
+    if (!win32_argv_utf8) {
         LocalFree(argv_w);
         return;
     }
@@ -290,10 +297,14 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
     if (po->flags & OPT_SPEC) {
         SpecifierOpt **so = dst;
         char *p = strchr(opt, ':');
+        char *str;
 
         dstcount = (int *)(so + 1);
         *so = grow_array(*so, sizeof(**so), dstcount, *dstcount + 1);
-        (*so)[*dstcount - 1].specifier = av_strdup(p ? p + 1 : "");
+        str = av_strdup(p ? p + 1 : "");
+        if (!str)
+            return AVERROR(ENOMEM);
+        (*so)[*dstcount - 1].specifier = str;
         dst = &(*so)[*dstcount - 1].u;
     }
 
@@ -301,6 +312,8 @@ static int write_option(void *optctx, const OptionDef *po, const char *opt,
         char *str;
         str = av_strdup(arg);
         av_freep(dst);
+        if (!str)
+            return AVERROR(ENOMEM);
         *(char **)dst = str;
     } else if (po->flags & OPT_BOOL || po->flags & OPT_INT) {
         *(int *)dst = parse_number_or_die(opt, arg, OPT_INT64, INT_MIN, INT_MAX);
@@ -444,7 +457,7 @@ int locate_option(int argc, char **argv, const OptionDef *options,
              (po->name && !strcmp(optname, po->name)))
             return i;
 
-        if (po->flags & HAS_ARG)
+        if (!po->name || po->flags & HAS_ARG)
             i++;
     }
     return 0;
@@ -474,10 +487,22 @@ static void dump_argument(const char *a)
     fputc('"', report_file);
 }
 
+static void check_options(const OptionDef *po)
+{
+    while (po->name) {
+        if (po->flags & OPT_PERFILE)
+            av_assert0(po->flags & (OPT_INPUT | OPT_OUTPUT));
+        po++;
+    }
+}
+
 void parse_loglevel(int argc, char **argv, const OptionDef *options)
 {
     int idx = locate_option(argc, argv, options, "loglevel");
     const char *env;
+
+    check_options(options);
+
     if (!idx)
         idx = locate_option(argc, argv, options, "v");
     if (idx && argv[idx + 1])
@@ -509,7 +534,7 @@ static const AVOption *opt_find(void *obj, const char *name, const char *unit,
     return o;
 }
 
-#define FLAGS (o->type == AV_OPT_TYPE_FLAGS) ? AV_DICT_APPEND : 0
+#define FLAGS (o->type == AV_OPT_TYPE_FLAGS && (arg[0]=='-' || arg[0]=='+')) ? AV_DICT_APPEND : 0
 int opt_default(void *optctx, const char *opt, const char *arg)
 {
     const AVOption *o;
@@ -545,14 +570,24 @@ int opt_default(void *optctx, const char *opt, const char *arg)
     }
 #if CONFIG_SWSCALE
     sc = sws_get_class();
-    if (!consumed && opt_find(&sc, opt, NULL, 0,
-                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ)) {
-        // XXX we only support sws_flags, not arbitrary sws options
-        int ret = av_opt_set(sws_opts, opt, arg, 0);
+    if (!consumed && (o = opt_find(&sc, opt, NULL, 0,
+                         AV_OPT_SEARCH_CHILDREN | AV_OPT_SEARCH_FAKE_OBJ))) {
+        struct SwsContext *sws = sws_alloc_context();
+        int ret = av_opt_set(sws, opt, arg, 0);
+        sws_freeContext(sws);
+        if (!strcmp(opt, "srcw") || !strcmp(opt, "srch") ||
+            !strcmp(opt, "dstw") || !strcmp(opt, "dsth") ||
+            !strcmp(opt, "src_format") || !strcmp(opt, "dst_format")) {
+            av_log(NULL, AV_LOG_ERROR, "Directly using swscale dimensions/format options is not supported, please use the -s or -pix_fmt options\n");
+            return AVERROR(EINVAL);
+        }
         if (ret < 0) {
             av_log(NULL, AV_LOG_ERROR, "Error setting option %s.\n", opt);
             return ret;
         }
+
+        av_dict_set(&sws_dict, opt, arg, FLAGS);
+
         consumed = 1;
     }
 #else
@@ -626,9 +661,7 @@ static void finish_group(OptionParseContext *octx, int group_idx,
     *g             = octx->cur_group;
     g->arg         = arg;
     g->group_def   = l->group_def;
-#if CONFIG_SWSCALE
-    g->sws_opts    = sws_opts;
-#endif
+    g->sws_dict    = sws_dict;
     g->swr_opts    = swr_opts;
     g->codec_opts  = codec_opts;
     g->format_opts = format_opts;
@@ -637,9 +670,7 @@ static void finish_group(OptionParseContext *octx, int group_idx,
     codec_opts  = NULL;
     format_opts = NULL;
     resample_opts = NULL;
-#if CONFIG_SWSCALE
-    sws_opts    = NULL;
-#endif
+    sws_dict    = NULL;
     swr_opts    = NULL;
     init_opts();
 
@@ -695,9 +726,8 @@ void uninit_parse_context(OptionParseContext *octx)
             av_dict_free(&l->groups[j].codec_opts);
             av_dict_free(&l->groups[j].format_opts);
             av_dict_free(&l->groups[j].resample_opts);
-#if CONFIG_SWSCALE
-            sws_freeContext(l->groups[j].sws_opts);
-#endif
+
+            av_dict_free(&l->groups[j].sws_dict);
             av_dict_free(&l->groups[j].swr_opts);
         }
         av_freep(&l->groups);
@@ -839,6 +869,7 @@ int opt_loglevel(void *optctx, const char *opt, const char *arg)
         { "info"   , AV_LOG_INFO    },
         { "verbose", AV_LOG_VERBOSE },
         { "debug"  , AV_LOG_DEBUG   },
+        { "trace"  , AV_LOG_TRACE   },
     };
     char *tail;
     int level;
@@ -959,9 +990,10 @@ static int init_report(const char *env)
 
     report_file = fopen(filename.str, "w");
     if (!report_file) {
+        int ret = AVERROR(errno);
         av_log(NULL, AV_LOG_ERROR, "Failed to open report \"%s\": %s\n",
                filename.str, strerror(errno));
-        return AVERROR(errno);
+        return ret;
     }
     av_log_set_callback(log_callback_report);
     av_log(NULL, AV_LOG_INFO,
@@ -1074,8 +1106,7 @@ static void print_program_info(int flags, int level)
         av_log(NULL, level, " Copyright (c) %d-%d the FFmpeg developers",
                program_birth_year, CONFIG_THIS_YEAR);
     av_log(NULL, level, "\n");
-    av_log(NULL, level, "%sbuilt on %s %s with %s\n",
-           indent, __DATE__, __TIME__, CC_IDENT);
+    av_log(NULL, level, "%sbuilt with %s\n", indent, CC_IDENT);
 
     av_log(NULL, level, "%sconfiguration: " FFMPEG_CONFIGURATION "\n", indent);
 }
@@ -1212,12 +1243,7 @@ static int is_device(const AVClass *avclass)
 {
     if (!avclass)
         return 0;
-    return avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_VIDEO_INPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_OUTPUT ||
-           avclass->category == AV_CLASS_CATEGORY_DEVICE_INPUT;
+    return AV_IS_INPUT_DEVICE(avclass->category) || AV_IS_OUTPUT_DEVICE(avclass->category);
 }
 
 static int show_formats_devices(void *optctx, const char *opt, const char *arg, int device_only)
@@ -1242,7 +1268,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             is_dev = is_device(ofmt->priv_class);
             if (!is_dev && device_only)
                 continue;
-            if ((name == NULL || strcmp(ofmt->name, name) < 0) &&
+            if ((!name || strcmp(ofmt->name, name) < 0) &&
                 strcmp(ofmt->name, last_name) > 0) {
                 name      = ofmt->name;
                 long_name = ofmt->long_name;
@@ -1253,7 +1279,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             is_dev = is_device(ifmt->priv_class);
             if (!is_dev && device_only)
                 continue;
-            if ((name == NULL || strcmp(ifmt->name, name) < 0) &&
+            if ((!name || strcmp(ifmt->name, name) < 0) &&
                 strcmp(ifmt->name, last_name) > 0) {
                 name      = ifmt->name;
                 long_name = ifmt->long_name;
@@ -1262,7 +1288,7 @@ static int show_formats_devices(void *optctx, const char *opt, const char *arg,
             if (name && strcmp(ifmt->name, name) == 0)
                 decode = 1;
         }
-        if (name == NULL)
+        if (!name)
             break;
         last_name = name;
 
@@ -1308,12 +1334,12 @@ static void print_codec(const AVCodec *c)
     if (c->type == AVMEDIA_TYPE_VIDEO ||
         c->type == AVMEDIA_TYPE_AUDIO) {
         printf("    Threading capabilities: ");
-        switch (c->capabilities & (CODEC_CAP_FRAME_THREADS |
-                                   CODEC_CAP_SLICE_THREADS)) {
-        case CODEC_CAP_FRAME_THREADS |
-             CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
-        case CODEC_CAP_FRAME_THREADS: printf("frame");           break;
-        case CODEC_CAP_SLICE_THREADS: printf("slice");           break;
+        switch (c->capabilities & (AV_CODEC_CAP_FRAME_THREADS |
+                                   AV_CODEC_CAP_SLICE_THREADS)) {
+        case AV_CODEC_CAP_FRAME_THREADS |
+             AV_CODEC_CAP_SLICE_THREADS: printf("frame and slice"); break;
+        case AV_CODEC_CAP_FRAME_THREADS: printf("frame");           break;
+        case AV_CODEC_CAP_SLICE_THREADS: printf("slice");           break;
         default:                      printf("no");              break;
         }
         printf("\n");
@@ -1487,11 +1513,11 @@ static void print_codecs(int encoder)
 
         while ((codec = next_codec_for_id(desc->id, codec, encoder))) {
             printf(" %c", get_media_type_char(desc->type));
-            printf((codec->capabilities & CODEC_CAP_FRAME_THREADS) ? "F" : ".");
-            printf((codec->capabilities & CODEC_CAP_SLICE_THREADS) ? "S" : ".");
-            printf((codec->capabilities & CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
-            printf((codec->capabilities & CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
-            printf((codec->capabilities & CODEC_CAP_DR1)           ? "D" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_FRAME_THREADS) ? "F" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_SLICE_THREADS) ? "S" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_EXPERIMENTAL)  ? "X" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_DRAW_HORIZ_BAND)?"B" : ".");
+            printf((codec->capabilities & AV_CODEC_CAP_DR1)           ? "D" : ".");
 
             printf(" %-20s %s", codec->name, codec->long_name ? codec->long_name : "");
             if (strcmp(codec->name, desc->name))
@@ -1534,16 +1560,17 @@ int show_protocols(void *optctx, const char *opt, const char *arg)
     printf("Supported file protocols:\n"
            "Input:\n");
     while ((name = avio_enum_protocols(&opaque, 0)))
-        printf("%s\n", name);
+        printf("  %s\n", name);
     printf("Output:\n");
     while ((name = avio_enum_protocols(&opaque, 1)))
-        printf("%s\n", name);
+        printf("  %s\n", name);
     return 0;
 }
 
 int show_filters(void *optctx, const char *opt, const char *arg)
 {
-    const AVFilter av_unused(*filter) = NULL;
+#if CONFIG_AVFILTER
+    const AVFilter *filter = NULL;
     char descr[64], *descr_cur;
     int i, j;
     const AVFilterPad *pad;
@@ -1551,12 +1578,11 @@ int show_filters(void *optctx, const char *opt, const char *arg)
     printf("Filters:\n"
            "  T.. = Timeline support\n"
            "  .S. = Slice threading\n"
-           "  ..C = Commmand support\n"
+           "  ..C = Command support\n"
            "  A = Audio input/output\n"
            "  V = Video input/output\n"
            "  N = Dynamic number and/or type of input/output\n"
            "  | = Source or sink filter\n");
-#if CONFIG_AVFILTER
     while ((filter = avfilter_next(filter))) {
         descr_cur = descr;
         for (i = 0; i < 2; i++) {
@@ -1565,10 +1591,10 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                 *(descr_cur++) = '>';
             }
             pad = i ? filter->outputs : filter->inputs;
-            for (j = 0; pad && pad[j].name; j++) {
+            for (j = 0; pad && avfilter_pad_get_name(pad, j); j++) {
                 if (descr_cur >= descr + sizeof(descr) - 4)
                     break;
-                *(descr_cur++) = get_media_type_char(pad[j].type);
+                *(descr_cur++) = get_media_type_char(avfilter_pad_get_type(pad, j));
             }
             if (!j)
                 *(descr_cur++) = ((!i && (filter->flags & AVFILTER_FLAG_DYNAMIC_INPUTS)) ||
@@ -1581,6 +1607,8 @@ int show_filters(void *optctx, const char *opt, const char *arg)
                filter->process_command                        ? 'C' : '.',
                filter->name, descr, filter->description);
     }
+#else
+    printf("No filters available: libavfilter disabled\n");
 #endif
     return 0;
 }
@@ -1639,19 +1667,19 @@ int show_layouts(void *optctx, const char *opt, const char *arg)
     const char *name, *descr;
 
     printf("Individual channels:\n"
-           "NAME        DESCRIPTION\n");
+           "NAME           DESCRIPTION\n");
     for (i = 0; i < 63; i++) {
         name = av_get_channel_name((uint64_t)1 << i);
         if (!name)
             continue;
         descr = av_get_channel_description((uint64_t)1 << i);
-        printf("%-12s%s\n", name, descr);
+        printf("%-14s %s\n", name, descr);
     }
     printf("\nStandard channel layouts:\n"
-           "NAME        DECOMPOSITION\n");
+           "NAME           DECOMPOSITION\n");
     for (i = 0; !av_get_standard_channel_layout(i, &layout, &name); i++) {
         if (name) {
-            printf("%-12s", name);
+            printf("%-14s ", name);
             for (j = 1; j; j <<= 1)
                 if ((layout & j))
                     printf("%s%s", (layout & (j - 1)) ? "+" : "", av_get_channel_name(j));
@@ -1818,6 +1846,8 @@ int show_help(void *optctx, const char *opt, const char *arg)
     av_log_set_callback(log_callback_help);
 
     topic = av_strdup(arg ? arg : "");
+    if (!topic)
+        return AVERROR(ENOMEM);
     par = strchr(topic, '=');
     if (par)
         *par++ = 0;
@@ -1855,48 +1885,6 @@ int read_yesno(void)
     return yesno;
 }
 
-int cmdutils_read_file(const char *filename, char **bufptr, size_t *size)
-{
-    int ret;
-    FILE *f = av_fopen_utf8(filename, "rb");
-
-    if (!f) {
-        av_log(NULL, AV_LOG_ERROR, "Cannot read file '%s': %s\n", filename,
-               strerror(errno));
-        return AVERROR(errno);
-    }
-    fseek(f, 0, SEEK_END);
-    *size = ftell(f);
-    fseek(f, 0, SEEK_SET);
-    if (*size == (size_t)-1) {
-        av_log(NULL, AV_LOG_ERROR, "IO error: %s\n", strerror(errno));
-        fclose(f);
-        return AVERROR(errno);
-    }
-    *bufptr = av_malloc(*size + 1);
-    if (!*bufptr) {
-        av_log(NULL, AV_LOG_ERROR, "Could not allocate file buffer\n");
-        fclose(f);
-        return AVERROR(ENOMEM);
-    }
-    ret = fread(*bufptr, 1, *size, f);
-    if (ret < *size) {
-        av_free(*bufptr);
-        if (ferror(f)) {
-            av_log(NULL, AV_LOG_ERROR, "Error while reading file '%s': %s\n",
-                   filename, strerror(errno));
-            ret = AVERROR(errno);
-        } else
-            ret = AVERROR_EOF;
-    } else {
-        ret = 0;
-        (*bufptr)[(*size)++] = '\0';
-    }
-
-    fclose(f);
-    return ret;
-}
-
 FILE *get_preset_file(char *filename, size_t filename_size,
                       const char *preset_name, int is_path,
                       const char *codec_name)
@@ -1992,7 +1980,7 @@ AVDictionary *filter_codec_opts(AVDictionary *opts, enum AVCodecID codec_id,
             switch (check_stream_specifier(s, st, p + 1)) {
             case  1: *p = 0; break;
             case  0:         continue;
-            default:         return NULL;
+            default:         exit_program(1);
             }
 
         if (av_opt_find(&cc, t->key, NULL, flags, AV_OPT_SEARCH_FAKE_OBJ) ||
@@ -2039,7 +2027,7 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
         exit_program(1);
     }
     if (*size < new_size) {
-        uint8_t *tmp = av_realloc(array, new_size*elem_size);
+        uint8_t *tmp = av_realloc_array(array, new_size, elem_size);
         if (!tmp) {
             av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
             exit_program(1);
@@ -2050,3 +2038,189 @@ void *grow_array(void *array, int elem_size, int *size, int new_size)
     }
     return array;
 }
+
+double get_rotation(AVStream *st)
+{
+    AVDictionaryEntry *rotate_tag = av_dict_get(st->metadata, "rotate", NULL, 0);
+    uint8_t* displaymatrix = av_stream_get_side_data(st,
+                                                     AV_PKT_DATA_DISPLAYMATRIX, NULL);
+    double theta = 0;
+
+    if (rotate_tag && *rotate_tag->value && strcmp(rotate_tag->value, "0")) {
+        char *tail;
+        theta = av_strtod(rotate_tag->value, &tail);
+        if (*tail)
+            theta = 0;
+    }
+    if (displaymatrix && !theta)
+        theta = -av_display_rotation_get((int32_t*) displaymatrix);
+
+    theta -= 360*floor(theta/360 + 0.9/360);
+
+    if (fabs(theta - 90*round(theta/90)) > 2)
+        av_log(NULL, AV_LOG_WARNING, "Odd rotation angle.\n"
+               "If you want to help, upload a sample "
+               "of this file to ftp://upload.ffmpeg.org/incoming/ "
+               "and contact the ffmpeg-devel mailing list. (ffmpeg-devel@ffmpeg.org)");
+
+    return theta;
+}
+
+#if CONFIG_AVDEVICE
+static int print_device_sources(AVInputFormat *fmt, AVDictionary *opts)
+{
+    int ret, i;
+    AVDeviceInfoList *device_list = NULL;
+
+    if (!fmt || !fmt->priv_class  || !AV_IS_INPUT_DEVICE(fmt->priv_class->category))
+        return AVERROR(EINVAL);
+
+    printf("Auto-detected sources for %s:\n", fmt->name);
+    if (!fmt->get_device_list) {
+        ret = AVERROR(ENOSYS);
+        printf("Cannot list sources. Not implemented.\n");
+        goto fail;
+    }
+
+    if ((ret = avdevice_list_input_sources(fmt, NULL, opts, &device_list)) < 0) {
+        printf("Cannot list sources.\n");
+        goto fail;
+    }
+
+    for (i = 0; i < device_list->nb_devices; i++) {
+        printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
+               device_list->devices[i]->device_name, device_list->devices[i]->device_description);
+    }
+
+  fail:
+    avdevice_free_list_devices(&device_list);
+    return ret;
+}
+
+static int print_device_sinks(AVOutputFormat *fmt, AVDictionary *opts)
+{
+    int ret, i;
+    AVDeviceInfoList *device_list = NULL;
+
+    if (!fmt || !fmt->priv_class  || !AV_IS_OUTPUT_DEVICE(fmt->priv_class->category))
+        return AVERROR(EINVAL);
+
+    printf("Auto-detected sinks for %s:\n", fmt->name);
+    if (!fmt->get_device_list) {
+        ret = AVERROR(ENOSYS);
+        printf("Cannot list sinks. Not implemented.\n");
+        goto fail;
+    }
+
+    if ((ret = avdevice_list_output_sinks(fmt, NULL, opts, &device_list)) < 0) {
+        printf("Cannot list sinks.\n");
+        goto fail;
+    }
+
+    for (i = 0; i < device_list->nb_devices; i++) {
+        printf("%s %s [%s]\n", device_list->default_device == i ? "*" : " ",
+               device_list->devices[i]->device_name, device_list->devices[i]->device_description);
+    }
+
+  fail:
+    avdevice_free_list_devices(&device_list);
+    return ret;
+}
+
+static int show_sinks_sources_parse_arg(const char *arg, char **dev, AVDictionary **opts)
+{
+    int ret;
+    if (arg) {
+        char *opts_str = NULL;
+        av_assert0(dev && opts);
+        *dev = av_strdup(arg);
+        if (!*dev)
+            return AVERROR(ENOMEM);
+        if ((opts_str = strchr(*dev, ','))) {
+            *(opts_str++) = '\0';
+            if (opts_str[0] && ((ret = av_dict_parse_string(opts, opts_str, "=", ":", 0)) < 0)) {
+                av_freep(dev);
+                return ret;
+            }
+        }
+    } else
+        printf("\nDevice name is not provided.\n"
+                "You can pass devicename[,opt1=val1[,opt2=val2...]] as an argument.\n\n");
+    return 0;
+}
+
+int show_sources(void *optctx, const char *opt, const char *arg)
+{
+    AVInputFormat *fmt = NULL;
+    char *dev = NULL;
+    AVDictionary *opts = NULL;
+    int ret = 0;
+    int error_level = av_log_get_level();
+
+    av_log_set_level(AV_LOG_ERROR);
+
+    if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
+        goto fail;
+
+    do {
+        fmt = av_input_audio_device_next(fmt);
+        if (fmt) {
+            if (!strcmp(fmt->name, "lavfi"))
+                continue; //it's pointless to probe lavfi
+            if (dev && !av_match_name(dev, fmt->name))
+                continue;
+            print_device_sources(fmt, opts);
+        }
+    } while (fmt);
+    do {
+        fmt = av_input_video_device_next(fmt);
+        if (fmt) {
+            if (dev && !av_match_name(dev, fmt->name))
+                continue;
+            print_device_sources(fmt, opts);
+        }
+    } while (fmt);
+  fail:
+    av_dict_free(&opts);
+    av_free(dev);
+    av_log_set_level(error_level);
+    return ret;
+}
+
+int show_sinks(void *optctx, const char *opt, const char *arg)
+{
+    AVOutputFormat *fmt = NULL;
+    char *dev = NULL;
+    AVDictionary *opts = NULL;
+    int ret = 0;
+    int error_level = av_log_get_level();
+
+    av_log_set_level(AV_LOG_ERROR);
+
+    if ((ret = show_sinks_sources_parse_arg(arg, &dev, &opts)) < 0)
+        goto fail;
+
+    do {
+        fmt = av_output_audio_device_next(fmt);
+        if (fmt) {
+            if (dev && !av_match_name(dev, fmt->name))
+                continue;
+            print_device_sinks(fmt, opts);
+        }
+    } while (fmt);
+    do {
+        fmt = av_output_video_device_next(fmt);
+        if (fmt) {
+            if (dev && !av_match_name(dev, fmt->name))
+                continue;
+            print_device_sinks(fmt, opts);
+        }
+    } while (fmt);
+  fail:
+    av_dict_free(&opts);
+    av_free(dev);
+    av_log_set_level(error_level);
+    return ret;
+}
+
+#endif

cmdutils.h

@@ -46,7 +46,7 @@ extern const int program_birth_year;
 
 extern AVCodecContext *avcodec_opts[AVMEDIA_TYPE_NB];
 extern AVFormatContext *avformat_opts;
-extern struct SwsContext *sws_opts;
+extern AVDictionary *sws_dict;
 extern AVDictionary *swr_opts;
 extern AVDictionary *format_opts, *codec_opts, *resample_opts;
 extern int hide_banner;
@@ -61,6 +61,11 @@ void register_exit(void (*cb)(int ret));
  */
 void exit_program(int ret) av_noreturn;
 
+/**
+ * Initialize dynamic library loading
+ */
+void init_dynload(void);
+
 /**
  * Initialize the cmdutils option system, in particular
  * allocate the *_opts contexts.
@@ -277,7 +282,7 @@ typedef struct OptionGroup {
     AVDictionary *codec_opts;
     AVDictionary *format_opts;
     AVDictionary *resample_opts;
-    struct SwsContext *sws_opts;
+    AVDictionary *sws_dict;
     AVDictionary *swr_opts;
 } OptionGroup;
 
@@ -443,6 +448,20 @@ int show_formats(void *optctx, const char *opt, const char *arg);
  */
 int show_devices(void *optctx, const char *opt, const char *arg);
 
+#if CONFIG_AVDEVICE
+/**
+ * Print a listing containing autodetected sinks of the output device.
+ * Device name with options may be passed as an argument to limit results.
+ */
+int show_sinks(void *optctx, const char *opt, const char *arg);
+
+/**
+ * Print a listing containing autodetected sources of the input device.
+ * Device name with options may be passed as an argument to limit results.
+ */
+int show_sources(void *optctx, const char *opt, const char *arg);
+#endif
+
 /**
  * Print a listing containing all the codecs supported by the
  * program.
@@ -515,18 +534,6 @@ int show_colors(void *optctx, const char *opt, const char *arg);
  */
 int read_yesno(void);
 
-/**
- * Read the file with name filename, and put its content in a newly
- * allocated 0-terminated buffer.
- *
- * @param filename file to read from
- * @param bufptr location where pointer to buffer is returned
- * @param size   location where size of buffer is returned
- * @return >= 0 in case of success, a negative value corresponding to an
- * AVERROR error code in case of failure.
- */
-int cmdutils_read_file(const char *filename, char **bufptr, size_t *size);
-
 /**
  * Get a file corresponding to a preset file.
  *
@@ -583,4 +590,6 @@ void *grow_array(void *array, int elem_size, int *size, int new_size);
     char name[128];\
     av_get_channel_layout_string(name, sizeof(name), 0, ch_layout);
 
+double get_rotation(AVStream *st);
+
 #endif /* CMDUTILS_H */

cmdutils_common_opts.h

@@ -27,3 +27,9 @@
     { "opencl_bench", OPT_EXIT, {.func_arg = opt_opencl_bench}, "run benchmark on all OpenCL devices and show results" },
     { "opencl_options", HAS_ARG, {.func_arg = opt_opencl},      "set OpenCL environment options" },
 #endif
+#if CONFIG_AVDEVICE
+    { "sources"    , OPT_EXIT | HAS_ARG, { .func_arg = show_sources },
+      "list sources of the input device", "device" },
+    { "sinks"      , OPT_EXIT | HAS_ARG, { .func_arg = show_sinks },
+      "list sinks of the output device", "device" },
+#endif

cmdutils_opencl.c

@@ -22,6 +22,7 @@
 #include "libavutil/time.h"
 #include "libavutil/log.h"
 #include "libavutil/opencl.h"
+#include "libavutil/avstring.h"
 #include "cmdutils.h"
 
 typedef struct {
@@ -238,7 +239,8 @@ int opt_opencl_bench(void *optctx, const char *opt, const char *arg)
                 devices[count].platform_idx = i;
                 devices[count].device_idx = j;
                 devices[count].runtime = score;
-                strcpy(devices[count].device_name, device_node->device_name);
+                av_strlcpy(devices[count].device_name, device_node->device_name,
+                           sizeof(devices[count].device_name));
                 count++;
             }
         }

common.mak

@@ -5,6 +5,14 @@
 # first so "all" becomes default target
 all: all-yes
 
+DEFAULT_YASMD=.dbg
+
+ifeq ($(DBG),1)
+YASMD=$(DEFAULT_YASMD)
+else
+YASMD=
+endif
+
 ifndef SUBDIR
 
 ifndef V
@@ -110,8 +118,9 @@ TOOLOBJS  := $(TOOLS:%=tools/%.o)
 TOOLS     := $(TOOLS:%=tools/%$(EXESUF))
 HEADERS   += $(HEADERS-yes)
 
-PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(CONFIG_SHARED:yes=S)LIBNAME))
-DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
+PATH_LIBNAME = $(foreach NAME,$(1),lib$(NAME)/$($(2)LIBNAME))
+DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib),$(CONFIG_SHARED:yes=S)))
+STATIC_DEP_LIBS := $(foreach lib,$(FFLIBS),$(call PATH_LIBNAME,$(lib)))
 
 SRC_DIR    := $(SRC_PATH)/lib$(NAME)
 ALLHEADERS := $(subst $(SRC_DIR)/,$(SUBDIR),$(wildcard $(SRC_DIR)/*.h $(SRC_DIR)/$(ARCH)/*.h))
@@ -138,17 +147,17 @@ $(TOOLOBJS): | tools
 
 OBJDIRS := $(OBJDIRS) $(dir $(OBJS) $(HOBJS) $(HOSTOBJS) $(SLIBOBJS) $(TESTOBJS))
 
-CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda
+CLEANSUFFIXES     = *.d *.o *~ *.h.c *.map *.ver *.ho *.gcno *.gcda *$(DEFAULT_YASMD).asm
 DISTCLEANSUFFIXES = *.pc
 LIBSUFFIXES       = *.a *.lib *.so *.so.* *.dylib *.dll *.def *.dll.a
 
 define RULES
 clean::
-	$(RM) $(OBJS) $(OBJS:.o=.d)
+	$(RM) $(OBJS) $(OBJS:.o=.d) $(OBJS:.o=$(DEFAULT_YASMD).d)
 	$(RM) $(HOSTPROGS)
 	$(RM) $(TOOLS)
 endef
 
 $(eval $(RULES))
 
--include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d))
+-include $(wildcard $(OBJS:.o=.d) $(HOSTOBJS:.o=.d) $(TESTOBJS:.o=.d) $(HOBJS:.o=.d) $(SLIBOBJS:.o=.d)) $(OBJS:.o=$(DEFAULT_YASMD).d)

compat/avisynth/avisynth_c.h

@@ -38,40 +38,9 @@
 #ifndef __AVISYNTH_C__
 #define __AVISYNTH_C__
 
-#ifdef __cplusplus
-#  define EXTERN_C extern "C"
-#else
-#  define EXTERN_C
-#endif
-
-#define AVSC_USE_STDCALL 1
-
-#ifndef AVSC_USE_STDCALL
-#  define AVSC_CC __cdecl
-#else
-#  define AVSC_CC __stdcall
-#endif
-
-#define AVSC_INLINE static __inline
-
-#ifdef AVISYNTH_C_EXPORTS
-#  define AVSC_EXPORT EXTERN_C
-#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
-#else
-#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
-#  ifndef AVSC_NO_DECLSPEC
-#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
-#  else
-#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
-#  endif
-#endif
-
-typedef unsigned char BYTE;
-#ifdef __GNUC__
-typedef long long int INT64;
-#else
-typedef __int64 INT64;
-#endif
+#include "avs/config.h"
+#include "avs/capi.h"
+#include "avs/types.h"
 
 
 /////////////////////////////////////////////////////////////////////
@@ -79,8 +48,8 @@ typedef __int64 INT64;
 // Constants
 //
 
-#ifndef __AVISYNTH_H__
-enum { AVISYNTH_INTERFACE_VERSION = 4 };
+#ifndef __AVISYNTH_6_H__
+enum { AVISYNTH_INTERFACE_VERSION = 6 };
 #endif
 
 enum {AVS_SAMPLE_INT8  = 1<<0,
@@ -112,8 +81,8 @@ enum {AVS_CS_BGR = 1<<28,
       AVS_CS_PLANAR = 1<<31,
 
       AVS_CS_SHIFT_SUB_WIDTH   = 0,
-      AVS_CS_SHIFT_SUB_HEIGHT  = 1 << 3,
-      AVS_CS_SHIFT_SAMPLE_BITS = 1 << 4,
+      AVS_CS_SHIFT_SUB_HEIGHT  = 8,
+      AVS_CS_SHIFT_SAMPLE_BITS = 16,
 
       AVS_CS_SUB_WIDTH_MASK    = 7 << AVS_CS_SHIFT_SUB_WIDTH,
       AVS_CS_SUB_WIDTH_1       = 3 << AVS_CS_SHIFT_SUB_WIDTH, // YV24
@@ -180,15 +149,66 @@ enum {  //SUBTYPES
   AVS_FILTER_OUTPUT_TYPE_DIFFERENT=4};
 
 enum {
-  AVS_CACHE_NOTHING=0,
-  AVS_CACHE_RANGE=1,
-  AVS_CACHE_ALL=2,
-  AVS_CACHE_AUDIO=3,
-  AVS_CACHE_AUDIO_NONE=4,
-  AVS_CACHE_AUDIO_AUTO=5
+  // New 2.6 explicitly defined cache hints.
+  AVS_CACHE_NOTHING=10, // Do not cache video.
+  AVS_CACHE_WINDOW=11, // Hard protect upto X frames within a range of X from the current frame N.
+  AVS_CACHE_GENERIC=12, // LRU cache upto X frames.
+  AVS_CACHE_FORCE_GENERIC=13, // LRU cache upto X frames, override any previous CACHE_WINDOW.
+
+  AVS_CACHE_GET_POLICY=30, // Get the current policy.
+  AVS_CACHE_GET_WINDOW=31, // Get the current window h_span.
+  AVS_CACHE_GET_RANGE=32, // Get the current generic frame range.
+
+  AVS_CACHE_AUDIO=50, // Explicitly do cache audio, X byte cache.
+  AVS_CACHE_AUDIO_NOTHING=51, // Explicitly do not cache audio.
+  AVS_CACHE_AUDIO_NONE=52, // Audio cache off (auto mode), X byte intial cache.
+  AVS_CACHE_AUDIO_AUTO=53, // Audio cache on (auto mode), X byte intial cache.
+
+  AVS_CACHE_GET_AUDIO_POLICY=70, // Get the current audio policy.
+  AVS_CACHE_GET_AUDIO_SIZE=71, // Get the current audio cache size.
+
+  AVS_CACHE_PREFETCH_FRAME=100, // Queue request to prefetch frame N.
+  AVS_CACHE_PREFETCH_GO=101, // Action video prefetches.
+
+  AVS_CACHE_PREFETCH_AUDIO_BEGIN=120, // Begin queue request transaction to prefetch audio (take critical section).
+  AVS_CACHE_PREFETCH_AUDIO_STARTLO=121, // Set low 32 bits of start.
+  AVS_CACHE_PREFETCH_AUDIO_STARTHI=122, // Set high 32 bits of start.
+  AVS_CACHE_PREFETCH_AUDIO_COUNT=123, // Set low 32 bits of length.
+  AVS_CACHE_PREFETCH_AUDIO_COMMIT=124, // Enqueue request transaction to prefetch audio (release critical section).
+  AVS_CACHE_PREFETCH_AUDIO_GO=125, // Action audio prefetches.
+
+  AVS_CACHE_GETCHILD_CACHE_MODE=200, // Cache ask Child for desired video cache mode.
+  AVS_CACHE_GETCHILD_CACHE_SIZE=201, // Cache ask Child for desired video cache size.
+  AVS_CACHE_GETCHILD_AUDIO_MODE=202, // Cache ask Child for desired audio cache mode.
+  AVS_CACHE_GETCHILD_AUDIO_SIZE=203, // Cache ask Child for desired audio cache size.
+
+  AVS_CACHE_GETCHILD_COST=220, // Cache ask Child for estimated processing cost.
+    AVS_CACHE_COST_ZERO=221, // Child response of zero cost (ptr arithmetic only).
+    AVS_CACHE_COST_UNIT=222, // Child response of unit cost (less than or equal 1 full frame blit).
+    AVS_CACHE_COST_LOW=223, // Child response of light cost. (Fast)
+    AVS_CACHE_COST_MED=224, // Child response of medium cost. (Real time)
+    AVS_CACHE_COST_HI=225, // Child response of heavy cost. (Slow)
+
+  AVS_CACHE_GETCHILD_THREAD_MODE=240, // Cache ask Child for thread safetyness.
+    AVS_CACHE_THREAD_UNSAFE=241, // Only 1 thread allowed for all instances. 2.5 filters default!
+    AVS_CACHE_THREAD_CLASS=242, // Only 1 thread allowed for each instance. 2.6 filters default!
+    AVS_CACHE_THREAD_SAFE=243, //  Allow all threads in any instance.
+    AVS_CACHE_THREAD_OWN=244, // Safe but limit to 1 thread, internally threaded.
+
+  AVS_CACHE_GETCHILD_ACCESS_COST=260, // Cache ask Child for preferred access pattern.
+    AVS_CACHE_ACCESS_RAND=261, // Filter is access order agnostic.
+    AVS_CACHE_ACCESS_SEQ0=262, // Filter prefers sequential access (low cost)
+    AVS_CACHE_ACCESS_SEQ1=263, // Filter needs sequential access (high cost)
   };
 
-#define AVS_FRAME_ALIGN 16
+#ifdef BUILDING_AVSCORE
+struct AVS_ScriptEnvironment {
+        IScriptEnvironment * env;
+        const char * error;
+        AVS_ScriptEnvironment(IScriptEnvironment * e = 0)
+                : env(e), error(0) {}
+};
+#endif
 
 typedef struct AVS_Clip AVS_Clip;
 typedef struct AVS_ScriptEnvironment AVS_ScriptEnvironment;
@@ -238,29 +258,23 @@ AVSC_INLINE int avs_is_yuv(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_yuy2(const AVS_VideoInfo * p)
         { return (p->pixel_type & AVS_CS_YUY2) == AVS_CS_YUY2; }
 
-AVSC_INLINE int avs_is_yv24(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV24  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv24)(const AVS_VideoInfo * p);
 
-AVSC_INLINE int avs_is_yv16(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV16  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv16)(const AVS_VideoInfo * p);
 
-AVSC_INLINE int avs_is_yv12(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV12  & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv12)(const AVS_VideoInfo * p) ;
 
-AVSC_INLINE int avs_is_yv411(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_YV411 & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_yv411)(const AVS_VideoInfo * p);
 
-AVSC_INLINE int avs_is_y8(const AVS_VideoInfo * p)
-        { return (p->pixel_type & AVS_CS_PLANAR_MASK) == (AVS_CS_Y8    & AVS_CS_PLANAR_FILTER); }
+AVSC_API(int, avs_is_y8)(const AVS_VideoInfo * p);
 
 AVSC_INLINE int avs_is_property(const AVS_VideoInfo * p, int property)
-        { return ((p->pixel_type & property)==property ); }
+        { return ((p->image_type & property)==property ); }
 
 AVSC_INLINE int avs_is_planar(const AVS_VideoInfo * p)
         { return !!(p->pixel_type & AVS_CS_PLANAR); }
 
-AVSC_INLINE int avs_is_color_space(const AVS_VideoInfo * p, int c_space)
-        { return avs_is_planar(p) ? ((p->pixel_type & AVS_CS_PLANAR_MASK) == (c_space & AVS_CS_PLANAR_FILTER)) : ((p->pixel_type & c_space) == c_space); }
+AVSC_API(int, avs_is_color_space)(const AVS_VideoInfo * p, int c_space);
 
 AVSC_INLINE int avs_is_field_based(const AVS_VideoInfo * p)
         { return !!(p->image_type & AVS_IT_FIELDBASED); }
@@ -274,25 +288,18 @@ AVSC_INLINE int avs_is_bff(const AVS_VideoInfo * p)
 AVSC_INLINE int avs_is_tff(const AVS_VideoInfo * p)
         { return !!(p->image_type & AVS_IT_TFF); }
 
-AVSC_INLINE int avs_bits_per_pixel(const AVS_VideoInfo * p)
-{
-  switch (p->pixel_type) {
-      case AVS_CS_BGR24: return 24;
-      case AVS_CS_BGR32: return 32;
-      case AVS_CS_YUY2:  return 16;
-      case AVS_CS_YV12:
-      case AVS_CS_I420:  return 12;
-      default:           return 0;
-    }
-}
-AVSC_INLINE int avs_bytes_from_pixels(const AVS_VideoInfo * p, int pixels)
-        { return pixels * (avs_bits_per_pixel(p)>>3); }   // Will work on planar images, but will return only luma planes
+AVSC_API(int, avs_get_plane_width_subsampling)(const AVS_VideoInfo * p, int plane);
 
-AVSC_INLINE int avs_row_size(const AVS_VideoInfo * p)
-        { return avs_bytes_from_pixels(p,p->width); }  // Also only returns first plane on planar images
+AVSC_API(int, avs_get_plane_height_subsampling)(const AVS_VideoInfo * p, int plane);
 
-AVSC_INLINE int avs_bmp_size(const AVS_VideoInfo * vi)
-        { if (avs_is_planar(vi)) {int p = vi->height * ((avs_row_size(vi)+3) & ~3); p+=p>>1; return p;  } return vi->height * ((avs_row_size(vi)+3) & ~3); }
+
+AVSC_API(int, avs_bits_per_pixel)(const AVS_VideoInfo * p);
+
+AVSC_API(int, avs_bytes_from_pixels)(const AVS_VideoInfo * p, int pixels);
+
+AVSC_API(int, avs_row_size)(const AVS_VideoInfo * p, int plane);
+
+AVSC_API(int, avs_bmp_size)(const AVS_VideoInfo * vi);
 
 AVSC_INLINE int avs_samples_per_second(const AVS_VideoInfo * p)
         { return p->audio_samples_per_second; }
@@ -350,11 +357,13 @@ AVSC_INLINE void avs_set_fps(AVS_VideoInfo * p, unsigned numerator, unsigned den
     p->fps_denominator = denominator/x;
 }
 
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE int avs_is_same_colorspace(AVS_VideoInfo * x, AVS_VideoInfo * y)
 {
         return (x->pixel_type == y->pixel_type)
                 || (avs_is_yv12(x) && avs_is_yv12(y));
 }
+#endif
 
 /////////////////////////////////////////////////////////////////////
 //
@@ -391,89 +400,38 @@ typedef struct AVS_VideoFrame {
 } AVS_VideoFrame;
 
 // Access functions for AVS_VideoFrame
-AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
-        return p->pitch;}
+AVSC_API(int, avs_get_pitch_p)(const AVS_VideoFrame * p, int plane);
 
-AVSC_INLINE int avs_get_pitch_p(const AVS_VideoFrame * p, int plane) {
-  switch (plane) {
-  case AVS_PLANAR_U: case AVS_PLANAR_V: return p->pitchUV;}
-  return p->pitch;}
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE int avs_get_pitch(const AVS_VideoFrame * p) {
+        return avs_get_pitch_p(p, 0);}
+#endif
+
+AVSC_API(int, avs_get_row_size_p)(const AVS_VideoFrame * p, int plane);
 
 AVSC_INLINE int avs_get_row_size(const AVS_VideoFrame * p) {
         return p->row_size; }
 
-AVSC_INLINE int avs_get_row_size_p(const AVS_VideoFrame * p, int plane) {
-        int r;
-    switch (plane) {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-                if (p->pitchUV) return p->row_sizeUV;
-                else            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-                if (p->pitchUV) {
-                        r = (p->row_sizeUV+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                        if (r < p->pitchUV)
-                                return r;
-                        return p->row_sizeUV;
-                } else return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-                r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-                if (r <= p->pitch)
-                        return r;
-                return p->row_size;
-    }
-    return p->row_size;
-}
+AVSC_API(int, avs_get_height_p)(const AVS_VideoFrame * p, int plane);
 
 AVSC_INLINE int avs_get_height(const AVS_VideoFrame * p) {
         return p->height;}
 
-AVSC_INLINE int avs_get_height_p(const AVS_VideoFrame * p, int plane) {
-        switch (plane) {
-                case AVS_PLANAR_U: case AVS_PLANAR_V:
-                        if (p->pitchUV) return p->heightUV;
-                        return 0;
-        }
-        return p->height;}
+AVSC_API(const BYTE *, avs_get_read_ptr_p)(const AVS_VideoFrame * p, int plane);
 
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
 AVSC_INLINE const BYTE* avs_get_read_ptr(const AVS_VideoFrame * p) {
-        return p->vfb->data + p->offset;}
+        return avs_get_read_ptr_p(p, 0);}
+#endif
 
-AVSC_INLINE const BYTE* avs_get_read_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        switch (plane) {
-                case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                default:           return p->vfb->data + p->offset;}
-}
+AVSC_API(int, avs_is_writable)(const AVS_VideoFrame * p);
 
-AVSC_INLINE int avs_is_writable(const AVS_VideoFrame * p) {
-        return (p->refcount == 1 && p->vfb->refcount == 1);}
-
-AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p)
-{
-        if (avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else
-                return 0;
-}
-
-AVSC_INLINE BYTE* avs_get_write_ptr_p(const AVS_VideoFrame * p, int plane)
-{
-        if (plane==AVS_PLANAR_Y && avs_is_writable(p)) {
-                ++p->vfb->sequence_number;
-                return p->vfb->data + p->offset;
-        } else if (plane==AVS_PLANAR_Y) {
-                return 0;
-        } else {
-                switch (plane) {
-                        case AVS_PLANAR_U: return p->vfb->data + p->offsetU;
-                        case AVS_PLANAR_V: return p->vfb->data + p->offsetV;
-                        default:       return p->vfb->data + p->offset;
-                }
-        }
-}
+AVSC_API(BYTE *, avs_get_write_ptr_p)(const AVS_VideoFrame * p, int plane);
 
+#ifdef AVS_IMPLICIT_FUNCTION_DECLARATION_ERROR
+AVSC_INLINE BYTE* avs_get_write_ptr(const AVS_VideoFrame * p) {
+        return avs_get_write_ptr_p(p, 0);}
+#endif
 
 AVSC_API(void, avs_release_video_frame)(AVS_VideoFrame *);
 // makes a shallow copy of a video frame
@@ -658,12 +616,16 @@ enum {
   AVS_CPUF_SSSE3      = 0x200,   //  Core 2
   AVS_CPUF_SSE4       = 0x400,   //  Penryn, Wolfdale, Yorkfield
   AVS_CPUF_SSE4_1     = 0x400,
-  AVS_CPUF_SSE4_2     = 0x800,   //  Nehalem
+//AVS_CPUF_AVX        = 0x800,   //  Sandy Bridge, Bulldozer
+  AVS_CPUF_SSE4_2    = 0x1000,   //  Nehalem
+//AVS_CPUF_AVX2      = 0x2000,   //  Haswell
+//AVS_CPUF_AVX512    = 0x4000,   //  Knights Landing
 };
 
+
 AVSC_API(const char *, avs_get_error)(AVS_ScriptEnvironment *); // return 0 if no error
 
-AVSC_API(long, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
+AVSC_API(int, avs_get_cpu_flags)(AVS_ScriptEnvironment *);
 AVSC_API(int, avs_check_version)(AVS_ScriptEnvironment *, int version);
 
 AVSC_API(char *, avs_save_string)(AVS_ScriptEnvironment *, const char* s, int length);
@@ -700,12 +662,12 @@ AVSC_API(AVS_VideoFrame *, avs_new_video_frame_a)(AVS_ScriptEnvironment *,
 AVSC_INLINE
 AVS_VideoFrame * avs_new_video_frame(AVS_ScriptEnvironment * env,
                                      const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
 
 AVSC_INLINE
 AVS_VideoFrame * avs_new_frame(AVS_ScriptEnvironment * env,
                                const AVS_VideoInfo * vi)
-  {return avs_new_video_frame_a(env,vi,AVS_FRAME_ALIGN);}
+  {return avs_new_video_frame_a(env,vi,FRAME_ALIGN);}
 #endif
 
 
@@ -773,7 +735,6 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_function_exists);
   AVSC_DECLARE_FUNC(avs_get_audio);
   AVSC_DECLARE_FUNC(avs_get_cpu_flags);
-  AVSC_DECLARE_FUNC(avs_get_error);
   AVSC_DECLARE_FUNC(avs_get_frame);
   AVSC_DECLARE_FUNC(avs_get_parity);
   AVSC_DECLARE_FUNC(avs_get_var);
@@ -798,6 +759,27 @@ struct AVS_Library {
   AVSC_DECLARE_FUNC(avs_subframe_planar);
   AVSC_DECLARE_FUNC(avs_take_clip);
   AVSC_DECLARE_FUNC(avs_vsprintf);
+
+  AVSC_DECLARE_FUNC(avs_get_error);
+  AVSC_DECLARE_FUNC(avs_is_yv24);
+  AVSC_DECLARE_FUNC(avs_is_yv16);
+  AVSC_DECLARE_FUNC(avs_is_yv12);
+  AVSC_DECLARE_FUNC(avs_is_yv411);
+  AVSC_DECLARE_FUNC(avs_is_y8);
+  AVSC_DECLARE_FUNC(avs_is_color_space);
+
+  AVSC_DECLARE_FUNC(avs_get_plane_width_subsampling);
+  AVSC_DECLARE_FUNC(avs_get_plane_height_subsampling);
+  AVSC_DECLARE_FUNC(avs_bits_per_pixel);
+  AVSC_DECLARE_FUNC(avs_bytes_from_pixels);
+  AVSC_DECLARE_FUNC(avs_row_size);
+  AVSC_DECLARE_FUNC(avs_bmp_size);
+  AVSC_DECLARE_FUNC(avs_get_pitch_p);
+  AVSC_DECLARE_FUNC(avs_get_row_size_p);
+  AVSC_DECLARE_FUNC(avs_get_height_p);
+  AVSC_DECLARE_FUNC(avs_get_read_ptr_p);
+  AVSC_DECLARE_FUNC(avs_is_writable);
+  AVSC_DECLARE_FUNC(avs_get_write_ptr_p);
 };
 
 #undef AVSC_DECLARE_FUNC
@@ -832,7 +814,6 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   AVSC_LOAD_FUNC(avs_function_exists);
   AVSC_LOAD_FUNC(avs_get_audio);
   AVSC_LOAD_FUNC(avs_get_cpu_flags);
-  AVSC_LOAD_FUNC(avs_get_error);
   AVSC_LOAD_FUNC(avs_get_frame);
   AVSC_LOAD_FUNC(avs_get_parity);
   AVSC_LOAD_FUNC(avs_get_var);
@@ -858,6 +839,27 @@ AVSC_INLINE AVS_Library * avs_load_library() {
   AVSC_LOAD_FUNC(avs_take_clip);
   AVSC_LOAD_FUNC(avs_vsprintf);
 
+  AVSC_LOAD_FUNC(avs_get_error);
+  AVSC_LOAD_FUNC(avs_is_yv24);
+  AVSC_LOAD_FUNC(avs_is_yv16);
+  AVSC_LOAD_FUNC(avs_is_yv12);
+  AVSC_LOAD_FUNC(avs_is_yv411);
+  AVSC_LOAD_FUNC(avs_is_y8);
+  AVSC_LOAD_FUNC(avs_is_color_space);
+
+  AVSC_LOAD_FUNC(avs_get_plane_width_subsampling);
+  AVSC_LOAD_FUNC(avs_get_plane_height_subsampling);
+  AVSC_LOAD_FUNC(avs_bits_per_pixel);
+  AVSC_LOAD_FUNC(avs_bytes_from_pixels);
+  AVSC_LOAD_FUNC(avs_row_size);
+  AVSC_LOAD_FUNC(avs_bmp_size);
+  AVSC_LOAD_FUNC(avs_get_pitch_p);
+  AVSC_LOAD_FUNC(avs_get_row_size_p);
+  AVSC_LOAD_FUNC(avs_get_height_p);
+  AVSC_LOAD_FUNC(avs_get_read_ptr_p);
+  AVSC_LOAD_FUNC(avs_is_writable);
+  AVSC_LOAD_FUNC(avs_get_write_ptr_p);
+
 #undef __AVSC_STRINGIFY
 #undef AVSC_STRINGIFY
 #undef AVSC_LOAD_FUNC

compat/avisynth/avisynth_c_25.h

@@ -1,68 +0,0 @@
-//  Copyright (c) 2011 FFmpegSource Project
-//
-//  Permission is hereby granted, free of charge, to any person obtaining a copy
-//  of this software and associated documentation files (the "Software"), to deal
-//  in the Software without restriction, including without limitation the rights
-//  to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
-//  copies of the Software, and to permit persons to whom the Software is
-//  furnished to do so, subject to the following conditions:
-//
-//  The above copyright notice and this permission notice shall be included in
-//  all copies or substantial portions of the Software.
-//
-//  THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-//  IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-//  FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
-//  AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
-//  LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
-//  OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
-//  THE SOFTWARE.
-
-/* these are defines/functions that are used and were changed in the switch to 2.6
- * and are needed to maintain full compatility with 2.5 */
-
-enum {
-  AVS_CS_YV12_25 = 1<<3 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-v-u, planar
-  AVS_CS_I420_25 = 1<<4 | AVS_CS_YUV | AVS_CS_PLANAR,  // y-u-v, planar
-};
-
-AVSC_INLINE int avs_get_height_p_25(const AVS_VideoFrame * p, int plane) {
-    switch (plane)
-    {
-        case AVS_PLANAR_U: case AVS_PLANAR_V:
-            if (p->pitchUV)
-                return p->height>>1;
-            return 0;
-    }
-    return p->height;}
-
-AVSC_INLINE int avs_get_row_size_p_25(const AVS_VideoFrame * p, int plane) {
-    int r;
-    switch (plane)
-    {
-    case AVS_PLANAR_U: case AVS_PLANAR_V:
-        if (p->pitchUV)
-            return p->row_size>>1;
-        else
-            return 0;
-    case AVS_PLANAR_U_ALIGNED: case AVS_PLANAR_V_ALIGNED:
-        if (p->pitchUV)
-        {
-            r = ((p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)) )>>1; // Aligned rowsize
-            if (r < p->pitchUV)
-                return r;
-            return p->row_size>>1;
-        }
-        else
-            return 0;
-    case AVS_PLANAR_Y_ALIGNED:
-        r = (p->row_size+AVS_FRAME_ALIGN-1)&(~(AVS_FRAME_ALIGN-1)); // Aligned rowsize
-        if (r <= p->pitch)
-            return r;
-        return p->row_size;
-    }
-    return p->row_size;
-}
-
-AVSC_INLINE int avs_is_yv12_25(const AVS_VideoInfo * p)
-    { return ((p->pixel_type & AVS_CS_YV12_25) == AVS_CS_YV12_25)||((p->pixel_type & AVS_CS_I420_25) == AVS_CS_I420_25); }

compat/avisynth/avs/capi.h

@@ -0,0 +1,62 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CAPI_H
+#define AVS_CAPI_H
+
+#ifdef __cplusplus
+#  define EXTERN_C extern "C"
+#else
+#  define EXTERN_C
+#endif
+
+#ifndef AVSC_USE_STDCALL
+#  define AVSC_CC __cdecl
+#else
+#  define AVSC_CC __stdcall
+#endif
+
+#define AVSC_INLINE static __inline
+
+#ifdef BUILDING_AVSCORE
+#  define AVSC_EXPORT EXTERN_C
+#  define AVSC_API(ret, name) EXTERN_C __declspec(dllexport) ret AVSC_CC name
+#else
+#  define AVSC_EXPORT EXTERN_C __declspec(dllexport)
+#  ifndef AVSC_NO_DECLSPEC
+#    define AVSC_API(ret, name) EXTERN_C __declspec(dllimport) ret AVSC_CC name
+#  else
+#    define AVSC_API(ret, name) typedef ret (AVSC_CC *name##_func)
+#  endif
+#endif
+
+#endif //AVS_CAPI_H

compat/avisynth/avs/config.h

@@ -0,0 +1,55 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_CONFIG_H
+#define AVS_CONFIG_H
+
+// Undefine this to get cdecl calling convention
+#define AVSC_USE_STDCALL 1
+
+// NOTE TO PLUGIN AUTHORS:
+// Because FRAME_ALIGN can be substantially higher than the alignment
+// a plugin actually needs, plugins should not use FRAME_ALIGN to check for
+// alignment. They should always request the exact alignment value they need.
+// This is to make sure that plugins work over the widest range of AviSynth
+// builds possible.
+#define FRAME_ALIGN 32
+
+#if   defined(_M_AMD64) || defined(__x86_64)
+#   define X86_64
+#elif defined(_M_IX86) || defined(__i386__)
+#   define X86_32
+#else
+#   error Unsupported CPU architecture.
+#endif
+
+#endif //AVS_CONFIG_H

compat/avisynth/avs/types.h

@@ -0,0 +1,51 @@
+// Avisynth C Interface Version 0.20
+// Copyright 2003 Kevin Atkinson
+
+// This program is free software; you can redistribute it and/or modify
+// it under the terms of the GNU General Public License as published by
+// the Free Software Foundation; either version 2 of the License, or
+// (at your option) any later version.
+//
+// This program is distributed in the hope that it will be useful,
+// but WITHOUT ANY WARRANTY; without even the implied warranty of
+// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+// GNU General Public License for more details.
+//
+// You should have received a copy of the GNU General Public License
+// along with this program; if not, write to the Free Software
+// Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA, or visit
+// http://www.gnu.org/copyleft/gpl.html .
+//
+// As a special exception, I give you permission to link to the
+// Avisynth C interface with independent modules that communicate with
+// the Avisynth C interface solely through the interfaces defined in
+// avisynth_c.h, regardless of the license terms of these independent
+// modules, and to copy and distribute the resulting combined work
+// under terms of your choice, provided that every copy of the
+// combined work is accompanied by a complete copy of the source code
+// of the Avisynth C interface and Avisynth itself (with the version
+// used to produce the combined work), being distributed under the
+// terms of the GNU General Public License plus this exception.  An
+// independent module is a module which is not derived from or based
+// on Avisynth C Interface, such as 3rd-party filters, import and
+// export plugins, or graphical user interfaces.
+
+#ifndef AVS_TYPES_H
+#define AVS_TYPES_H
+
+// Define all types necessary for interfacing with avisynth.dll
+
+// Raster types used by VirtualDub & Avisynth
+typedef unsigned int    Pixel32;
+typedef unsigned char   BYTE;
+
+// Audio Sample information
+typedef float SFLOAT;
+
+#ifdef __GNUC__
+typedef long long int INT64;
+#else
+typedef __int64 INT64;
+#endif
+
+#endif //AVS_TYPES_H

compat/avisynth/avxsynth_c.h

@@ -513,21 +513,21 @@ AVSC_INLINE AVS_Value avs_array_elt(AVS_Value v, int index)
 // only use these functions on am AVS_Value that does not already have
 // an active value.  Remember, treat AVS_Value as a fat pointer.
 AVSC_INLINE AVS_Value avs_new_value_bool(int v0)
-        { AVS_Value v; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
+        { AVS_Value v = {0}; v.type = 'b'; v.d.boolean = v0 == 0 ? 0 : 1; return v; }
 AVSC_INLINE AVS_Value avs_new_value_int(int v0)
-        { AVS_Value v; v.type = 'i'; v.d.integer = v0; return v; }
+        { AVS_Value v = {0}; v.type = 'i'; v.d.integer = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_string(const char * v0)
-        { AVS_Value v; v.type = 's'; v.d.string = v0; return v; }
+        { AVS_Value v = {0}; v.type = 's'; v.d.string = v0; return v; }
 AVSC_INLINE AVS_Value avs_new_value_float(float v0)
-        { AVS_Value v; v.type = 'f'; v.d.floating_pt = v0; return v;}
+        { AVS_Value v = {0}; v.type = 'f'; v.d.floating_pt = v0; return v;}
 AVSC_INLINE AVS_Value avs_new_value_error(const char * v0)
-        { AVS_Value v; v.type = 'e'; v.d.string = v0; return v; }
+        { AVS_Value v = {0}; v.type = 'e'; v.d.string = v0; return v; }
 #ifndef AVSC_NO_DECLSPEC
 AVSC_INLINE AVS_Value avs_new_value_clip(AVS_Clip * v0)
-        { AVS_Value v; avs_set_to_clip(&v, v0); return v; }
+        { AVS_Value v = {0}; avs_set_to_clip(&v, v0); return v; }
 #endif
 AVSC_INLINE AVS_Value avs_new_value_array(AVS_Value * v0, int size)
-        { AVS_Value v; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
+        { AVS_Value v = {0}; v.type = 'a'; v.d.array = v0; v.array_size = size; return v; }
 
 /////////////////////////////////////////////////////////////////////
 //

compat/avisynth/windowsPorts/windows2linux.h

@@ -52,8 +52,8 @@ namespace avxsynth {
 //
 // Functions
 //
-#define MAKEDWORD(a,b,c,d) ((a << 24) | (b << 16) | (c << 8) | (d))
-#define MAKEWORD(a,b) ((a << 8) | (b))
+#define MAKEDWORD(a,b,c,d) (((a) << 24) | ((b) << 16) | ((c) << 8) | (d))
+#define MAKEWORD(a,b) (((a) << 8) | (b))
 
 #define lstrlen                             strlen
 #define lstrcpy                             strcpy

compat/getopt.c

@@ -54,7 +54,7 @@ static int getopt(int argc, char *argv[], char *opts)
         }
     }
     optopt = c = argv[optind][sp];
-    if (c == ':' || (cp = strchr(opts, c)) == NULL) {
+    if (c == ':' || !(cp = strchr(opts, c))) {
         fprintf(stderr, ": illegal option -- %c\n", c);
         if (argv[optind][++sp] == '\0') {
             optind++;

compat/os2threads.h

@@ -66,6 +66,8 @@ static av_always_inline int pthread_create(pthread_t *thread, const pthread_attr
     struct thread_arg *thread_arg;
 
     thread_arg = av_mallocz(sizeof(struct thread_arg));
+    if (!thread_arg)
+        return ENOMEM;
 
     thread_arg->start_routine = start_routine;
     thread_arg->arg = arg;

compat/w32pthreads.h

@@ -39,6 +39,7 @@
 #include <windows.h>
 #include <process.h>
 
+#include "libavutil/attributes.h"
 #include "libavutil/common.h"
 #include "libavutil/internal.h"
 #include "libavutil/mem.h"
@@ -54,36 +55,30 @@ typedef struct pthread_t {
  * not mutexes */
 typedef CRITICAL_SECTION pthread_mutex_t;
 
-/* This is the CONDITIONAL_VARIABLE typedef for using Window's native
- * conditional variables on kernels 6.0+.
- * MinGW does not currently have this typedef. */
-typedef struct pthread_cond_t {
-    void *ptr;
-} pthread_cond_t;
-
-/* function pointers to conditional variable API on windows 6.0+ kernels */
-#if _WIN32_WINNT < 0x0600
-static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
-static void (WINAPI *cond_init)(pthread_cond_t *cond);
-static void (WINAPI *cond_signal)(pthread_cond_t *cond);
-static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
-                                DWORD milliseconds);
+/* This is the CONDITION_VARIABLE typedef for using Windows' native
+ * conditional variables on kernels 6.0+. */
+#if HAVE_CONDITION_VARIABLE_PTR
+typedef CONDITION_VARIABLE pthread_cond_t;
 #else
-#define cond_init      InitializeConditionVariable
-#define cond_broadcast WakeAllConditionVariable
-#define cond_signal    WakeConditionVariable
-#define cond_wait      SleepConditionVariableCS
+typedef struct pthread_cond_t {
+    void *Ptr;
+} pthread_cond_t;
 #endif
 
-static unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
+#if _WIN32_WINNT >= 0x0600
+#define InitializeCriticalSection(x) InitializeCriticalSectionEx(x, 0, 0)
+#define WaitForSingleObject(a, b) WaitForSingleObjectEx(a, b, FALSE)
+#endif
+
+static av_unused unsigned __stdcall attribute_align_arg win32thread_worker(void *arg)
 {
     pthread_t *h = arg;
     h->ret = h->func(h->arg);
     return 0;
 }
 
-static int pthread_create(pthread_t *thread, const void *unused_attr,
-                          void *(*start_routine)(void*), void *arg)
+static av_unused int pthread_create(pthread_t *thread, const void *unused_attr,
+                                    void *(*start_routine)(void*), void *arg)
 {
     thread->func   = start_routine;
     thread->arg    = arg;
@@ -92,7 +87,7 @@ static int pthread_create(pthread_t *thread, const void *unused_attr,
     return !thread->handle;
 }
 
-static void pthread_join(pthread_t thread, void **value_ptr)
+static av_unused void pthread_join(pthread_t thread, void **value_ptr)
 {
     DWORD ret = WaitForSingleObject(thread.handle, INFINITE);
     if (ret != WAIT_OBJECT_0)
@@ -123,6 +118,36 @@ static inline int pthread_mutex_unlock(pthread_mutex_t *m)
     return 0;
 }
 
+#if _WIN32_WINNT >= 0x0600
+static inline int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+{
+    InitializeConditionVariable(cond);
+    return 0;
+}
+
+/* native condition variables do not destroy */
+static inline void pthread_cond_destroy(pthread_cond_t *cond)
+{
+    return;
+}
+
+static inline void pthread_cond_broadcast(pthread_cond_t *cond)
+{
+    WakeAllConditionVariable(cond);
+}
+
+static inline int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+{
+    SleepConditionVariableCS(cond, mutex, INFINITE);
+    return 0;
+}
+
+static inline void pthread_cond_signal(pthread_cond_t *cond)
+{
+    WakeConditionVariable(cond);
+}
+
+#else // _WIN32_WINNT < 0x0600
 /* for pre-Windows 6.0 platforms we need to define and use our own condition
  * variable and api */
 typedef struct  win32_cond_t {
@@ -134,7 +159,14 @@ typedef struct  win32_cond_t {
     volatile int is_broadcast;
 } win32_cond_t;
 
-static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
+/* function pointers to conditional variable API on windows 6.0+ kernels */
+static void (WINAPI *cond_broadcast)(pthread_cond_t *cond);
+static void (WINAPI *cond_init)(pthread_cond_t *cond);
+static void (WINAPI *cond_signal)(pthread_cond_t *cond);
+static BOOL (WINAPI *cond_wait)(pthread_cond_t *cond, pthread_mutex_t *mutex,
+                                DWORD milliseconds);
+
+static av_unused int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
 {
     win32_cond_t *win32_cond = NULL;
     if (cond_init) {
@@ -146,7 +178,7 @@ static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
     win32_cond = av_mallocz(sizeof(win32_cond_t));
     if (!win32_cond)
         return ENOMEM;
-    cond->ptr = win32_cond;
+    cond->Ptr = win32_cond;
     win32_cond->semaphore = CreateSemaphore(NULL, 0, 0x7fffffff, NULL);
     if (!win32_cond->semaphore)
         return ENOMEM;
@@ -159,9 +191,9 @@ static int pthread_cond_init(pthread_cond_t *cond, const void *unused_attr)
     return 0;
 }
 
-static void pthread_cond_destroy(pthread_cond_t *cond)
+static av_unused void pthread_cond_destroy(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->ptr;
+    win32_cond_t *win32_cond = cond->Ptr;
     /* native condition variables do not destroy */
     if (cond_init)
         return;
@@ -172,12 +204,12 @@ static void pthread_cond_destroy(pthread_cond_t *cond)
     pthread_mutex_destroy(&win32_cond->mtx_waiter_count);
     pthread_mutex_destroy(&win32_cond->mtx_broadcast);
     av_freep(&win32_cond);
-    cond->ptr = NULL;
+    cond->Ptr = NULL;
 }
 
-static void pthread_cond_broadcast(pthread_cond_t *cond)
+static av_unused void pthread_cond_broadcast(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->ptr;
+    win32_cond_t *win32_cond = cond->Ptr;
     int have_waiter;
 
     if (cond_broadcast) {
@@ -206,9 +238,9 @@ static void pthread_cond_broadcast(pthread_cond_t *cond)
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
 
-static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
+static av_unused int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
 {
-    win32_cond_t *win32_cond = cond->ptr;
+    win32_cond_t *win32_cond = cond->Ptr;
     int last_waiter;
     if (cond_wait) {
         cond_wait(cond, mutex, INFINITE);
@@ -238,9 +270,9 @@ static int pthread_cond_wait(pthread_cond_t *cond, pthread_mutex_t *mutex)
     return pthread_mutex_lock(mutex);
 }
 
-static void pthread_cond_signal(pthread_cond_t *cond)
+static av_unused void pthread_cond_signal(pthread_cond_t *cond)
 {
-    win32_cond_t *win32_cond = cond->ptr;
+    win32_cond_t *win32_cond = cond->Ptr;
     int have_waiter;
     if (cond_signal) {
         cond_signal(cond);
@@ -262,8 +294,9 @@ static void pthread_cond_signal(pthread_cond_t *cond)
 
     pthread_mutex_unlock(&win32_cond->mtx_broadcast);
 }
+#endif
 
-static void w32thread_init(void)
+static av_unused void w32thread_init(void)
 {
 #if _WIN32_WINNT < 0x0600
     HANDLE kernel_dll = GetModuleHandle(TEXT("kernel32.dll"));

compat/windows/mslink

@@ -0,0 +1,9 @@
+#!/bin/sh
+
+LINK_EXE_PATH=$(dirname "$(command -v cl)")/link
+if [ -x "$LINK_EXE_PATH" ]; then
+    "$LINK_EXE_PATH" $@
+else
+    link $@
+fi
+exit $?

doc/APIchanges

@@ -2,19 +2,315 @@ Never assume the API of libav* to be stable unless at least 1 month has passed
 since the last major version increase or the API was added.
 
 The last version increases were:
-libavcodec:    2013-03-xx
-libavdevice:   2013-03-xx
-libavfilter:   2013-12-xx
-libavformat:   2013-03-xx
-libavresample: 2012-10-05
-libpostproc:   2011-04-18
-libswresample: 2011-09-19
-libswscale:    2011-06-20
-libavutil:     2012-10-22
+libavcodec:    2014-08-09
+libavdevice:   2014-08-09
+libavfilter:   2014-08-09
+libavformat:   2014-08-09
+libavresample: 2014-08-09
+libpostproc:   2014-08-09
+libswresample: 2014-08-09
+libswscale:    2014-08-09
+libavutil:     2014-08-09
 
 
 API changes, most recent first:
 
+-------- 8< --------- FFmpeg 2.8 was cut here -------- 8< ---------
+
+2015-08-27 - 1dd854e1 - lavc 56.58.100 - vaapi.h
+  Deprecate old VA-API context (vaapi_context) fields that were only
+  set and used by libavcodec. They are all managed internally now.
+
+2015-08-19 - 9f8e57ef - lavu 54.31.100 - pixfmt.h
+  Add a unique pixel format for VA-API (AV_PIX_FMT_VAAPI) that
+  indicates the nature of the underlying storage: a VA surface. This
+  yields the same value as AV_PIX_FMT_VAAPI_VLD.
+  Deprecate old VA-API related pixel formats: AV_PIX_FMT_VAAPI_MOCO,
+  AV_PIX_FMT_VAAPI_IDCT, AV_PIX_FMT_VAAPI_VLD.
+
+2015-08-02 - lavu 54.30.100 / 54.17.0
+  9ed59f1 / 7a7df34c -  Add av_blowfish_alloc().
+  a130ec9 / ae365453 -  Add av_rc4_alloc().
+  9ca1997 / 5d8bea3b -  Add av_xtea_alloc().
+  3cf08e9 / d9e8b47e -  Add av_des_alloc().
+
+2015-07-27 - lavc 56.56.100 / 56.35.0 - avcodec.h
+  94d68a4 / 7c6eb0a1 - Rename CODEC_FLAG* defines to AV_CODEC_FLAG*.
+  444e987 / def97856 - Rename CODEC_CAP_* defines to AV_CODEC_CAP_*.
+  29d147c / 059a9348 - Rename FF_INPUT_BUFFER_PADDING_SIZE and FF_MIN_BUFFER_SIZE
+              to AV_INPUT_BUFFER_PADDING_SIZE and AV_INPUT_BUFFER_MIN_SIZE.
+
+2015-07-22 - c40ecff - lavc 56.51.100 - avcodec.h
+  Add AV_PKT_DATA_QUALITY_STATS to export the quality value, PSNR, and pict_type
+  of an AVPacket.
+
+2015-07-16 - 8dad213 - lavc 56.49.100
+  Add av_codec_get_codec_properties(), FF_CODEC_PROPERTY_LOSSLESS
+  and FF_CODEC_PROPERTY_CLOSED_CAPTIONS
+
+2015-07-03 - d563e13 / 83212943 - lavu 54.28.100 / 56.15.0
+  Add av_version_info().
+
+-------- 8< --------- FFmpeg 2.7 was cut here -------- 8< ---------
+
+2015-06-04 - cc17b43 - lswr  1.2.100
+  Add swr_get_out_samples()
+
+2015-05-27 - c312bfa - lavu 54.26.100 - cpu.h
+  Add AV_CPU_FLAG_AVXSLOW.
+
+2015-05-26 - 1fb9b2a - lavu 54.25.100 - rational.h
+  Add av_q2intfloat().
+
+2015-05-13 - cc48409 / e7c5e17 - lavc 56.39.100 / 56.23.0
+  Add av_vda_default_init2.
+
+2015-05-11 - 541d75f - lavf 56.33.100 - avformat.h
+  Add AVOpenCallback AVFormatContext.open_cb
+
+2015-05-07 - a7dd933 - 56.38.100 - avcodec.h
+  Add av_packet_side_data_name().
+
+2015-05-07 - 01e59d4 - 56.37.102 - avcodec.h
+  Add FF_PROFILE_VP9_2 and FF_PROFILE_VP9_3.
+
+2015-05-04 - 079b7f6 - 56.37.100 - avcodec.h
+  Add FF_PROFILE_VP9_0 and FF_PROFILE_VP9_1.
+
+2015-04-22 - 748d481 - lavf 56.31.100 - avformat.h
+  Add AVFMT_FLAG_FAST_SEEK flag. Some formats (initially mp3) use it to enable
+  fast, but inaccurate seeking.
+
+2015-04-20 - 8e8219e / c253340 - lavu 54.23.100 / 54.12.0 - log.h
+  Add AV_LOG_TRACE for extremely verbose debugging.
+
+2015-04-02 - 26e0e393 - lavf 56.29.100 - avio.h
+  Add AVIODirEntryType.AVIO_ENTRY_SERVER.
+  Add AVIODirEntryType.AVIO_ENTRY_SHARE.
+  Add AVIODirEntryType.AVIO_ENTRY_WORKGROUP.
+
+2015-03-31 - 3188696 - lavu 54.22.100 - avstring.h
+  Add av_append_path_component()
+
+2015-03-27 - 184084c - lavf 56.27.100 - avio.h url.h
+  New directory listing API.
+
+  Add AVIODirEntryType enum.
+  Add AVIODirEntry, AVIODirContext structures.
+  Add avio_open_dir(), avio_read_dir(), avio_close_dir(), avio_free_directory_entry().
+  Add ff_alloc_dir_entry().
+  Extend URLProtocol with url_open_dir(), url_read_dir(), url_close_dir().
+
+2015-03-29 - 268ff17 / c484561 - lavu 54.21.100 / 54.10.0 - pixfmt.h
+  Add AV_PIX_FMT_MMAL for MMAL hardware acceleration.
+
+2015-03-19 - 11fe56c - 56.29.100 / lavc 56.22.0
+  Add FF_PROFILE_DTS_EXPRESS.
+
+-------- 8< --------- FFmpeg 2.6 was cut here -------- 8< ---------
+
+2015-03-04 - cca4476 - lavf 56.25.100
+  Add avformat_flush()
+
+2015-03-03 - 81a9126 - lavf 56.24.100
+  Add avio_put_str16be()
+
+2015-02-19 - 560eb71 / 31d2039 - lavc 56.23.100 / 56.13.0
+  Add width, height, coded_width, coded_height and format to
+  AVCodecParserContext.
+
+2015-02-19 - e375511 / 5b1d9ce - lavu 54.19.100 / 54.9.0
+  Add AV_PIX_FMT_QSV for QSV hardware acceleration.
+
+2015-02-14 - ba22295 - lavc 56.21.102
+  Deprecate VIMA decoder.
+
+2015-01-27 - 62a82c6 / 728685f - lavc 56.21.100 / 56.12.0, lavu 54.18.100 / 54.8.0 - avcodec.h, frame.h
+  Add AV_PKT_DATA_AUDIO_SERVICE_TYPE and AV_FRAME_DATA_AUDIO_SERVICE_TYPE for
+  storing the audio service type as side data.
+
+2015-01-16 - a47c933 - lavf 56.19.100 - avformat.h
+  Add data_codec and data_codec_id for storing codec of data stream
+
+2015-01-11 - 007c33d - lavd 56.4.100 - avdevice.h
+  Add avdevice_list_input_sources().
+  Add avdevice_list_output_sinks().
+
+2014-12-25 - d7aaeea / c220a60 - lavc 56.19.100 / 56.10.0 - vdpau.h
+  Add av_vdpau_get_surface_parameters().
+
+2014-12-25 - ddb9a24 / 6c99c92 - lavc 56.18.100 / 56.9.0 - avcodec.h
+  Add AV_HWACCEL_FLAG_ALLOW_HIGH_DEPTH flag to av_vdpau_bind_context().
+
+2014-12-25 - d16079a / 57b6704 - lavc 56.17.100 / 56.8.0 - avcodec.h
+  Add AVCodecContext.sw_pix_fmt.
+
+2014-12-04 - 6e9ac02 - lavc 56.14.100 - dv_profile.h
+  Add av_dv_codec_profile2().
+
+-------- 8< --------- FFmpeg 2.5 was cut here -------- 8< ---------
+
+2014-11-21 - ab922f9 - lavu 54.15.100 - dict.h
+   Add av_dict_get_string().
+
+2014-11-18 - a54a51c - lavu 54.14.100 - float_dsp.h
+  Add avpriv_float_dsp_alloc().
+
+2014-11-16 - 6690d4c3 - lavf 56.13.100 - avformat.h
+  Add AVStream.recommended_encoder_configuration with accessors.
+
+2014-11-16 - bee5844d - lavu 54.13.100 - opt.h
+  Add av_opt_serialize().
+
+2014-11-16 - eec69332 - lavu 54.12.100 - opt.h
+  Add av_opt_is_set_to_default().
+
+2014-11-06 - 44fa267 / 5e80fb7 - lavc 56.11.100 / 56.6.0 - vorbis_parser.h
+  Add a public API for parsing vorbis packets.
+
+2014-10-15 - 17085a0 / 7ea1b34 - lavc 56.7.100 / 56.5.0 - avcodec.h
+  Replace AVCodecContext.time_base used for decoding
+  with AVCodecContext.framerate.
+
+2014-10-15 - 51c810e / d565fef1 - lavc 56.6.100 / 56.4.0 - avcodec.h
+  Add AV_HWACCEL_FLAG_IGNORE_LEVEL flag to av_vdpau_bind_context().
+
+2014-10-13 - da21895 / 2df0c32e - lavc 56.5.100 / 56.3.0 - avcodec.h
+  Add AVCodecContext.initial_padding. Deprecate the use of AVCodecContext.delay
+  for audio encoding.
+
+2014-10-08 - bb44f7d / 5a419b2 - lavu 54.10.100 / 54.4.0 - pixdesc.h
+  Add API to return the name of frame and context color properties.
+
+2014-10-06 - a61899a / e3e158e - lavc 56.3.100 / 56.2.0 - vdpau.h
+  Add av_vdpau_bind_context(). This function should now be used for creating
+  (or resetting) a AVVDPAUContext instead of av_vdpau_alloc_context().
+
+2014-10-02 - cdd6f05 - lavc 56.2.100 - avcodec.h
+2014-10-02 - cdd6f05 - lavu 54.9.100 - frame.h
+  Add AV_FRAME_DATA_SKIP_SAMPLES. Add lavc CODEC_FLAG2_SKIP_MANUAL and
+  AVOption "skip_manual", which makes lavc export skip information via
+  AV_FRAME_DATA_SKIP_SAMPLES AVFrame side data, instead of skipping and
+  discarding samples automatically.
+
+2014-10-02 - 0d92b0d - lavu 54.8.100 - avstring.h
+  Add av_match_list()
+
+2014-09-24 - ac68295 - libpostproc 53.1.100
+  Add visualization support
+
+2014-09-19 - 6edd6a4 - lavc 56.1.101 - dv_profile.h
+  deprecate avpriv_dv_frame_profile2(), which was made public by accident.
+
+
+-------- 8< --------- FFmpeg 2.4 was cut here -------- 8< ---------
+
+2014-08-25 - 215db29 / b263f8f - lavf 56.3.100 / 56.3.0 - avformat.h
+  Add AVFormatContext.max_ts_probe.
+
+2014-08-28 - f30a815 / 9301486 - lavc 56.1.100 / 56.1.0 - avcodec.h
+  Add AV_PKT_DATA_STEREO3D to export container-level stereo3d information.
+
+2014-08-23 - 8fc9bd0 - lavu 54.7.100 - dict.h
+  AV_DICT_DONT_STRDUP_KEY and AV_DICT_DONT_STRDUP_VAL arguments are now
+  freed even on error. This is consistent with the behaviour all users
+  of it we could find expect.
+
+2014-08-21 - 980a5b0 - lavu 54.6.100 - frame.h motion_vector.h
+  Add AV_FRAME_DATA_MOTION_VECTORS side data and AVMotionVector structure
+
+2014-08-16 - b7d5e01 - lswr 1.1.100 - swresample.h
+  Add AVFrame based API
+
+2014-08-16 - c2829dc - lavu 54.4.100 - dict.h
+  Add av_dict_set_int helper function.
+
+2014-08-13 - c8571c6 / 8ddc326 - lavu 54.3.100 / 54.3.0 - mem.h
+  Add av_strndup().
+
+2014-08-13 - 2ba4577 / a8c104a - lavu 54.2.100 / 54.2.0 - opt.h
+  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 - 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.
+
+2014-08-10 - 78eaaa8 / fb1ddcd - lavr 2.1.0 - avresample.h
+  Add avresample_convert_frame() and avresample_config().
+
+2014-08-10 - 78eaaa8 / fb1ddcd - lavu 54.1.100 / 54.1.0 - error.h
+  Add AVERROR_INPUT_CHANGED and AVERROR_OUTPUT_CHANGED.
+
+2014-08-08 - 3841f2a / d35b94f - lavc 55.73.102 / 55.57.4 - avcodec.h
+  Deprecate FF_IDCT_XVIDMMX define and xvidmmx idct option.
+  Replaced by FF_IDCT_XVID and xvid respectively.
+
+2014-08-08 - 5c3c671 - lavf 55.53.100 - avio.h
+  Add avio_feof() and deprecate url_feof().
+
+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
+  reordered_opaque is not going to be removed in the future.
+
+2014-08-02 - 28a2107 - lavu 52.98.100 - pixelutils.h
+  Add pixelutils API with SAD functions
+
+2014-08-04 - 6017c98 / e9abafc - lavu 52.97.100 / 53.22.0 - pixfmt.h
+  Add AV_PIX_FMT_YA16 pixel format for 16 bit packed gray with alpha.
+
+2014-08-04 - 4c8bc6f / e96c3b8 - lavu 52.96.101 / 53.21.1 - avstring.h
+  Rename AV_PIX_FMT_Y400A to AV_PIX_FMT_YA8 to better identify the format.
+  An alias pixel format and color space name are provided for compatibility.
+
+2014-08-04 - 073c074 / d2962e9 - lavu 52.96.100 / 53.21.0 - pixdesc.h
+  Support name aliases for pixel formats.
+
+2014-08-03 - 71d008e / 1ef9e83 - lavc 55.72.101 / 55.57.2 - avcodec.h
+2014-08-03 - 71d008e / 1ef9e83 - lavu 52.95.100 / 53.20.0 - frame.h
+  Deprecate AVCodecContext.dtg_active_format and use side-data instead.
+
+2014-08-03 - e680c73 - lavc 55.72.100 - avcodec.h
+  Add get_pixels() to AVDCT
+
+2014-08-03 - 9400603 / 9f17685 - lavc 55.71.101 / 55.57.1 - avcodec.h
+  Deprecate unused FF_IDCT_IPP define and ipp avcodec option.
+  Deprecate unused FF_DEBUG_PTS define and pts avcodec option.
+  Deprecate unused FF_CODER_TYPE_DEFLATE define and deflate avcodec option.
+  Deprecate unused FF_DCT_INT define and int avcodec option.
+  Deprecate unused avcodec option scenechange_factor.
+
+2014-07-30 - ba3e331 - lavu 52.94.100 - frame.h
+  Add av_frame_side_data_name()
+
+2014-07-29 - 80a3a66 / 3a19405 - lavf 56.01.100 / 56.01.0 - avformat.h
+  Add mime_type field to AVProbeData, which now MUST be initialized in
+  order to avoid uninitialized reads of the mime_type pointer, likely
+  leading to crashes.
+  Typically, this means you will do 'AVProbeData pd = { 0 };' instead of
+  'AVProbeData pd;'.
+
+2014-07-29 - 31e0b5d / 69e7336 - lavu 52.92.100 / 53.19.0 - avstring.h
+  Make name matching function from lavf public as av_match_name().
+
+2014-07-28 - 2e5c8b0 / c5fca01 - lavc 55.71.100 / 55.57.0 - avcodec.h
+  Add AV_CODEC_PROP_REORDER to mark codecs supporting frame reordering.
+
+2014-07-27 - ff9a154 - lavf 55.50.100 - avformat.h
+  New field int64_t probesize2 instead of deprecated
+  field int probesize.
+
+2014-07-27 - 932ff70 - lavc 55.70.100 - avdct.h
+  Add AVDCT / avcodec_dct_alloc() / avcodec_dct_init().
+
+2014-07-23 - 8a4c086 - lavf 55.49.100 - avio.h
+  Add avio_read_to_bprint()
+
+
+-------- 8< --------- FFmpeg 2.3 was cut here -------- 8< ---------
+
 2014-07-14 - 62227a7 - lavf 55.47.100 - avformat.h
   Add av_stream_get_parser()
 
@@ -35,7 +331,11 @@ API changes, most recent first:
   is now setting AVStream.time_base, instead of AVStream.codec.time_base as was
   done previously. The old method is now deprecated.
 
-2014-06-10 - xxxxxxx - lavf 55.43.100 - avformat.h
+2014-06-11 - 67d29da - lavc 55.66.101 - avcodec.h
+  Increase FF_INPUT_BUFFER_PADDING_SIZE to 32 due to some corner cases needing
+  it
+
+2014-06-10 - 5482780 - lavf 55.43.100 - avformat.h
   New field int64_t max_analyze_duration2 instead of deprecated
   int max_analyze_duration.
 
@@ -55,28 +355,28 @@ API changes, most recent first:
 2014-05-29 - bdb2e80 / b2d4565 - lavr 1.3.0 - avresample.h
   Add avresample_max_output_samples
 
-2014-05-24 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
+2014-05-28 - d858ee7 / 6d21259 - lavf 55.42.100 / 55.19.0 - avformat.h
   Add strict_std_compliance and related AVOptions to support experimental
   muxing.
 
-2014-05-26 - xxxxxxx - lavu 52.87.100 - threadmessage.h
+2014-05-26 - 55cc60c - lavu 52.87.100 - threadmessage.h
   Add thread message queue API.
 
 2014-05-26 - c37d179 - lavf 55.41.100 - avformat.h
   Add format_probesize to AVFormatContext.
 
-2014-05-19 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
+2014-05-20 - 7d25af1 / c23c96b - lavf 55.39.100 / 55.18.0 - avformat.h
   Add av_stream_get_side_data() to access stream-level side data
   in the same way as av_packet_get_side_data().
 
-2014-05-xx - xxxxxxx - lavu 52.86.100 - fifo.h
+2014-05-20 - 7336e39 - lavu 52.86.100 - fifo.h
   Add av_fifo_alloc_array() function.
 
 2014-05-19 - ef1d4ee / bddd8cb - lavu 52.85.100 / 53.15.0 - frame.h, display.h
   Add AV_FRAME_DATA_DISPLAYMATRIX for exporting frame-level
   spatial rendering on video frames for proper display.
 
-2014-05-xx - xxxxxxx - lavc 55.64.100 / 55.53.0 - avcodec.h
+2014-05-19 - ef1d4ee / bddd8cb - lavc 55.64.100 / 55.53.0 - avcodec.h
   Add AV_PKT_DATA_DISPLAYMATRIX for exporting packet-level
   spatial rendering on video frames for proper display.
 
@@ -88,11 +388,11 @@ API changes, most recent first:
   Add avcodec_free_context(). From now on it should be used for freeing
   AVCodecContext.
 
-2014-05-17 - 0eec06e - lavu 52.84.100 - time.h
+2014-05-17 - 0eec06e / 1bd0bdc - lavu 52.84.100 / 54.5.0 - time.h
   Add av_gettime_relative() av_gettime_relative_is_monotonic()
 
 2014-05-15 - eacf7d6 / 0c1959b - lavf 55.38.100 / 55.17.0 - avformat.h
-  Add AVMFT_FLAG_BITEXACT flag. Muxers now use it instead of checking
+  Add AVFMT_FLAG_BITEXACT flag. Muxers now use it instead of checking
   CODEC_FLAG_BITEXACT on the first stream.
 
 2014-05-15 - 96cb4c8 - lswr 0.19.100 - swresample.h
@@ -101,10 +401,10 @@ API changes, most recent first:
 2014-05-11 - 14aef38 / 66e6c8a - lavu 52.83.100 / 53.14.0 - pixfmt.h
   Add AV_PIX_FMT_VDA for new-style VDA acceleration.
 
-2014-05-xx - xxxxxxx - lavu 52.82.0 - fifo.h
+2014-05-07 - 351f611 - lavu 52.82.100 - fifo.h
   Add av_fifo_freep() function.
 
-2014-05-02 - ba52fb11 - lavu 52.81.0 - opt.h
+2014-05-02 - ba52fb11 - lavu 52.81.100 - opt.h
   Add av_opt_set_dict2() function.
 
 2014-05-01 - e77b985 / a2941c8 - lavc 55.60.103 / 55.50.3 - avcodec.h
@@ -123,10 +423,14 @@ API changes, most recent first:
   Deprecate CODEC_FLAG_INPUT_PRESERVED. Its functionality is replaced by passing
   reference-counted frames to encoders.
 
+2014-04-30 - 617e866 - lavu 52.81.100 - pixdesc.h
+  Add av_find_best_pix_fmt_of_2(), av_get_pix_fmt_loss()
+  Deprecate avcodec_get_pix_fmt_loss(), avcodec_find_best_pix_fmt_of_2()
+
 2014-04-29 - 1bf6396 - lavc 55.60.100 - avcodec.h
   Add AVCodecDescriptor.mime_types field.
 
-2014-04-29 - xxxxxxx - lavu 52.80.0 - hash.h
+2014-04-29 - b804eb4 - lavu 52.80.100 - hash.h
   Add av_hash_final_bin(), av_hash_final_hex() and av_hash_final_b64().
 
 2014-03-07 - 8b2a130 - lavc 55.50.0 / 55.53.100 - dxva2.h
@@ -138,7 +442,7 @@ API changes, most recent first:
 2014-04-17 - a8d01a7 / 0983d48 - lavu 53.12.0 / 52.77.100 - crc.h
   Add AV_CRC_16_ANSI_LE crc variant.
 
-2014-04-XX - xxxxxxx - lavf xx.xx.1xx - avformat.h
+2014-04-15 - ef818d8 - lavf 55.37.101 - avformat.h
   Add av_format_inject_global_side_data()
 
 2014-04-12 - 4f698be - lavu 52.76.100 - log.h
@@ -184,6 +488,9 @@ API changes, most recent first:
   Give the name AVPacketSideData to the previously anonymous struct used for
   AVPacket.side_data.
 
+
+-------- 8< --------- FFmpeg 2.2 was cut here -------- 8< ---------
+
 2014-03-18 - 37c07d4 - lsws 2.5.102
   Make gray16 full-scale.
 
@@ -215,7 +522,7 @@ API changes, most recent first:
 2014-02-19 - f4c8d00 / 6bb8720 - lavu 52.64.101 / 53.3.1 - opt.h
   Deprecate unused AV_OPT_FLAG_METADATA.
 
-2014-02-xx - xxxxxxx - lavd 55.10.100 - avdevice.h
+2014-02-16 - 81c3f81 - lavd 55.10.100 - avdevice.h
   Add avdevice_list_devices() and avdevice_free_list_devices()
 
 2014-02-16 - db3c970 - lavf 55.33.100 - avio.h
@@ -256,7 +563,7 @@ API changes, most recent first:
 2014-01-19 - 1a193c4 - lavf 55.25.100 - avformat.h
   Add avformat_get_mov_video_tags() and avformat_get_mov_audio_tags().
 
-2014-01-19 - xxxxxxx - lavu 52.63.100 - rational.h
+2014-01-19 - 3532dd5 - lavu 52.63.100 - rational.h
   Add av_make_q() function.
 
 2014-01-05 - 4cf4da9 / 5b4797a - lavu 52.62.100 / 53.2.0 - frame.h
@@ -326,6 +633,9 @@ API changes, most recent first:
 2013-10-31 - 78265fc / 28096e0 - lavu 52.49.100 / 52.17.0 - frame.h
   Add AVFrame.flags and AV_FRAME_FLAG_CORRUPT.
 
+
+-------- 8< --------- FFmpeg 2.1 was cut here -------- 8< ---------
+
 2013-10-27 - dbe6f9f - lavc 55.39.100 - avcodec.h
   Add CODEC_CAP_DELAY support to avcodec_decode_subtitle2.
 
@@ -398,6 +708,9 @@ API changes, most recent first:
   Add avcodec_chroma_pos_to_enum()
   Add avcodec_enum_to_chroma_pos()
 
+
+-------- 8< --------- FFmpeg 2.0 was cut here -------- 8< ---------
+
 2013-07-03 - 838bd73 - lavfi 3.78.100 - avfilter.h
   Deprecate avfilter_graph_parse() in favor of the equivalent
   avfilter_graph_parse_ptr().
@@ -415,6 +728,9 @@ API changes, most recent first:
    av_ripemd_update()
    av_ripemd_final()
 
+2013-06-10 - 82ef670 - lavu 52.35.101 - hmac.h
+  Add AV_HMAC_SHA224, AV_HMAC_SHA256, AV_HMAC_SHA384, AV_HMAC_SHA512
+
 2013-06-04 - 30b491f / fc962d4 - lavu 52.35.100 / 52.13.0 - mem.h
   Add av_realloc_array and av_reallocp_array
 
@@ -470,6 +786,9 @@ API changes, most recent first:
 2013-03-17 - 7aa9af5 - lavu 52.20.100 - opt.h
   Add AV_OPT_TYPE_VIDEO_RATE value to AVOptionType enum.
 
+
+-------- 8< --------- FFmpeg 1.2 was cut here -------- 8< ---------
+
 2013-03-07 - 9767ec6 - lavu 52.18.100 - avstring.h,bprint.h
   Add av_escape() and av_bprint_escape() API.
 
@@ -482,6 +801,9 @@ API changes, most recent first:
 2013-01-01 - 2eb2e17 - lavfi 3.34.100
   Add avfilter_get_audio_buffer_ref_from_arrays_channels.
 
+
+-------- 8< --------- FFmpeg 1.1 was cut here -------- 8< ---------
+
 2012-12-20 - 34de47aa - lavfi 3.29.100 - avfilter.h
   Add AVFilterLink.channels, avfilter_link_get_channels()
   and avfilter_ref_get_channels().
@@ -527,6 +849,9 @@ API changes, most recent first:
   Add LIBSWRESAMPLE_VERSION, LIBSWRESAMPLE_BUILD
   and LIBSWRESAMPLE_IDENT symbols.
 
+
+-------- 8< --------- FFmpeg 1.0 was cut here -------- 8< ---------
+
 2012-09-06 - 29e972f - lavu 51.72.100 - parseutils.h
   Add av_small_strptime() time parsing function.
 
@@ -731,6 +1056,9 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
   avresample_read() are now uint8_t** instead of void**.
   Libavresample is now stable.
 
+2012-09-26 - 3ba0dab7 / 1384df64 - lavf 54.29.101 / 56.06.3 - avformat.h
+  Add AVFormatContext.avoid_negative_ts.
+
 2012-09-24 - 46a3595 / a42aada - lavc 54.59.100 / 54.28.0 - avcodec.h
   Add avcodec_free_frame(). This function must now
   be used for freeing an AVFrame.
@@ -945,6 +1273,9 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2012-01-12 - b18e17e / 3167dc9 - lavfi 2.59.100 / 2.15.0
   Add a new installed header -- libavfilter/version.h -- with version macros.
 
+
+-------- 8< --------- FFmpeg 0.9 was cut here -------- 8< ---------
+
 2011-12-08 - a502939 - lavfi 2.52.0
   Add av_buffersink_poll_frame() to buffersink.h.
 
@@ -1154,6 +1485,13 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2011-06-28 - 5129336 - lavu 51.11.0 - avutil.h
   Define the AV_PICTURE_TYPE_NONE value in AVPictureType enum.
 
+
+-------- 8< --------- FFmpeg 0.7 was cut here -------- 8< ---------
+
+
+
+-------- 8< --------- FFmpeg 0.8 was cut here -------- 8< ---------
+
 2011-06-19 - fd2c0a5 - lavfi 2.23.0 - avfilter.h
   Add layout negotiation fields and helper functions.
 
@@ -1831,6 +2169,9 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-06-02 - 7e566bb - lavc 52.73.0 - av_get_codec_tag_string()
   Add av_get_codec_tag_string().
 
+
+-------- 8< --------- FFmpeg 0.6 was cut here -------- 8< ---------
+
 2010-06-01 - 2b99142 - lsws 0.11.0 - convertPalette API
   Add sws_convertPalette8ToPacked32() and sws_convertPalette8ToPacked24().
 
@@ -1848,10 +2189,6 @@ lavd 54.4.100 / 54.0.0, lavfi 3.5.0
 2010-05-09 - b6bc205 - lavfi 1.20.0 - AVFilterPicRef
   Add interlaced and top_field_first fields to AVFilterPicRef.
 
-------------------------------8<-------------------------------------
-                   0.6 branch was cut here
------------------------------>8--------------------------------------
-
 2010-05-01 - 8e2ee18 - lavf 52.62.0 - probe function
   Add av_probe_input_format2 to API, it allows ignoring probe
   results below given score and returns the actual probe score.

doc/Doxyfile

@@ -31,7 +31,7 @@ PROJECT_NAME           = FFmpeg
 # This could be handy for archiving the generated documentation or
 # if some version control system is used.
 
-PROJECT_NUMBER         =
+PROJECT_NUMBER         = 2.8.19
 
 # With the PROJECT_LOGO tag one can specify a logo or icon that is included
 # in the documentation. The maximum height of the logo should not exceed 55

doc/Makefile

@@ -36,14 +36,18 @@ DOCS-$(CONFIG_MANPAGES)  += $(MANPAGES)
 DOCS-$(CONFIG_TXTPAGES)  += $(TXTPAGES)
 DOCS = $(DOCS-yes)
 
+DOC_EXAMPLES-$(CONFIG_AVIO_DIR_CMD_EXAMPLE)      += avio_dir_cmd
 DOC_EXAMPLES-$(CONFIG_AVIO_READING_EXAMPLE)      += avio_reading
 DOC_EXAMPLES-$(CONFIG_AVCODEC_EXAMPLE)           += avcodec
+DOC_EXAMPLES-$(CONFIG_DECODING_ENCODING_EXAMPLE) += decoding_encoding
 DOC_EXAMPLES-$(CONFIG_DEMUXING_DECODING_EXAMPLE) += demuxing_decoding
+DOC_EXAMPLES-$(CONFIG_EXTRACT_MVS_EXAMPLE)       += extract_mvs
 DOC_EXAMPLES-$(CONFIG_FILTER_AUDIO_EXAMPLE)      += filter_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_AUDIO_EXAMPLE)   += filtering_audio
 DOC_EXAMPLES-$(CONFIG_FILTERING_VIDEO_EXAMPLE)   += filtering_video
 DOC_EXAMPLES-$(CONFIG_METADATA_EXAMPLE)          += metadata
 DOC_EXAMPLES-$(CONFIG_MUXING_EXAMPLE)            += muxing
+DOC_EXAMPLES-$(CONFIG_QSVDEC_EXAMPLE)            += qsvdec
 DOC_EXAMPLES-$(CONFIG_REMUXING_EXAMPLE)          += remuxing
 DOC_EXAMPLES-$(CONFIG_RESAMPLING_AUDIO_EXAMPLE)  += resampling_audio
 DOC_EXAMPLES-$(CONFIG_SCALING_VIDEO_EXAMPLE)     += scaling_video
@@ -80,14 +84,25 @@ $(GENTEXI): doc/avoptions_%.texi: doc/print_options$(HOSTEXESUF)
 	$(M)doc/print_options $* > $@
 
 doc/%.html: TAG = HTML
+doc/%-all.html: TAG = HTML
+
+ifdef HAVE_MAKEINFO_HTML
+doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
+	$(Q)$(TEXIDEP)
+	$(M)makeinfo --html -I doc --no-split -D config-not-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
+
+doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.pm $(GENTEXI)
+	$(Q)$(TEXIDEP)
+	$(M)makeinfo --html -I doc --no-split -D config-all --init-file=$(SRC_PATH)/doc/t2h.pm --output $@ $<
+else
 doc/%.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-not-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
 
-doc/%-all.html: TAG = HTML
 doc/%-all.html: doc/%.texi $(SRC_PATH)/doc/t2h.init $(GENTEXI)
 	$(Q)$(TEXIDEP)
 	$(M)texi2html -I doc -monolithic --D=config-all --init-file $(SRC_PATH)/doc/t2h.init --output $@ $<
+endif
 
 doc/%.pod: TAG = POD
 doc/%.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
@@ -101,9 +116,9 @@ doc/%-all.pod: doc/%.texi $(SRC_PATH)/doc/texi2pod.pl $(GENTEXI)
 
 doc/%.1 doc/%.3: TAG = MAN
 doc/%.1: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=1 --center=" " --release=" " $< > $@
+	$(M)pod2man --section=1 --center=" " --release=" " --date=" " $< > $@
 doc/%.3: doc/%.pod $(GENTEXI)
-	$(M)pod2man --section=3 --center=" " --release=" " $< > $@
+	$(M)pod2man --section=3 --center=" " --release=" " --date=" " $< > $@
 
 $(DOCS) doc/doxy/html: | doc/
 $(DOC_EXAMPLES:%$(EXESUF)=%.o): | doc/examples

doc/bitstream_filters.texi

@@ -13,7 +13,16 @@ bitstream filter using the option @code{--disable-bsf=BSF}.
 The option @code{-bsfs} of the ff* tools will display the list of
 all the supported bitstream filters included in your build.
 
-Below is a description of the currently available bitstream filters.
+The ff* tools have a -bsf option applied per stream, taking a
+comma-separated list of filters, whose parameters follow the filter
+name after a '='.
+
+@example
+ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1/opt2=str2][,filter2] OUTPUT
+@end example
+
+Below is a description of the currently available bitstream filters,
+with their parameters, if any.
 
 @section aac_adtstoasc
 
@@ -130,14 +139,41 @@ ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
 
 @section mp3_header_decompress
 
+@section mpeg4_unpack_bframes
+
+Unpack DivX-style packed B-frames.
+
+DivX-style packed B-frames are not valid MPEG-4 and were only a
+workaround for the broken Video for Windows subsystem.
+They use more space, can cause minor AV sync issues, require more
+CPU power to decode (unless the player has some decoded picture queue
+to compensate the 2,0,2,0 frame per packet style) and cause
+trouble if copied into a standard container like mp4 or mpeg-ps/ts,
+because MPEG-4 decoders may not be able to decode them, since they are
+not valid MPEG-4.
+
+For example to fix an AVI file containing an MPEG-4 stream with
+DivX-style packed B-frames using @command{ffmpeg}, you can use the command:
+
+@example
+ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
+@end example
+
 @section noise
 
 Damages the contents of packets without damaging the container. Can be
 used for fuzzing or testing error resilience/concealment.
 
+Parameters:
+A numeral string, whose value is related to how often output bytes will
+be modified. Therefore, values below or equal to 0 are forbidden, and
+the lower the more frequent bytes will be modified, with 1 meaning
+every byte is modified.
+
 @example
-ffmpeg -i INPUT -c copy -bsf noise output.mkv
+ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
 @end example
+applies the modification to every byte.
 
 @section remove_extra
 

doc/build_system.txt

@@ -7,10 +7,18 @@ V
     Disable the default terse mode, the full command issued by make and its
     output will be shown on the screen.
 
+DBG
+    Preprocess x86 external assembler files to a .dbg.asm file in the object
+    directory, which then gets compiled. Helps developping those assembler
+    files.
+
 DESTDIR
     Destination directory for the install targets, useful to prepare packages
     or install FFmpeg in cross-environments.
 
+GEN
+    Set to ‘1’ to generate the missing or mismatched references.
+
 Makefile targets:
 
 all

doc/codecs.texi

@@ -7,7 +7,7 @@ all the encoders and decoders. In addition each codec may support
 so-called private options, which are specific for a given codec.
 
 Sometimes, a global option may only affect a specific kind of codec,
-and may be unsensical or ignored by another, so you need to be aware
+and may be nonsensical or ignored by another, so you need to be aware
 of the meaning of the specified options. Also some options are
 meant only for decoding or encoding.
 
@@ -71,7 +71,9 @@ Force low delay.
 @item global_header
 Place global headers in extradata instead of every keyframe.
 @item bitexact
-Use only bitexact stuff (except (I)DCT).
+Only write platform-, build- and time-independent data. (except (I)DCT).
+This ensures that file and data checksums are reproducible and match between
+platforms. Its primary use is for regression testing.
 @item aic
 Apply H263 advanced intra coding / mpeg4 ac prediction.
 @item cbp
@@ -473,6 +475,9 @@ per-block quantization parameter (QP)
 motion vector
 @item dct_coeff
 
+@item green_metadata
+display complexity metadata for the upcoming frame, GoP or for a given duration.
+
 @item skip
 
 @item startcode
@@ -493,11 +498,15 @@ visualize block types
 picture buffer allocations
 @item thread_ops
 threading operations
+@item nomc
+skip motion compensation
 @end table
 
 @item vismv @var{integer} (@emph{decoding,video})
 Visualize motion vectors (MVs).
 
+This option is deprecated, see the codecview filter instead.
+
 Possible values:
 @table @samp
 @item pf
@@ -797,6 +806,9 @@ Frame data might be split into multiple chunks.
 Show all frames before the first keyframe.
 @item skiprd
 Deprecated, use mpegvideo private options instead.
+@item export_mvs
+Export motion vectors into frame side-data (see @code{AV_FRAME_DATA_MOTION_VECTORS})
+for codecs that support it. See also @file{doc/examples/export_mvs.c}.
 @end table
 
 @item error @var{integer} (@emph{encoding,video})
@@ -856,6 +868,14 @@ Possible values:
 
 @item mpeg2_aac_he
 
+@item mpeg4_sp
+
+@item mpeg4_core
+
+@item mpeg4_main
+
+@item mpeg4_asp
+
 @item dts
 
 @item dts_es
@@ -1025,7 +1045,11 @@ Possible values:
 @item color_primaries @var{integer} (@emph{decoding/encoding,video})
 @item color_trc @var{integer} (@emph{decoding/encoding,video})
 @item colorspace @var{integer} (@emph{decoding/encoding,video})
+
 @item color_range @var{integer} (@emph{decoding/encoding,video})
+If used as input parameter, it serves as a hint to the decoder, which
+color_range the input has.
+
 @item chroma_sample_location @var{integer} (@emph{decoding/encoding,video})
 
 @item log_level_offset @var{integer}
@@ -1109,6 +1133,19 @@ Interlaced video, bottom coded first, top displayed first
 Set to 1 to disable processing alpha (transparency). This works like the
 @samp{gray} flag in the @option{flags} option which skips chroma information
 instead of alpha. Default is 0.
+
+@item codec_whitelist @var{list} (@emph{input})
+"," separated List of allowed decoders. By default all are allowed.
+
+@item dump_separator @var{string} (@emph{input})
+Separator used to separate the fields printed on the command line about the
+Stream parameters.
+For example to separate the fields with newlines and indention:
+@example
+ffprobe -dump_separator "
+                          "  -i ~/videos/matrixbench_mpeg2.mpg
+@end example
+
 @end table
 
 @c man end CODEC OPTIONS

doc/decoders.texi

@@ -25,6 +25,13 @@ enabled decoders.
 A description of some of the currently available video decoders
 follows.
 
+@section hevc
+
+HEVC / H.265 decoder.
+
+Note: the @option{skip_loop_filter} option has effect only at level
+@code{all}.
+
 @section rawvideo
 
 Raw video decoder.
@@ -83,6 +90,23 @@ Loud sounds are fully compressed.  Soft sounds are enhanced.
 
 @end table
 
+@section flac
+
+FLAC audio decoder.
+
+This decoder aims to implement the complete FLAC specification from Xiph.
+
+@subsection FLAC Decoder options
+
+@table @option
+
+@item -use_buggy_lpc
+The lavc FLAC encoder used to produce buggy streams with high lpc values
+(like the default value). This option makes it possible to decode such streams
+correctly by using lavc's old buggy lpc logic for decoding.
+
+@end table
+
 @section ffwavesynth
 
 Internal wave synthetizer.
@@ -163,11 +187,33 @@ Requires the presence of the libopus headers and library during
 configuration. You need to explicitly configure the build with
 @code{--enable-libopus}.
 
+An FFmpeg native decoder for Opus exists, so users can decode Opus
+without this library.
+
 @c man end AUDIO DECODERS
 
 @chapter Subtitles Decoders
 @c man begin SUBTILES DECODERS
 
+@section dvbsub
+
+@subsection Options
+
+@table @option
+@item compute_clut
+@table @option
+@item -1
+Compute clut if no matching CLUT is in the stream.
+@item 0
+Never compute CLUT
+@item 1
+Always compute CLUT and override the one provided in the stream.
+@end table
+@item dvb_substream
+Selects the dvb substream, or all substreams if -1 which is default.
+
+@end table
+
 @section dvdsub
 
 This codec decodes the bitmap subtitles used in DVDs; the same subtitles can
@@ -187,6 +233,15 @@ The format for this option is a string containing 16 24-bits hexadecimal
 numbers (without 0x prefix) separated by comas, for example @code{0d00ee,
 ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b}.
+
+@item ifo_palette
+Specify the IFO file from which the global palette is obtained.
+(experimental)
+
+@item forced_subs_only
+Only decode subtitle entries marked as forced. Some titles have forced
+and non-forced subtitles in the same track. Setting this flag to @code{1}
+will only keep the forced subtitles. Default value is @code{0}.
 @end table
 
 @section libzvbi-teletext

doc/demuxers.texi

@@ -18,6 +18,12 @@ enabled demuxers.
 
 The description of some of the currently available demuxers follows.
 
+@section aa
+
+Audible Format 2, 3, and 4 demuxer.
+
+This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
+
 @section applehttp
 
 Apple HTTP Live Streaming demuxer.
@@ -29,6 +35,26 @@ the caller can decide which variant streams to actually receive.
 The total bitrate of the variant that the stream belongs to is
 available in a metadata key named "variant_bitrate".
 
+@section apng
+
+Animated Portable Network Graphics demuxer.
+
+This demuxer is used to demux APNG files.
+All headers, but the PNG signature, up to (but not including) the first
+fcTL chunk are transmitted as extradata.
+Frames are then split as being all the chunks between two fcTL ones, or
+between the last fcTL and IEND chunks.
+
+@table @option
+@item -ignore_loop @var{bool}
+Ignore the loop variable in the file if set.
+@item -max_fps @var{int}
+Maximum framerate in frames per second (0 for no limit).
+@item -default_fps @var{int}
+Default framerate in frames per second when none is specified in the file
+(0 meaning as fast as possible).
+@end table
+
 @section asf
 
 Advanced Systems Format demuxer.
@@ -78,7 +104,7 @@ All subsequent file-related directives apply to that file.
 
 @item @code{ffconcat version 1.0}
 Identify the script type and version. It also sets the @option{safe} option
-to 1 if it was to its default -1.
+to 1 if it was -1.
 
 To make FFmpeg recognize the format automatically, this directive must
 appears exactly as is (no extra space or byte-order-mark) on the very first
@@ -92,6 +118,47 @@ file is not available or accurate.
 If the duration is set for all files, then it is possible to seek in the
 whole concatenated video.
 
+@item @code{inpoint @var{timestamp}}
+In point of the file. When the demuxer opens the file it instantly seeks to the
+specified timestamp. Seeking is done so that all streams can be presented
+successfully at In point.
+
+This directive works best with intra frame codecs, because for non-intra frame
+ones you will usually get extra packets before the actual In point and the
+decoded content will most likely contain frames before In point too.
+
+For each file, packets before the file In point will have timestamps less than
+the calculated start timestamp of the file (negative in case of the first
+file), and the duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified In point.
+
+Because of potential packets before the specified In point, packet timestamps
+may overlap between two concatenated files.
+
+@item @code{outpoint @var{timestamp}}
+Out point of the file. When the demuxer reaches the specified decoding
+timestamp in any of the streams, it handles it as an end of file condition and
+skips the current and all the remaining packets from all streams.
+
+Out point is exclusive, which means that the demuxer will not output packets
+with a decoding timestamp greater or equal to Out point.
+
+This directive works best with intra frame codecs and formats where all streams
+are tightly interleaved. For non-intra frame codecs you will usually get
+additional packets with presentation timestamp after Out point therefore the
+decoded content will most likely contain frames after Out point too. If your
+streams are not tightly interleaved you may not get all the packets from all
+streams before Out point and you may only will be able to decode the earliest
+stream until Out point.
+
+The duration of the files (if not specified by the @code{duration}
+directive) will be reduced based on their specified Out point.
+
+@item @code{file_packet_metadata @var{key=value}}
+Metadata of the packets of the file. The specified metadata will be set for
+each file packet. You can specify this directive multiple times to add multiple
+metadata entries.
+
 @item @code{stream}
 Introduce a stream in the virtual file.
 All subsequent stream-related directives apply to the last introduced
@@ -125,12 +192,15 @@ component.
 
 If set to 0, any file name is accepted.
 
-The default is -1, it is equivalent to 1 if the format was automatically
+The default is 1.
+
+-1 is equivalent to 1 if the format was automatically
 probed and 0 otherwise.
 
 @item auto_convert
 If set to 1, try to perform automatic conversions on packet data to make the
 streams concatenable.
+The default is 1.
 
 Currently, the only conversion is adding the h264_mp4toannexb bitstream
 filter to H.264 streams in MP4 format. This is necessary in particular if
@@ -174,6 +244,63 @@ See @url{http://quvi.sourceforge.net/} for more information.
 FFmpeg needs to be built with @code{--enable-libquvi} for this demuxer to be
 enabled.
 
+@section gif
+
+Animated GIF demuxer.
+
+It accepts the following options:
+
+@table @option
+@item min_delay
+Set the minimum valid delay between frames in hundredths of seconds.
+Range is 0 to 6000. Default value is 2.
+
+@item max_gif_delay
+Set the maximum valid delay between frames in hundredth of seconds.
+Range is 0 to 65535. Default value is 65535 (nearly eleven minutes),
+the maximum value allowed by the specification.
+
+@item default_delay
+Set the default delay between frames in hundredths of seconds.
+Range is 0 to 6000. Default value is 10.
+
+@item ignore_loop
+GIF files can contain information to loop a certain number of times (or
+infinitely). If @option{ignore_loop} is set to 1, then the loop setting
+from the input will be ignored and looping will not occur. If set to 0,
+then looping will occur and will cycle the number of times according to
+the GIF. Default value is 1.
+@end table
+
+For example, with the overlay filter, place an infinitely looping GIF
+over another video:
+@example
+ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
+@end example
+
+Note that in the above example the shortest option for overlay filter is
+used to end the output video at the length of the shortest input file,
+which in this case is @file{input.mp4} as the GIF in this example loops
+infinitely.
+
+@section hls
+
+HLS demuxer
+
+It accepts the following options:
+
+@table @option
+@item live_start_index
+segment index to start live streams at (negative values are from the end).
+
+@item allowed_extensions
+',' separated list of file extensions that hls is allowed to access.
+
+@item max_reload
+Maximum number of times a insufficient list is attempted to be reloaded.
+Default value is 1000.
+@end table
+
 @section image2
 
 Image file demuxer.
@@ -199,6 +326,10 @@ Select the pattern type used to interpret the provided filename.
 
 @var{pattern_type} accepts one of the following values.
 @table @option
+@item none
+Disable pattern matching, therefore the video will only contain the specified
+image. You should use this option if you do not want to create sequences from
+multiple images and your filenames may contain special pattern characters.
 @item sequence
 Select a sequence pattern type, used to specify a sequence of files
 indexed by sequential numbers.
@@ -305,17 +436,47 @@ ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
 @end example
 @end itemize
 
+@section mov/mp4/3gp/Quicktme
+
+Quicktime / MP4 demuxer.
+
+This demuxer accepts the following options:
+@table @option
+@item enable_drefs
+Enable loading of external tracks, disabled by default.
+Enabling this can theoretically leak information in some use cases.
+
+@item use_absolute_path
+Allows loading of external tracks via absolute paths, disabled by default.
+Enabling this poses a security risk. It should only be enabled if the source
+is known to be non malicious.
+
+@end table
+
 @section mpegts
 
 MPEG-2 transport stream demuxer.
 
+This demuxer accepts the following options:
 @table @option
+@item resync_size
+Set size limit for looking up a new synchronization. Default value is
+65536.
 
 @item fix_teletext_pts
-Overrides teletext packet PTS and DTS values with the timestamps calculated
+Override teletext packet PTS and DTS values with the timestamps calculated
 from the PCR of the first program which the teletext stream is part of and is
 not discarded. Default value is 1, set this option to 0 if you want your
 teletext packet PTS and DTS values untouched.
+
+@item ts_packetsize
+Output option carrying the raw packet size in bytes.
+Show the detected raw packet size, cannot be set by the user.
+
+@item scan_all_pmts
+Scan and combine all PMTs. The value is an integer with value from -1
+to 1 (-1 means automatic setting, 1 means enabled, 0 means
+disabled). Default value is -1.
 @end table
 
 @section rawvideo

doc/developer.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Developer Documentation
 @titlepage
@@ -227,7 +228,7 @@ autocmd InsertEnter * match ForbiddenWhitespace /\t\|\s\+\%#\@@<!$/
 @end example
 
 For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
-@example
+@lisp
 (c-add-style "ffmpeg"
              '("k&r"
                (c-basic-offset . 4)
@@ -238,7 +239,7 @@ For Emacs, add these roughly equivalent lines to your @file{.emacs.d/init.el}:
                )
              )
 (setq c-default-style "ffmpeg")
-@end example
+@end lisp
 
 @section Development Policy
 
@@ -323,9 +324,12 @@ Always fill out the commit log message. Describe in a few lines what you
 changed and why. You can refer to mailing list postings if you fix a
 particular bug. Comments such as "fixed!" or "Changed it." are unacceptable.
 Recommended format:
+
+@example
 area changed: Short 1 line description
 
 details describing what and why and giving references.
+@end example
 
 @item
 Make sure the author of the commit is set correctly. (see git commit --author)
@@ -396,6 +400,35 @@ finding a new maintainer and also don't forget updating the @file{MAINTAINERS} f
 
 We think our rules are not too hard. If you have comments, contact us.
 
+@section Code of conduct
+
+Be friendly and respectful towards others and third parties.
+Treat others the way you yourself want to be treated.
+
+Be considerate. Not everyone shares the same viewpoint and priorities as you do.
+Different opinions and interpretations help the project.
+Looking at issues from a different perspective assists development.
+
+Do not assume malice for things that can be attributed to incompetence. Even if
+it is malice, it's rarely good to start with that as initial assumption.
+
+Stay friendly even if someone acts contrarily. Everyone has a bad day
+once in a while.
+If you yourself have a bad day or are angry then try to take a break and reply
+once you are calm and without anger if you have to.
+
+Try to help other team members and cooperate if you can.
+
+The goal of software development is to create technical excellence, not for any
+individual to be better and "win" against the others. Large software projects
+are only possible and successful through teamwork.
+
+If someone struggles do not put them down. Give them a helping hand
+instead and point them in the right direction.
+
+Finally, keep in mind the immortal words of Bill and Ted,
+"Be excellent to each other."
+
 @anchor{Submitting patches}
 @section Submitting patches
 
@@ -539,6 +572,10 @@ tools/trasher, the noise bitstream filter, and
 should not crash, end in a (near) infinite loop, or allocate ridiculous
 amounts of memory when fed damaged data.
 
+@item
+Did you test your decoder or demuxer against sample files?
+Samples may be obtained at @url{http://samples.ffmpeg.org}.
+
 @item
 Does the patch not mix functional and cosmetic changes?
 
@@ -629,6 +666,10 @@ not related to the comments received during review. Such patches will
 be rejected. Instead, submit significant changes or new features as
 separate patches.
 
+Everyone is welcome to review patches. Also if you are waiting for your patch
+to be reviewed, please consider helping to review other patches, that is a great
+way to get everyone's patches reviewed sooner.
+
 @anchor{Regression tests}
 @section Regression tests
 
@@ -644,12 +685,12 @@ accordingly].
 @subsection Adding files to the fate-suite dataset
 
 When there is no muxer or encoder available to generate test media for a
-specific test then the media has to be inlcuded in the fate-suite.
+specific test then the media has to be included in the fate-suite.
 First please make sure that the sample file is as small as possible to test the
 respective decoder or demuxer sufficiently. Large files increase network
 bandwidth and disk space requirements.
 Once you have a working fate test and fate sample, provide in the commit
-message or introductionary message for the patch series that you post to
+message or introductory message for the patch series that you post to
 the ffmpeg-devel mailing list, a direct link to download the sample media.
 
 

doc/doxy-wrapper.sh

@@ -6,8 +6,16 @@ DOXYGEN="${3}"
 
 shift 3
 
+if [ -e "$SRC_PATH/VERSION" ]; then
+    VERSION=`cat "$SRC_PATH/VERSION"`
+else
+    VERSION=`cd "$SRC_PATH"; git describe`
+fi
+
 $DOXYGEN - <<EOF
 @INCLUDE        = ${DOXYFILE}
 INPUT           = $@
 EXAMPLE_PATH    = ${SRC_PATH}/doc/examples
+HTML_TIMESTAMP  = NO
+PROJECT_NUMBER  = $VERSION
 EOF

doc/encoders.texi

@@ -494,6 +494,85 @@ Selected by Encoder (default)
 
 @end table
 
+@anchor{flac}
+@section flac
+
+FLAC (Free Lossless Audio Codec) Encoder
+
+@subsection Options
+
+The following options are supported by FFmpeg's flac encoder.
+
+@table @option
+@item compression_level
+Sets the compression level, which chooses defaults for many other options
+if they are not set explicitly.
+
+@item frame_size
+Sets the size of the frames in samples per channel.
+
+@item lpc_coeff_precision
+Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the
+default.
+
+@item lpc_type
+Sets the first stage LPC algorithm
+@table @samp
+@item none
+LPC is not used
+
+@item fixed
+fixed LPC coefficients
+
+@item levinson
+
+@item cholesky
+@end table
+
+@item lpc_passes
+Number of passes to use for Cholesky factorization during LPC analysis
+
+@item min_partition_order
+The minimum partition order
+
+@item max_partition_order
+The maximum partition order
+
+@item prediction_order_method
+@table @samp
+@item estimation
+@item 2level
+@item 4level
+@item 8level
+@item search
+Bruteforce search
+@item log
+@end table
+
+@item ch_mode
+Channel mode
+@table @samp
+@item auto
+The mode is chosen automatically for each frame
+@item indep
+Chanels are independently coded
+@item left_side
+@item right_side
+@item mid_side
+@end table
+
+@item exact_rice_parameters
+Chooses if rice parameters are calculated exactly or approximately.
+if set to 1 then they are chosen exactly, which slows the code down slightly and
+improves compression slightly.
+
+@item multi_dim_quant
+Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is
+applied after the first stage to finetune the coefficients. This is quite slow
+and slightly improves compression.
+
+@end table
+
 @anchor{libfaac}
 @section libfaac
 
@@ -1032,7 +1111,7 @@ configuration. You need to explicitly configure the build with
 
 @subsection Option Mapping
 
-Most libopus options are modeled after the @command{opusenc} utility from
+Most libopus options are modelled after the @command{opusenc} utility from
 opus-tools. The following is an option mapping chart describing options
 supported by the libopus wrapper, and their @command{opusenc}-equivalent
 in parentheses.
@@ -1263,6 +1342,30 @@ disabled
 A description of some of the currently available video encoders
 follows.
 
+@section jpeg2000
+
+The native jpeg 2000 encoder is lossy by default, the @code{-q:v}
+option can be used to set the encoding quality. Lossless encoding
+can be selected with @code{-pred 1}.
+
+@subsection Options
+
+@table @option
+@item format
+Can be set to either @code{j2k} or @code{jp2} (the default) that
+makes it possible to store non-rgb pix_fmts.
+
+@end table
+
+@section snow
+
+@subsection Options
+
+@table @option
+@item iterative_dia_size
+dia size for the iterative motion estimation
+@end table
+
 @section libtheora
 
 libtheora Theora encoder wrapper.
@@ -1330,117 +1433,160 @@ ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
 
 @section libvpx
 
-VP8 format supported through libvpx.
+VP8/VP9 format supported through libvpx.
 
 Requires the presence of the libvpx headers and library during configuration.
 You need to explicitly configure the build with @code{--enable-libvpx}.
 
 @subsection Options
 
-Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
+The following options are supported by the libvpx wrapper. The
+@command{vpxenc}-equivalent options or values are listed in parentheses
+for easy migration.
+
+To reduce the duplication of documentation, only the private options
+and some others requiring special attention are documented here. For
+the documentation of the undocumented generic options, see
+@ref{codec-options,,the Codec Options chapter}.
+
+To get more documentation of the libvpx options, invoke the command
+@command{ffmpeg -h encoder=libvpx}, @command{ffmpeg -h encoder=libvpx-vp9} or
+@command{vpxenc --help}. Further information is available in the libvpx API
+documentation.
 
 @table @option
 
-@item threads
-g_threads
+@item b (@emph{target-bitrate})
+Set bitrate in bits/s. Note that FFmpeg's @option{b} option is
+expressed in bits/s, while @command{vpxenc}'s @option{target-bitrate} is in
+kilobits/s.
 
-@item profile
-g_profile
+@item g (@emph{kf-max-dist})
 
-@item vb
-rc_target_bitrate
+@item keyint_min (@emph{kf-min-dist})
 
-@item g
-kf_max_dist
+@item qmin (@emph{min-q})
 
-@item keyint_min
-kf_min_dist
+@item qmax (@emph{max-q})
 
-@item qmin
-rc_min_quantizer
+@item bufsize (@emph{buf-sz}, @emph{buf-optimal-sz})
+Set ratecontrol buffer size (in bits). Note @command{vpxenc}'s options are
+specified in milliseconds, the libvpx wrapper converts this value as follows:
+@code{buf-sz = bufsize * 1000 / bitrate},
+@code{buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6}.
 
-@item qmax
-rc_max_quantizer
+@item rc_init_occupancy (@emph{buf-initial-sz})
+Set number of bits which should be loaded into the rc buffer before decoding
+starts. Note @command{vpxenc}'s option is specified in milliseconds, the libvpx
+wrapper converts this value as follows:
+@code{rc_init_occupancy * 1000 / bitrate}.
 
-@item bufsize, vb
-rc_buf_sz
-@code{(bufsize * 1000 / vb)}
+@item undershoot-pct
+Set datarate undershoot (min) percentage of the target bitrate.
 
-rc_buf_optimal_sz
-@code{(bufsize * 1000 / vb * 5 / 6)}
+@item overshoot-pct
+Set datarate overshoot (max) percentage of the target bitrate.
 
-@item rc_init_occupancy, vb
-rc_buf_initial_sz
-@code{(rc_init_occupancy * 1000 / vb)}
+@item skip_threshold (@emph{drop-frame})
 
-@item rc_buffer_aggressivity
-rc_undershoot_pct
+@item qcomp (@emph{bias-pct})
 
-@item skip_threshold
-rc_dropframe_thresh
+@item maxrate (@emph{maxsection-pct})
+Set GOP max bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+percentage of the target bitrate, the libvpx wrapper converts this value as
+follows: @code{(maxrate * 100 / bitrate)}.
 
-@item qcomp
-rc_2pass_vbr_bias_pct
+@item minrate (@emph{minsection-pct})
+Set GOP min bitrate in bits/s. Note @command{vpxenc}'s option is specified as a
+percentage of the target bitrate, the libvpx wrapper converts this value as
+follows: @code{(minrate * 100 / bitrate)}.
 
-@item maxrate, vb
-rc_2pass_vbr_maxsection_pct
-@code{(maxrate * 100 / vb)}
+@item minrate, maxrate, b @emph{end-usage=cbr}
+@code{(minrate == maxrate == bitrate)}.
 
-@item minrate, vb
-rc_2pass_vbr_minsection_pct
-@code{(minrate * 100 / vb)}
+@item crf (@emph{end-usage=cq}, @emph{cq-level})
 
-@item minrate, maxrate, vb
-@code{VPX_CBR}
-@code{(minrate == maxrate == vb)}
-
-@item crf
-@code{VPX_CQ}, @code{VP8E_SET_CQ_LEVEL}
-
-@item quality
-@table @option
-@item @var{best}
-@code{VPX_DL_BEST_QUALITY}
-@item @var{good}
-@code{VPX_DL_GOOD_QUALITY}
-@item @var{realtime}
-@code{VPX_DL_REALTIME}
+@item quality, deadline (@emph{deadline})
+@table @samp
+@item best
+Use best quality deadline. Poorly named and quite slow, this option should be
+avoided as it may give worse quality output than good.
+@item good
+Use good quality deadline. This is a good trade-off between speed and quality
+when used with the @option{cpu-used} option.
+@item realtime
+Use realtime quality deadline.
 @end table
 
-@item speed
-@code{VP8E_SET_CPUUSED}
+@item speed, cpu-used (@emph{cpu-used})
+Set quality/speed ratio modifier. Higher values speed up the encode at the cost
+of quality.
 
-@item nr
-@code{VP8E_SET_NOISE_SENSITIVITY}
+@item nr (@emph{noise-sensitivity})
 
-@item mb_threshold
-@code{VP8E_SET_STATIC_THRESHOLD}
+@item static-thresh
+Set a change threshold on blocks below which they will be skipped by the
+encoder.
 
-@item slices
-@code{VP8E_SET_TOKEN_PARTITIONS}
+@item slices (@emph{token-parts})
+Note that FFmpeg's @option{slices} option gives the total number of partitions,
+while @command{vpxenc}'s @option{token-parts} is given as
+@code{log2(partitions)}.
 
 @item max-intra-rate
-@code{VP8E_SET_MAX_INTRA_BITRATE_PCT}
+Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0
+means unlimited.
 
 @item force_key_frames
 @code{VPX_EFLAG_FORCE_KF}
 
 @item Alternate reference frame related
 @table @option
-@item vp8flags altref
-@code{VP8E_SET_ENABLEAUTOALTREF}
-@item @var{arnr_max_frames}
-@code{VP8E_SET_ARNR_MAXFRAMES}
-@item @var{arnr_type}
-@code{VP8E_SET_ARNR_TYPE}
-@item @var{arnr_strength}
-@code{VP8E_SET_ARNR_STRENGTH}
-@item @var{rc_lookahead}
-g_lag_in_frames
+@item auto-alt-ref
+Enable use of alternate reference frames (2-pass only).
+@item arnr-max-frames
+Set altref noise reduction max frame count.
+@item arnr-type
+Set altref noise reduction filter type: backward, forward, centered.
+@item arnr-strength
+Set altref noise reduction filter strength.
+@item rc-lookahead, lag-in-frames (@emph{lag-in-frames})
+Set number of frames to look ahead for frametype and ratecontrol.
 @end table
 
-@item vp8flags error_resilient
-g_error_resilient
+@item error-resilient
+Enable error resiliency features.
+
+@item VP9-specific options
+@table @option
+@item lossless
+Enable lossless mode.
+@item tile-columns
+Set number of tile columns to use. Note this is given as
+@code{log2(tile_columns)}. For example, 8 tile columns would be requested by
+setting the @option{tile-columns} option to 3.
+@item tile-rows
+Set number of tile rows to use. Note this is given as @code{log2(tile_rows)}.
+For example, 4 tile rows would be requested by setting the @option{tile-rows}
+option to 2.
+@item frame-parallel
+Enable frame parallel decodability features.
+@item aq-mode
+Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
+cyclic refresh).
+@item colorspace @emph{color-space}
+Set input color space. The VP9 bitstream supports signaling the following
+colorspaces:
+@table @option
+@item @samp{rgb} @emph{sRGB}
+@item @samp{bt709} @emph{bt709}
+@item @samp{unspecified} @emph{unknown}
+@item @samp{bt470bg} @emph{bt601}
+@item @samp{smpte170m} @emph{smpte170}
+@item @samp{smpte240m} @emph{smpte240}
+@item @samp{bt2020_ncl} @emph{bt2020}
+@end table
+@end table
 
 @end table
 
@@ -1742,6 +1888,10 @@ Enable calculation and printing SSIM stats after the encoding.
 Enable the use of Periodic Intra Refresh instead of IDR frames when set
 to 1.
 
+@item avcintra-class (@emph{class})
+Configure the encoder to generate AVC-Intra.
+Valid values are 50,100 and 200
+
 @item bluray-compat (@emph{bluray-compat})
 Configure the encoder to be compatible with the bluray standard.
 It is a shorthand for setting "bluray-compat=1 force-cfr=1".
@@ -1879,6 +2029,34 @@ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
 Encoding ffpresets for common usages are provided so they can be used with the
 general presets system (e.g. passing the @option{pre} option).
 
+@section libx265
+
+x265 H.265/HEVC encoder wrapper.
+
+This encoder requires the presence of the libx265 headers and library
+during configuration. You need to explicitly configure the build with
+@option{--enable-libx265}.
+
+@subsection Options
+
+@table @option
+@item preset
+Set the x265 preset.
+
+@item tune
+Set the x265 tune parameter.
+
+@item x265-params
+Set x265 options using a list of @var{key}=@var{value} couples separated
+by ":". See @command{x265 --help} for a list of options.
+
+For example to specify libx265 encoding options with @option{-x265-params}:
+
+@example
+ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
+@end example
+@end table
+
 @section libxvid
 
 Xvid MPEG-4 Part 2 encoder wrapper.
@@ -2136,7 +2314,7 @@ Use @var{0} to disable alpha plane coding.
 @subsection Speed considerations
 
 In the default mode of operation the encoder has to honor frame constraints
-(i.e. not produc frames with size bigger than requested) while still making
+(i.e. not produce frames with size bigger than requested) while still making
 output picture as good as possible.
 A frame containing a lot of small details is harder to compress and the encoder
 would spend more time searching for appropriate quantizers for each slice.
@@ -2146,6 +2324,30 @@ Setting a higher @option{bits_per_mb} limit will improve the speed.
 For the fastest encoding speed set the @option{qscale} parameter (4 is the
 recommended value) and do not set a size constraint.
 
+@section libkvazaar
+
+Kvazaar H.265/HEVC encoder.
+
+Requires the presence of the libkvazaar headers and library during
+configuration. You need to explicitly configure the build with
+@option{--enable-libkvazaar}.
+
+@subsection Options
+
+@table @option
+
+@item b
+Set target video bitrate in bit/s and enable rate control.
+
+@item threads
+Set number of encoding threads.
+
+@item kvazaar-params
+Set kvazaar parameters as a list of @var{name}=@var{value} pairs separated
+by commas (,). See kvazaar documentation for a list of options.
+
+@end table
+
 @c man end VIDEO ENCODERS
 
 @chapter Subtitles Encoders

doc/examples/Makefile

@@ -11,11 +11,14 @@ CFLAGS += -Wall -g
 CFLAGS := $(shell pkg-config --cflags $(FFMPEG_LIBS)) $(CFLAGS)
 LDLIBS := $(shell pkg-config --libs $(FFMPEG_LIBS)) $(LDLIBS)
 
-EXAMPLES=       avio_reading                       \
+EXAMPLES=       avio_dir_cmd                       \
+                avio_reading                       \
                 decoding_encoding                  \
                 demuxing_decoding                  \
+                extract_mvs                        \
                 filtering_video                    \
                 filtering_audio                    \
+                http_multiclient                   \
                 metadata                           \
                 muxing                             \
                 remuxing                           \
@@ -28,6 +31,7 @@ OBJS=$(addsuffix .o,$(EXAMPLES))
 
 # the following examples make explicit use of the math library
 avcodec:           LDLIBS += -lm
+decoding_encoding: LDLIBS += -lm
 muxing:            LDLIBS += -lm
 resampling_audio:  LDLIBS += -lm
 

doc/examples/avio_dir_cmd.c

@@ -0,0 +1,180 @@
+/*
+ * Copyright (c) 2014 Lukasz Marek
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavformat/avio.h>
+
+static const char *type_string(int type)
+{
+    switch (type) {
+    case AVIO_ENTRY_DIRECTORY:
+        return "<DIR>";
+    case AVIO_ENTRY_FILE:
+        return "<FILE>";
+    case AVIO_ENTRY_BLOCK_DEVICE:
+        return "<BLOCK DEVICE>";
+    case AVIO_ENTRY_CHARACTER_DEVICE:
+        return "<CHARACTER DEVICE>";
+    case AVIO_ENTRY_NAMED_PIPE:
+        return "<PIPE>";
+    case AVIO_ENTRY_SYMBOLIC_LINK:
+        return "<LINK>";
+    case AVIO_ENTRY_SOCKET:
+        return "<SOCKET>";
+    case AVIO_ENTRY_SERVER:
+        return "<SERVER>";
+    case AVIO_ENTRY_SHARE:
+        return "<SHARE>";
+    case AVIO_ENTRY_WORKGROUP:
+        return "<WORKGROUP>";
+    case AVIO_ENTRY_UNKNOWN:
+    default:
+        break;
+    }
+    return "<UNKNOWN>";
+}
+
+static int list_op(const char *input_dir)
+{
+    AVIODirEntry *entry = NULL;
+    AVIODirContext *ctx = NULL;
+    int cnt, ret;
+    char filemode[4], uid_and_gid[20];
+
+    if ((ret = avio_open_dir(&ctx, input_dir, NULL)) < 0) {
+        av_log(NULL, AV_LOG_ERROR, "Cannot open directory: %s.\n", av_err2str(ret));
+        goto fail;
+    }
+
+    cnt = 0;
+    for (;;) {
+        if ((ret = avio_read_dir(ctx, &entry)) < 0) {
+            av_log(NULL, AV_LOG_ERROR, "Cannot list directory: %s.\n", av_err2str(ret));
+            goto fail;
+        }
+        if (!entry)
+            break;
+        if (entry->filemode == -1) {
+            snprintf(filemode, 4, "???");
+        } else {
+            snprintf(filemode, 4, "%3"PRIo64, entry->filemode);
+        }
+        snprintf(uid_and_gid, 20, "%"PRId64"(%"PRId64")", entry->user_id, entry->group_id);
+        if (cnt == 0)
+            av_log(NULL, AV_LOG_INFO, "%-9s %12s %30s %10s %s %16s %16s %16s\n",
+                   "TYPE", "SIZE", "NAME", "UID(GID)", "UGO", "MODIFIED",
+                   "ACCESSED", "STATUS_CHANGED");
+        av_log(NULL, AV_LOG_INFO, "%-9s %12"PRId64" %30s %10s %s %16"PRId64" %16"PRId64" %16"PRId64"\n",
+               type_string(entry->type),
+               entry->size,
+               entry->name,
+               uid_and_gid,
+               filemode,
+               entry->modification_timestamp,
+               entry->access_timestamp,
+               entry->status_change_timestamp);
+        avio_free_directory_entry(&entry);
+        cnt++;
+    };
+
+  fail:
+    avio_close_dir(&ctx);
+    return ret;
+}
+
+static int del_op(const char *url)
+{
+    int ret = avpriv_io_delete(url);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot delete '%s': %s.\n", url, av_err2str(ret));
+    return ret;
+}
+
+static int move_op(const char *src, const char *dst)
+{
+    int ret = avpriv_io_move(src, dst);
+    if (ret < 0)
+        av_log(NULL, AV_LOG_ERROR, "Cannot move '%s' into '%s': %s.\n", src, dst, av_err2str(ret));
+    return ret;
+}
+
+
+static void usage(const char *program_name)
+{
+    fprintf(stderr, "usage: %s OPERATION entry1 [entry2]\n"
+            "API example program to show how to manipulate resources "
+            "accessed through AVIOContext.\n"
+            "OPERATIONS:\n"
+            "list      list content of the directory\n"
+            "move      rename content in directory\n"
+            "del       delete content in directory\n",
+            program_name);
+}
+
+int main(int argc, char *argv[])
+{
+    const char *op = NULL;
+    int ret;
+
+    av_log_set_level(AV_LOG_DEBUG);
+
+    if (argc < 2) {
+        usage(argv[0]);
+        return 1;
+    }
+
+    /* register codecs and formats and other lavf/lavc components*/
+    av_register_all();
+    avformat_network_init();
+
+    op = argv[1];
+    if (strcmp(op, "list") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for list operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = list_op(argv[2]);
+        }
+    } else if (strcmp(op, "del") == 0) {
+        if (argc < 3) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for del operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = del_op(argv[2]);
+        }
+    } else if (strcmp(op, "move") == 0) {
+        if (argc < 4) {
+            av_log(NULL, AV_LOG_INFO, "Missing argument for move operation.\n");
+            ret = AVERROR(EINVAL);
+        } else {
+            ret = move_op(argv[2], argv[3]);
+        }
+    } else {
+        av_log(NULL, AV_LOG_INFO, "Invalid operation %s\n", op);
+        ret = AVERROR(EINVAL);
+    }
+
+    avformat_network_deinit();
+
+    return ret < 0 ? 1 : 0;
+}

doc/examples/decoding_encoding.c

@@ -245,7 +245,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     AVCodecContext *c= NULL;
     int len;
     FILE *f, *outfile;
-    uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[AUDIO_INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
     AVFrame *decoded_frame = NULL;
 
@@ -288,6 +288,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
     avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
 
     while (avpkt.size > 0) {
+        int i, ch;
         int got_frame = 0;
 
         if (!decoded_frame) {
@@ -304,15 +305,15 @@ static void audio_decode_example(const char *outfilename, const char *filename)
         }
         if (got_frame) {
             /* if a frame has been decoded, output it */
-            int data_size = av_samples_get_buffer_size(NULL, c->channels,
-                                                       decoded_frame->nb_samples,
-                                                       c->sample_fmt, 1);
+            int data_size = av_get_bytes_per_sample(c->sample_fmt);
             if (data_size < 0) {
                 /* This should not occur, checking just for paranoia */
                 fprintf(stderr, "Failed to calculate data size\n");
                 exit(1);
             }
-            fwrite(decoded_frame->data[0], 1, data_size, outfile);
+            for (i=0; i<decoded_frame->nb_samples; i++)
+                for (ch=0; ch<c->channels; ch++)
+                    fwrite(decoded_frame->data[ch] + data_size*i, 1, data_size, outfile);
         }
         avpkt.size -= len;
         avpkt.data += len;
@@ -520,7 +521,7 @@ static int decode_write_frame(const char *outfilename, AVCodecContext *avctx,
         /* the picture is allocated by the decoder, no need to free it */
         snprintf(buf, sizeof(buf), outfilename, *frame_count);
         pgm_save(frame->data[0], frame->linesize[0],
-                 avctx->width, avctx->height, buf);
+                 frame->width, frame->height, buf);
         (*frame_count)++;
     }
     if (pkt->data) {
@@ -537,13 +538,13 @@ static void video_decode_example(const char *outfilename, const char *filename)
     int frame_count;
     FILE *f;
     AVFrame *frame;
-    uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+    uint8_t inbuf[INBUF_SIZE + AV_INPUT_BUFFER_PADDING_SIZE];
     AVPacket avpkt;
 
     av_init_packet(&avpkt);
 
     /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
-    memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+    memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
 
     printf("Decode video file %s to %s\n", filename, outfilename);
 
@@ -560,8 +561,8 @@ static void video_decode_example(const char *outfilename, const char *filename)
         exit(1);
     }
 
-    if(codec->capabilities&CODEC_CAP_TRUNCATED)
-        c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
+    if (codec->capabilities & AV_CODEC_CAP_TRUNCATED)
+        c->flags |= AV_CODEC_FLAG_TRUNCATED; // we do not send complete frames
 
     /* For some codecs, such as msmpeg4 and mpeg4, width and height
        MUST be initialized there because this information is not
@@ -650,7 +651,7 @@ int main(int argc, char **argv)
         video_encode_example("test.h264", AV_CODEC_ID_H264);
     } else if (!strcmp(output_type, "mp2")) {
         audio_encode_example("test.mp2");
-        audio_decode_example("test.sw", "test.mp2");
+        audio_decode_example("test.pcm", "test.mp2");
     } else if (!strcmp(output_type, "mpg")) {
         video_encode_example("test.mpg", AV_CODEC_ID_MPEG1VIDEO);
         video_decode_example("test%02d.pgm", "test.mpg");

doc/examples/demuxing_decoding.c

@@ -36,6 +36,8 @@
 
 static AVFormatContext *fmt_ctx = NULL;
 static AVCodecContext *video_dec_ctx = NULL, *audio_dec_ctx;
+static int width, height;
+static enum AVPixelFormat pix_fmt;
 static AVStream *video_stream = NULL, *audio_stream = NULL;
 static const char *src_filename = NULL;
 static const char *video_dst_filename = NULL;
@@ -81,6 +83,22 @@ static int decode_packet(int *got_frame, int cached)
         }
 
         if (*got_frame) {
+
+            if (frame->width != width || frame->height != height ||
+                frame->format != pix_fmt) {
+                /* To handle this change, one could call av_image_alloc again and
+                 * decode the following frames into another rawvideo file. */
+                fprintf(stderr, "Error: Width, height and pixel format have to be "
+                        "constant in a rawvideo file, but the width, height or "
+                        "pixel format of the input video changed:\n"
+                        "old: width = %d, height = %d, format = %s\n"
+                        "new: width = %d, height = %d, format = %s\n",
+                        width, height, av_get_pix_fmt_name(pix_fmt),
+                        frame->width, frame->height,
+                        av_get_pix_fmt_name(frame->format));
+                return -1;
+            }
+
             printf("video_frame%s n:%d coded_n:%d pts:%s\n",
                    cached ? "(cached)" : "",
                    video_frame_count++, frame->coded_picture_number,
@@ -90,7 +108,7 @@ static int decode_packet(int *got_frame, int cached)
              * this is required since rawvideo expects non aligned data */
             av_image_copy(video_dst_data, video_dst_linesize,
                           (const uint8_t **)(frame->data), frame->linesize,
-                          video_dec_ctx->pix_fmt, video_dec_ctx->width, video_dec_ctx->height);
+                          pix_fmt, width, height);
 
             /* write to rawvideo file */
             fwrite(video_dst_data[0], 1, video_dst_bufsize, video_dst_file);
@@ -138,7 +156,7 @@ static int decode_packet(int *got_frame, int cached)
 static int open_codec_context(int *stream_idx,
                               AVFormatContext *fmt_ctx, enum AVMediaType type)
 {
-    int ret;
+    int ret, stream_index;
     AVStream *st;
     AVCodecContext *dec_ctx = NULL;
     AVCodec *dec = NULL;
@@ -150,8 +168,8 @@ static int open_codec_context(int *stream_idx,
                 av_get_media_type_string(type), src_filename);
         return ret;
     } else {
-        *stream_idx = ret;
-        st = fmt_ctx->streams[*stream_idx];
+        stream_index = ret;
+        st = fmt_ctx->streams[stream_index];
 
         /* find decoder for the stream */
         dec_ctx = st->codec;
@@ -170,6 +188,7 @@ static int open_codec_context(int *stream_idx,
                     av_get_media_type_string(type));
             return ret;
         }
+        *stream_idx = stream_index;
     }
 
     return 0;
@@ -264,9 +283,11 @@ int main (int argc, char **argv)
         }
 
         /* allocate image where the decoded image will be put */
+        width = video_dec_ctx->width;
+        height = video_dec_ctx->height;
+        pix_fmt = video_dec_ctx->pix_fmt;
         ret = av_image_alloc(video_dst_data, video_dst_linesize,
-                             video_dec_ctx->width, video_dec_ctx->height,
-                             video_dec_ctx->pix_fmt, 1);
+                             width, height, pix_fmt, 1);
         if (ret < 0) {
             fprintf(stderr, "Could not allocate raw video buffer\n");
             goto end;
@@ -341,7 +362,7 @@ int main (int argc, char **argv)
     if (video_stream) {
         printf("Play the output video file with the command:\n"
                "ffplay -f rawvideo -pix_fmt %s -video_size %dx%d %s\n",
-               av_get_pix_fmt_name(video_dec_ctx->pix_fmt), video_dec_ctx->width, video_dec_ctx->height,
+               av_get_pix_fmt_name(pix_fmt), width, height,
                video_dst_filename);
     }
 

doc/examples/extract_mvs.c

@@ -0,0 +1,185 @@
+/*
+ * Copyright (c) 2012 Stefano Sabatini
+ * Copyright (c) 2014 Clément Bœsch
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+#include <libavutil/motion_vector.h>
+#include <libavformat/avformat.h>
+
+static AVFormatContext *fmt_ctx = NULL;
+static AVCodecContext *video_dec_ctx = NULL;
+static AVStream *video_stream = NULL;
+static const char *src_filename = NULL;
+
+static int video_stream_idx = -1;
+static AVFrame *frame = NULL;
+static AVPacket pkt;
+static int video_frame_count = 0;
+
+static int decode_packet(int *got_frame, int cached)
+{
+    int decoded = pkt.size;
+
+    *got_frame = 0;
+
+    if (pkt.stream_index == video_stream_idx) {
+        int ret = avcodec_decode_video2(video_dec_ctx, frame, got_frame, &pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error decoding video frame (%s)\n", av_err2str(ret));
+            return ret;
+        }
+
+        if (*got_frame) {
+            int i;
+            AVFrameSideData *sd;
+
+            video_frame_count++;
+            sd = av_frame_get_side_data(frame, AV_FRAME_DATA_MOTION_VECTORS);
+            if (sd) {
+                const AVMotionVector *mvs = (const AVMotionVector *)sd->data;
+                for (i = 0; i < sd->size / sizeof(*mvs); i++) {
+                    const AVMotionVector *mv = &mvs[i];
+                    printf("%d,%2d,%2d,%2d,%4d,%4d,%4d,%4d,0x%"PRIx64"\n",
+                           video_frame_count, mv->source,
+                           mv->w, mv->h, mv->src_x, mv->src_y,
+                           mv->dst_x, mv->dst_y, mv->flags);
+                }
+            }
+        }
+    }
+
+    return decoded;
+}
+
+static int open_codec_context(int *stream_idx,
+                              AVFormatContext *fmt_ctx, enum AVMediaType type)
+{
+    int ret;
+    AVStream *st;
+    AVCodecContext *dec_ctx = NULL;
+    AVCodec *dec = NULL;
+    AVDictionary *opts = NULL;
+
+    ret = av_find_best_stream(fmt_ctx, type, -1, -1, NULL, 0);
+    if (ret < 0) {
+        fprintf(stderr, "Could not find %s stream in input file '%s'\n",
+                av_get_media_type_string(type), src_filename);
+        return ret;
+    } else {
+        *stream_idx = ret;
+        st = fmt_ctx->streams[*stream_idx];
+
+        /* find decoder for the stream */
+        dec_ctx = st->codec;
+        dec = avcodec_find_decoder(dec_ctx->codec_id);
+        if (!dec) {
+            fprintf(stderr, "Failed to find %s codec\n",
+                    av_get_media_type_string(type));
+            return AVERROR(EINVAL);
+        }
+
+        /* Init the video decoder */
+        av_dict_set(&opts, "flags2", "+export_mvs", 0);
+        if ((ret = avcodec_open2(dec_ctx, dec, &opts)) < 0) {
+            fprintf(stderr, "Failed to open %s codec\n",
+                    av_get_media_type_string(type));
+            return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    int ret = 0, got_frame;
+
+    if (argc != 2) {
+        fprintf(stderr, "Usage: %s <video>\n", argv[0]);
+        exit(1);
+    }
+    src_filename = argv[1];
+
+    av_register_all();
+
+    if (avformat_open_input(&fmt_ctx, src_filename, NULL, NULL) < 0) {
+        fprintf(stderr, "Could not open source file %s\n", src_filename);
+        exit(1);
+    }
+
+    if (avformat_find_stream_info(fmt_ctx, NULL) < 0) {
+        fprintf(stderr, "Could not find stream information\n");
+        exit(1);
+    }
+
+    if (open_codec_context(&video_stream_idx, fmt_ctx, AVMEDIA_TYPE_VIDEO) >= 0) {
+        video_stream = fmt_ctx->streams[video_stream_idx];
+        video_dec_ctx = video_stream->codec;
+    }
+
+    av_dump_format(fmt_ctx, 0, src_filename, 0);
+
+    if (!video_stream) {
+        fprintf(stderr, "Could not find video stream in the input, aborting\n");
+        ret = 1;
+        goto end;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        fprintf(stderr, "Could not allocate frame\n");
+        ret = AVERROR(ENOMEM);
+        goto end;
+    }
+
+    printf("framenum,source,blockw,blockh,srcx,srcy,dstx,dsty,flags\n");
+
+    /* initialize packet, set data to NULL, let the demuxer fill it */
+    av_init_packet(&pkt);
+    pkt.data = NULL;
+    pkt.size = 0;
+
+    /* read frames from the file */
+    while (av_read_frame(fmt_ctx, &pkt) >= 0) {
+        AVPacket orig_pkt = pkt;
+        do {
+            ret = decode_packet(&got_frame, 0);
+            if (ret < 0)
+                break;
+            pkt.data += ret;
+            pkt.size -= ret;
+        } while (pkt.size > 0);
+        av_free_packet(&orig_pkt);
+    }
+
+    /* flush cached frames */
+    pkt.data = NULL;
+    pkt.size = 0;
+    do {
+        decode_packet(&got_frame, 1);
+    } while (got_frame);
+
+end:
+    avcodec_close(video_dec_ctx);
+    avformat_close_input(&fmt_ctx);
+    av_frame_free(&frame);
+    return ret < 0;
+}

doc/examples/filter_audio.c

@@ -45,6 +45,7 @@
 
 #include "libavutil/channel_layout.h"
 #include "libavutil/md5.h"
+#include "libavutil/mem.h"
 #include "libavutil/opt.h"
 #include "libavutil/samplefmt.h"
 

doc/examples/filtering_audio.c

@@ -145,12 +145,28 @@ static int init_filters(const char *filters_descr)
         goto end;
     }
 
-    /* Endpoints for the filter graph. */
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
     outputs->name       = av_strdup("in");
     outputs->filter_ctx = buffersrc_ctx;
     outputs->pad_idx    = 0;
     outputs->next       = NULL;
 
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
     inputs->name       = av_strdup("out");
     inputs->filter_ctx = buffersink_ctx;
     inputs->pad_idx    = 0;

doc/examples/filtering_video.c

@@ -38,7 +38,10 @@
 #include <libavfilter/buffersrc.h>
 #include <libavutil/opt.h>
 
-const char *filter_descr = "scale=78:24";
+const char *filter_descr = "scale=78:24,transpose=cclock";
+/* other way:
+   scale=78:24 [scl]; [scl] transpose=cclock // assumes "[in]" and "[out]" to be input output pads respectively
+ */
 
 static AVFormatContext *fmt_ctx;
 static AVCodecContext *dec_ctx;
@@ -90,6 +93,7 @@ static int init_filters(const char *filters_descr)
     AVFilter *buffersink = avfilter_get_by_name("buffersink");
     AVFilterInOut *outputs = avfilter_inout_alloc();
     AVFilterInOut *inputs  = avfilter_inout_alloc();
+    AVRational time_base = fmt_ctx->streams[video_stream_index]->time_base;
     enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_GRAY8, AV_PIX_FMT_NONE };
 
     filter_graph = avfilter_graph_alloc();
@@ -102,7 +106,7 @@ static int init_filters(const char *filters_descr)
     snprintf(args, sizeof(args),
             "video_size=%dx%d:pix_fmt=%d:time_base=%d/%d:pixel_aspect=%d/%d",
             dec_ctx->width, dec_ctx->height, dec_ctx->pix_fmt,
-            dec_ctx->time_base.num, dec_ctx->time_base.den,
+            time_base.num, time_base.den,
             dec_ctx->sample_aspect_ratio.num, dec_ctx->sample_aspect_ratio.den);
 
     ret = avfilter_graph_create_filter(&buffersrc_ctx, buffersrc, "in",
@@ -127,12 +131,28 @@ static int init_filters(const char *filters_descr)
         goto end;
     }
 
-    /* Endpoints for the filter graph. */
+    /*
+     * Set the endpoints for the filter graph. The filter_graph will
+     * be linked to the graph described by filters_descr.
+     */
+
+    /*
+     * The buffer source output must be connected to the input pad of
+     * the first filter described by filters_descr; since the first
+     * filter input label is not specified, it is set to "in" by
+     * default.
+     */
     outputs->name       = av_strdup("in");
     outputs->filter_ctx = buffersrc_ctx;
     outputs->pad_idx    = 0;
     outputs->next       = NULL;
 
+    /*
+     * The buffer sink input must be connected to the output pad of
+     * the last filter described by filters_descr; since the last
+     * filter output label is not specified, it is set to "out" by
+     * default.
+     */
     inputs->name       = av_strdup("out");
     inputs->filter_ctx = buffersink_ctx;
     inputs->pad_idx    = 0;

doc/examples/http_multiclient.c

@@ -0,0 +1,155 @@
+/*
+ * Copyright (c) 2015 Stephan Holljes
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * libavformat multi-client network API usage example.
+ *
+ * @example http_multiclient.c
+ * This example will serve a file without decoding or demuxing it over http.
+ * Multiple clients can connect and will receive the same file.
+ */
+
+#include <libavformat/avformat.h>
+#include <libavutil/opt.h>
+#include <unistd.h>
+
+void process_client(AVIOContext *client, const char *in_uri)
+{
+    AVIOContext *input = NULL;
+    uint8_t buf[1024];
+    int ret, n, reply_code;
+    char *resource = NULL;
+    while ((ret = avio_handshake(client)) > 0) {
+        av_opt_get(client, "resource", AV_OPT_SEARCH_CHILDREN, &resource);
+        // check for strlen(resource) is necessary, because av_opt_get()
+        // may return empty string.
+        if (resource && strlen(resource))
+            break;
+    }
+    if (ret < 0)
+        goto end;
+    av_log(client, AV_LOG_TRACE, "resource=%p\n", resource);
+    if (resource && resource[0] == '/' && !strcmp((resource + 1), in_uri)) {
+        reply_code = 200;
+    } else {
+        reply_code = AVERROR_HTTP_NOT_FOUND;
+    }
+    if ((ret = av_opt_set_int(client, "reply_code", reply_code, AV_OPT_SEARCH_CHILDREN)) < 0) {
+        av_log(client, AV_LOG_ERROR, "Failed to set reply_code: %s.\n", av_err2str(ret));
+        goto end;
+    }
+    av_log(client, AV_LOG_TRACE, "Set reply code to %d\n", reply_code);
+
+    while ((ret = avio_handshake(client)) > 0);
+
+    if (ret < 0)
+        goto end;
+
+    fprintf(stderr, "Handshake performed.\n");
+    if (reply_code != 200)
+        goto end;
+    fprintf(stderr, "Opening input file.\n");
+    if ((ret = avio_open2(&input, in_uri, AVIO_FLAG_READ, NULL, NULL)) < 0) {
+        av_log(input, AV_LOG_ERROR, "Failed to open input: %s: %s.\n", in_uri,
+               av_err2str(ret));
+        goto end;
+    }
+    for(;;) {
+        n = avio_read(input, buf, sizeof(buf));
+        if (n < 0) {
+            if (n == AVERROR_EOF)
+                break;
+            av_log(input, AV_LOG_ERROR, "Error reading from input: %s.\n",
+                   av_err2str(n));
+            break;
+        }
+        avio_write(client, buf, n);
+        avio_flush(client);
+    }
+end:
+    fprintf(stderr, "Flushing client\n");
+    avio_flush(client);
+    fprintf(stderr, "Closing client\n");
+    avio_close(client);
+    fprintf(stderr, "Closing input\n");
+    avio_close(input);
+}
+
+int main(int argc, char **argv)
+{
+    av_log_set_level(AV_LOG_TRACE);
+    AVDictionary *options = NULL;
+    AVIOContext *client = NULL, *server = NULL;
+    const char *in_uri, *out_uri;
+    int ret, pid;
+    if (argc < 3) {
+        printf("usage: %s input http://hostname[:port]\n"
+               "API example program to serve http to multiple clients.\n"
+               "\n", argv[0]);
+        return 1;
+    }
+
+    in_uri = argv[1];
+    out_uri = argv[2];
+
+    av_register_all();
+    avformat_network_init();
+
+    if ((ret = av_dict_set(&options, "listen", "2", 0)) < 0) {
+        fprintf(stderr, "Failed to set listen mode for server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    if ((ret = avio_open2(&server, out_uri, AVIO_FLAG_WRITE, NULL, &options)) < 0) {
+        fprintf(stderr, "Failed to open server: %s\n", av_err2str(ret));
+        return ret;
+    }
+    fprintf(stderr, "Entering main loop.\n");
+    for(;;) {
+        if ((ret = avio_accept(server, &client)) < 0)
+            goto end;
+        fprintf(stderr, "Accepted client, forking process.\n");
+        // XXX: Since we don't reap our children and don't ignore signals
+        //      this produces zombie processes.
+        pid = fork();
+        if (pid < 0) {
+            perror("Fork failed");
+            ret = AVERROR(errno);
+            goto end;
+        }
+        if (pid == 0) {
+            fprintf(stderr, "In child.\n");
+            process_client(client, in_uri);
+            avio_close(server);
+            exit(0);
+        }
+        if (pid > 0)
+            avio_close(client);
+    }
+end:
+    avio_close(server);
+    if (ret < 0 && ret != AVERROR_EOF) {
+        fprintf(stderr, "Some errors occurred: %s\n", av_err2str(ret));
+        return 1;
+    }
+    return 0;
+}

doc/examples/muxing.c

@@ -55,6 +55,7 @@ typedef struct OutputStream {
 
     /* pts of the next frame that will be generated */
     int64_t next_pts;
+    int samples_count;
 
     AVFrame *frame;
     AVFrame *tmp_frame;
@@ -62,6 +63,7 @@ typedef struct OutputStream {
     float t, tincr, tincr2;
 
     struct SwsContext *sws_ctx;
+    struct SwrContext *swr_ctx;
 } OutputStream;
 
 static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
@@ -92,6 +94,7 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
                        enum AVCodecID codec_id)
 {
     AVCodecContext *c;
+    int i;
 
     /* find the encoder */
     *codec = avcodec_find_encoder(codec_id);
@@ -115,8 +118,24 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
             (*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
         c->bit_rate    = 64000;
         c->sample_rate = 44100;
-        c->channels    = 2;
+        if ((*codec)->supported_samplerates) {
+            c->sample_rate = (*codec)->supported_samplerates[0];
+            for (i = 0; (*codec)->supported_samplerates[i]; i++) {
+                if ((*codec)->supported_samplerates[i] == 44100)
+                    c->sample_rate = 44100;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
         c->channel_layout = AV_CH_LAYOUT_STEREO;
+        if ((*codec)->channel_layouts) {
+            c->channel_layout = (*codec)->channel_layouts[0];
+            for (i = 0; (*codec)->channel_layouts[i]; i++) {
+                if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
+                    c->channel_layout = AV_CH_LAYOUT_STEREO;
+            }
+        }
+        c->channels        = av_get_channel_layout_nb_channels(c->channel_layout);
+        ost->st->time_base = (AVRational){ 1, c->sample_rate };
         break;
 
     case AVMEDIA_TYPE_VIDEO:
@@ -130,8 +149,9 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
          * of which frame timestamps are represented. For fixed-fps content,
          * timebase should be 1/framerate and timestamp increments should be
          * identical to 1. */
-        c->time_base.den = STREAM_FRAME_RATE;
-        c->time_base.num = 1;
+        ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
+        c->time_base       = ost->st->time_base;
+
         c->gop_size      = 12; /* emit one intra frame every twelve frames at most */
         c->pix_fmt       = STREAM_PIX_FMT;
         if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
@@ -152,25 +172,53 @@ static void add_stream(OutputStream *ost, AVFormatContext *oc,
 
     /* Some formats want stream headers to be separate. */
     if (oc->oformat->flags & AVFMT_GLOBALHEADER)
-        c->flags |= CODEC_FLAG_GLOBAL_HEADER;
+        c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 }
 
 /**************************************************************/
 /* audio output */
 
-int samples_count;
+static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
+                                  uint64_t channel_layout,
+                                  int sample_rate, int nb_samples)
+{
+    AVFrame *frame = av_frame_alloc();
+    int ret;
 
-struct SwrContext *swr_ctx = NULL;
+    if (!frame) {
+        fprintf(stderr, "Error allocating an audio frame\n");
+        exit(1);
+    }
 
-static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
+    frame->format = sample_fmt;
+    frame->channel_layout = channel_layout;
+    frame->sample_rate = sample_rate;
+    frame->nb_samples = nb_samples;
+
+    if (nb_samples) {
+        ret = av_frame_get_buffer(frame, 0);
+        if (ret < 0) {
+            fprintf(stderr, "Error allocating an audio buffer\n");
+            exit(1);
+        }
+    }
+
+    return frame;
+}
+
+static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
     AVCodecContext *c;
+    int nb_samples;
     int ret;
+    AVDictionary *opt = NULL;
 
     c = ost->st->codec;
 
     /* open it */
-    ret = avcodec_open2(c, codec, NULL);
+    av_dict_copy(&opt, opt_arg, 0);
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
     if (ret < 0) {
         fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
         exit(1);
@@ -182,84 +230,52 @@ static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
     /* increment frequency by 110 Hz per second */
     ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
 
-    ost->frame = av_frame_alloc();
-    if (!ost->frame)
-        exit(1);
-
-    ost->frame->sample_rate    = c->sample_rate;
-    ost->frame->format         = AV_SAMPLE_FMT_S16;
-    ost->frame->channel_layout = c->channel_layout;
-
-    if (c->codec->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE)
-        ost->frame->nb_samples = 10000;
+    if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
+        nb_samples = 10000;
     else
-        ost->frame->nb_samples = c->frame_size;
+        nb_samples = c->frame_size;
 
-    ost->tmp_frame = av_frame_alloc();
-    if (!ost->frame)
-        exit(1);
-
-    ost->tmp_frame->sample_rate    = c->sample_rate;
-    ost->tmp_frame->format         = c->sample_fmt;
-    ost->tmp_frame->channel_layout = c->channel_layout;
-    ost->tmp_frame->nb_samples     = ost->frame->nb_samples;
+    ost->frame     = alloc_audio_frame(c->sample_fmt, c->channel_layout,
+                                       c->sample_rate, nb_samples);
+    ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
+                                       c->sample_rate, nb_samples);
 
     /* create resampler context */
-    if (c->sample_fmt != AV_SAMPLE_FMT_S16) {
-        swr_ctx = swr_alloc();
-        if (!swr_ctx) {
+        ost->swr_ctx = swr_alloc();
+        if (!ost->swr_ctx) {
             fprintf(stderr, "Could not allocate resampler context\n");
             exit(1);
         }
 
         /* set options */
-        av_opt_set_int       (swr_ctx, "in_channel_count",   c->channels,       0);
-        av_opt_set_int       (swr_ctx, "in_sample_rate",     c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
-        av_opt_set_int       (swr_ctx, "out_channel_count",  c->channels,       0);
-        av_opt_set_int       (swr_ctx, "out_sample_rate",    c->sample_rate,    0);
-        av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
+        av_opt_set_int       (ost->swr_ctx, "in_channel_count",   c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "in_sample_rate",     c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt",      AV_SAMPLE_FMT_S16, 0);
+        av_opt_set_int       (ost->swr_ctx, "out_channel_count",  c->channels,       0);
+        av_opt_set_int       (ost->swr_ctx, "out_sample_rate",    c->sample_rate,    0);
+        av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt",     c->sample_fmt,     0);
 
         /* initialize the resampling context */
-        if ((ret = swr_init(swr_ctx)) < 0) {
+        if ((ret = swr_init(ost->swr_ctx)) < 0) {
             fprintf(stderr, "Failed to initialize the resampling context\n");
             exit(1);
         }
-    }
-
-    ret = av_frame_get_buffer(ost->frame, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate an audio frame.\n");
-        exit(1);
-    }
-    ret = av_frame_get_buffer(ost->tmp_frame, 0);
-    if (ret < 0) {
-        fprintf(stderr, "Could not allocate an audio frame.\n");
-        exit(1);
-    }
 }
 
 /* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
  * 'nb_channels' channels. */
 static AVFrame *get_audio_frame(OutputStream *ost)
 {
-    int j, i, v, ret;
-    int16_t *q = (int16_t*)ost->frame->data[0];
+    AVFrame *frame = ost->tmp_frame;
+    int j, i, v;
+    int16_t *q = (int16_t*)frame->data[0];
 
     /* check if we want to generate more frames */
     if (av_compare_ts(ost->next_pts, ost->st->codec->time_base,
                       STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
         return NULL;
 
-    /* when we pass a frame to the encoder, it may keep a reference to it
-     * internally;
-     * make sure we do not overwrite it here
-     */
-    ret = av_frame_make_writable(ost->frame);
-    if (ret < 0)
-        exit(1);
-
-    for (j = 0; j < ost->frame->nb_samples; j++) {
+    for (j = 0; j <frame->nb_samples; j++) {
         v = (int)(sin(ost->t) * 10000);
         for (i = 0; i < ost->st->codec->channels; i++)
             *q++ = v;
@@ -267,10 +283,10 @@ static AVFrame *get_audio_frame(OutputStream *ost)
         ost->tincr += ost->tincr2;
     }
 
-    ost->frame->pts = ost->next_pts;
-    ost->next_pts  += ost->frame->nb_samples;
+    frame->pts = ost->next_pts;
+    ost->next_pts  += frame->nb_samples;
 
-    return ost->frame;
+    return frame;
 }
 
 /*
@@ -293,27 +309,31 @@ static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
 
     if (frame) {
         /* convert samples from native format to destination codec format, using the resampler */
-        if (swr_ctx) {
             /* compute destination number of samples */
-            dst_nb_samples = av_rescale_rnd(swr_get_delay(swr_ctx, c->sample_rate) + frame->nb_samples,
+            dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
                                             c->sample_rate, c->sample_rate, AV_ROUND_UP);
             av_assert0(dst_nb_samples == frame->nb_samples);
 
+        /* when we pass a frame to the encoder, it may keep a reference to it
+         * internally;
+         * make sure we do not overwrite it here
+         */
+        ret = av_frame_make_writable(ost->frame);
+        if (ret < 0)
+            exit(1);
+
             /* convert to destination format */
-            ret = swr_convert(swr_ctx,
-                              ost->tmp_frame->data, dst_nb_samples,
+            ret = swr_convert(ost->swr_ctx,
+                              ost->frame->data, dst_nb_samples,
                               (const uint8_t **)frame->data, frame->nb_samples);
             if (ret < 0) {
                 fprintf(stderr, "Error while converting\n");
                 exit(1);
             }
-            frame = ost->tmp_frame;
-        } else {
-            dst_nb_samples = frame->nb_samples;
-        }
+            frame = ost->frame;
 
-        frame->pts = av_rescale_q(samples_count, (AVRational){1, c->sample_rate}, c->time_base);
-        samples_count += dst_nb_samples;
+        frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
+        ost->samples_count += dst_nb_samples;
     }
 
     ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
@@ -360,13 +380,17 @@ static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
     return picture;
 }
 
-static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost)
+static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
 {
     int ret;
     AVCodecContext *c = ost->st->codec;
+    AVDictionary *opt = NULL;
+
+    av_dict_copy(&opt, opt_arg, 0);
 
     /* open the codec */
-    ret = avcodec_open2(c, codec, NULL);
+    ret = avcodec_open2(c, codec, &opt);
+    av_dict_free(&opt);
     if (ret < 0) {
         fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
         exit(1);
@@ -523,6 +547,7 @@ static void close_stream(AVFormatContext *oc, OutputStream *ost)
     av_frame_free(&ost->frame);
     av_frame_free(&ost->tmp_frame);
     sws_freeContext(ost->sws_ctx);
+    swr_free(&ost->swr_ctx);
 }
 
 /**************************************************************/
@@ -538,11 +563,12 @@ int main(int argc, char **argv)
     int ret;
     int have_video = 0, have_audio = 0;
     int encode_video = 0, encode_audio = 0;
+    AVDictionary *opt = NULL;
 
     /* Initialize libavcodec, and register all codecs and formats. */
     av_register_all();
 
-    if (argc != 2) {
+    if (argc < 2) {
         printf("usage: %s output_file\n"
                "API example program to output a media file with libavformat.\n"
                "This program generates a synthetic audio and video stream, encodes and\n"
@@ -554,6 +580,9 @@ int main(int argc, char **argv)
     }
 
     filename = argv[1];
+    if (argc > 3 && !strcmp(argv[2], "-flags")) {
+        av_dict_set(&opt, argv[2]+1, argv[3], 0);
+    }
 
     /* allocate the output media context */
     avformat_alloc_output_context2(&oc, NULL, NULL, filename);
@@ -582,10 +611,10 @@ int main(int argc, char **argv)
     /* Now that all the parameters are set, we can open the audio and
      * video codecs and allocate the necessary encode buffers. */
     if (have_video)
-        open_video(oc, video_codec, &video_st);
+        open_video(oc, video_codec, &video_st, opt);
 
     if (have_audio)
-        open_audio(oc, audio_codec, &audio_st);
+        open_audio(oc, audio_codec, &audio_st, opt);
 
     av_dump_format(oc, 0, filename, 1);
 
@@ -600,7 +629,7 @@ int main(int argc, char **argv)
     }
 
     /* Write the stream header, if any. */
-    ret = avformat_write_header(oc, NULL);
+    ret = avformat_write_header(oc, &opt);
     if (ret < 0) {
         fprintf(stderr, "Error occurred when opening output file: %s\n",
                 av_err2str(ret));
@@ -632,7 +661,7 @@ int main(int argc, char **argv)
 
     if (!(fmt->flags & AVFMT_NOFILE))
         /* Close the output file. */
-        avio_close(oc->pb);
+        avio_closep(&oc->pb);
 
     /* free the stream */
     avformat_free_context(oc);

doc/examples/qsvdec.c

@@ -0,0 +1,484 @@
+/*
+ * Copyright (c) 2015 Anton Khirnov
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy
+ * of this software and associated documentation files (the "Software"), to deal
+ * in the Software without restriction, including without limitation the rights
+ * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+ * copies of the Software, and to permit persons to whom the Software is
+ * furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
+ * THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
+ * THE SOFTWARE.
+ */
+
+/**
+ * @file
+ * Intel QSV-accelerated H.264 decoding example.
+ *
+ * @example qsvdec.c
+ * This example shows how to do QSV-accelerated H.264 decoding with output
+ * frames in the VA-API video surfaces.
+ */
+
+#include "config.h"
+
+#include <stdio.h>
+
+#include <mfx/mfxvideo.h>
+
+#include <va/va.h>
+#include <va/va_x11.h>
+#include <X11/Xlib.h>
+
+#include "libavformat/avformat.h"
+#include "libavformat/avio.h"
+
+#include "libavcodec/avcodec.h"
+#include "libavcodec/qsv.h"
+
+#include "libavutil/error.h"
+#include "libavutil/mem.h"
+
+typedef struct DecodeContext {
+    mfxSession mfx_session;
+    VADisplay va_dpy;
+
+    VASurfaceID *surfaces;
+    mfxMemId    *surface_ids;
+    int         *surface_used;
+    int       nb_surfaces;
+
+    mfxFrameInfo frame_info;
+} DecodeContext;
+
+static mfxStatus frame_alloc(mfxHDL pthis, mfxFrameAllocRequest *req,
+                             mfxFrameAllocResponse *resp)
+{
+    DecodeContext *decode = pthis;
+    int err, i;
+
+    if (decode->surfaces) {
+        fprintf(stderr, "Multiple allocation requests.\n");
+        return MFX_ERR_MEMORY_ALLOC;
+    }
+    if (!(req->Type & MFX_MEMTYPE_VIDEO_MEMORY_DECODER_TARGET)) {
+        fprintf(stderr, "Unsupported surface type: %d\n", req->Type);
+        return MFX_ERR_UNSUPPORTED;
+    }
+    if (req->Info.BitDepthLuma != 8 || req->Info.BitDepthChroma != 8 ||
+        req->Info.Shift || req->Info.FourCC != MFX_FOURCC_NV12 ||
+        req->Info.ChromaFormat != MFX_CHROMAFORMAT_YUV420) {
+        fprintf(stderr, "Unsupported surface properties.\n");
+        return MFX_ERR_UNSUPPORTED;
+    }
+
+    decode->surfaces     = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surfaces));
+    decode->surface_ids  = av_malloc_array (req->NumFrameSuggested, sizeof(*decode->surface_ids));
+    decode->surface_used = av_mallocz_array(req->NumFrameSuggested, sizeof(*decode->surface_used));
+    if (!decode->surfaces || !decode->surface_ids || !decode->surface_used)
+        goto fail;
+
+    err = vaCreateSurfaces(decode->va_dpy, VA_RT_FORMAT_YUV420,
+                           req->Info.Width, req->Info.Height,
+                           decode->surfaces, req->NumFrameSuggested,
+                           NULL, 0);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Error allocating VA surfaces\n");
+        goto fail;
+    }
+    decode->nb_surfaces = req->NumFrameSuggested;
+
+    for (i = 0; i < decode->nb_surfaces; i++)
+        decode->surface_ids[i] = &decode->surfaces[i];
+
+    resp->mids           = decode->surface_ids;
+    resp->NumFrameActual = decode->nb_surfaces;
+
+    decode->frame_info = req->Info;
+
+    return MFX_ERR_NONE;
+fail:
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+
+    return MFX_ERR_MEMORY_ALLOC;
+}
+
+static mfxStatus frame_free(mfxHDL pthis, mfxFrameAllocResponse *resp)
+{
+    DecodeContext *decode = pthis;
+
+    if (decode->surfaces)
+        vaDestroySurfaces(decode->va_dpy, decode->surfaces, decode->nb_surfaces);
+    av_freep(&decode->surfaces);
+    av_freep(&decode->surface_ids);
+    av_freep(&decode->surface_used);
+    decode->nb_surfaces = 0;
+
+    return MFX_ERR_NONE;
+}
+
+static mfxStatus frame_lock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_unlock(mfxHDL pthis, mfxMemId mid, mfxFrameData *ptr)
+{
+    return MFX_ERR_UNSUPPORTED;
+}
+
+static mfxStatus frame_get_hdl(mfxHDL pthis, mfxMemId mid, mfxHDL *hdl)
+{
+    *hdl = mid;
+    return MFX_ERR_NONE;
+}
+
+static void free_buffer(void *opaque, uint8_t *data)
+{
+    int *used = opaque;
+    *used = 0;
+    av_freep(&data);
+}
+
+static int get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
+{
+    DecodeContext *decode = avctx->opaque;
+
+    mfxFrameSurface1 *surf;
+    AVBufferRef *surf_buf;
+    int idx;
+
+    for (idx = 0; idx < decode->nb_surfaces; idx++) {
+        if (!decode->surface_used[idx])
+            break;
+    }
+    if (idx == decode->nb_surfaces) {
+        fprintf(stderr, "No free surfaces\n");
+        return AVERROR(ENOMEM);
+    }
+
+    surf = av_mallocz(sizeof(*surf));
+    if (!surf)
+        return AVERROR(ENOMEM);
+    surf_buf = av_buffer_create((uint8_t*)surf, sizeof(*surf), free_buffer,
+                                &decode->surface_used[idx], AV_BUFFER_FLAG_READONLY);
+    if (!surf_buf) {
+        av_freep(&surf);
+        return AVERROR(ENOMEM);
+    }
+
+    surf->Info       = decode->frame_info;
+    surf->Data.MemId = &decode->surfaces[idx];
+
+    frame->buf[0]  = surf_buf;
+    frame->data[3] = (uint8_t*)surf;
+
+    decode->surface_used[idx] = 1;
+
+    return 0;
+}
+
+static int get_format(AVCodecContext *avctx, const enum AVPixelFormat *pix_fmts)
+{
+    while (*pix_fmts != AV_PIX_FMT_NONE) {
+        if (*pix_fmts == AV_PIX_FMT_QSV) {
+            if (!avctx->hwaccel_context) {
+                DecodeContext *decode = avctx->opaque;
+                AVQSVContext *qsv = av_qsv_alloc_context();
+                if (!qsv)
+                    return AV_PIX_FMT_NONE;
+
+                qsv->session   = decode->mfx_session;
+                qsv->iopattern = MFX_IOPATTERN_OUT_VIDEO_MEMORY;
+
+                avctx->hwaccel_context = qsv;
+            }
+
+            return AV_PIX_FMT_QSV;
+        }
+
+        pix_fmts++;
+    }
+
+    fprintf(stderr, "The QSV pixel format not offered in get_format()\n");
+
+    return AV_PIX_FMT_NONE;
+}
+
+static int decode_packet(DecodeContext *decode, AVCodecContext *decoder_ctx,
+                         AVFrame *frame, AVPacket *pkt,
+                         AVIOContext *output_ctx)
+{
+    int ret = 0;
+    int got_frame = 1;
+
+    while (pkt->size > 0 || (!pkt->data && got_frame)) {
+        ret = avcodec_decode_video2(decoder_ctx, frame, &got_frame, pkt);
+        if (ret < 0) {
+            fprintf(stderr, "Error during decoding\n");
+            return ret;
+        }
+
+        pkt->data += ret;
+        pkt->size -= ret;
+
+        /* A real program would do something useful with the decoded frame here.
+         * We just retrieve the raw data and write it to a file, which is rather
+         * useless but pedagogic. */
+        if (got_frame) {
+            mfxFrameSurface1 *surf = (mfxFrameSurface1*)frame->data[3];
+            VASurfaceID    surface = *(VASurfaceID*)surf->Data.MemId;
+
+            VAImageFormat img_fmt = {
+                .fourcc         = VA_FOURCC_NV12,
+                .byte_order     = VA_LSB_FIRST,
+                .bits_per_pixel = 8,
+                .depth          = 8,
+            };
+
+            VAImage img;
+
+            VAStatus err;
+            uint8_t *data;
+            int i, j;
+
+            img.buf      = VA_INVALID_ID;
+            img.image_id = VA_INVALID_ID;
+
+            err = vaCreateImage(decode->va_dpy, &img_fmt,
+                                frame->width, frame->height, &img);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error creating an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaGetImage(decode->va_dpy, surface, 0, 0,
+                             frame->width, frame->height,
+                             img.image_id);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error getting an image: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            err = vaMapBuffer(decode->va_dpy, img.buf, (void**)&data);
+            if (err != VA_STATUS_SUCCESS) {
+                fprintf(stderr, "Error mapping the image buffer: %s\n",
+                        vaErrorStr(err));
+                ret = AVERROR_UNKNOWN;
+                goto fail;
+            }
+
+            for (i = 0; i < img.num_planes; i++)
+                for (j = 0; j < (img.height >> (i > 0)); j++)
+                    avio_write(output_ctx, data + img.offsets[i] + j * img.pitches[i], img.width);
+
+fail:
+            if (img.buf != VA_INVALID_ID)
+                vaUnmapBuffer(decode->va_dpy, img.buf);
+            if (img.image_id != VA_INVALID_ID)
+                vaDestroyImage(decode->va_dpy, img.image_id);
+            av_frame_unref(frame);
+
+            if (ret < 0)
+                return ret;
+        }
+    }
+
+    return 0;
+}
+
+int main(int argc, char **argv)
+{
+    AVFormatContext *input_ctx = NULL;
+    AVStream *video_st = NULL;
+    AVCodecContext *decoder_ctx = NULL;
+    const AVCodec *decoder;
+
+    AVPacket pkt = { 0 };
+    AVFrame *frame = NULL;
+
+    DecodeContext decode = { NULL };
+
+    Display *dpy = NULL;
+    int va_ver_major, va_ver_minor;
+
+    mfxIMPL mfx_impl = MFX_IMPL_AUTO_ANY;
+    mfxVersion mfx_ver = { { 1, 1 } };
+
+    mfxFrameAllocator frame_allocator = {
+        .pthis = &decode,
+        .Alloc = frame_alloc,
+        .Lock  = frame_lock,
+        .Unlock = frame_unlock,
+        .GetHDL = frame_get_hdl,
+        .Free   = frame_free,
+    };
+
+    AVIOContext *output_ctx = NULL;
+
+    int ret, i, err;
+
+    av_register_all();
+
+    if (argc < 3) {
+        fprintf(stderr, "Usage: %s <input file> <output file>\n", argv[0]);
+        return 1;
+    }
+
+    /* open the input file */
+    ret = avformat_open_input(&input_ctx, argv[1], NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Cannot open input file '%s': ", argv[1]);
+        goto finish;
+    }
+
+    /* find the first H.264 video stream */
+    for (i = 0; i < input_ctx->nb_streams; i++) {
+        AVStream *st = input_ctx->streams[i];
+
+        if (st->codec->codec_id == AV_CODEC_ID_H264 && !video_st)
+            video_st = st;
+        else
+            st->discard = AVDISCARD_ALL;
+    }
+    if (!video_st) {
+        fprintf(stderr, "No H.264 video stream in the input file\n");
+        goto finish;
+    }
+
+    /* initialize VA-API */
+    dpy = XOpenDisplay(NULL);
+    if (!dpy) {
+        fprintf(stderr, "Cannot open the X display\n");
+        goto finish;
+    }
+    decode.va_dpy = vaGetDisplay(dpy);
+    if (!decode.va_dpy) {
+        fprintf(stderr, "Cannot open the VA display\n");
+        goto finish;
+    }
+
+    err = vaInitialize(decode.va_dpy, &va_ver_major, &va_ver_minor);
+    if (err != VA_STATUS_SUCCESS) {
+        fprintf(stderr, "Cannot initialize VA: %s\n", vaErrorStr(err));
+        goto finish;
+    }
+    fprintf(stderr, "Initialized VA v%d.%d\n", va_ver_major, va_ver_minor);
+
+    /* initialize an MFX session */
+    err = MFXInit(mfx_impl, &mfx_ver, &decode.mfx_session);
+    if (err != MFX_ERR_NONE) {
+        fprintf(stderr, "Error initializing an MFX session\n");
+        goto finish;
+    }
+
+    MFXVideoCORE_SetHandle(decode.mfx_session, MFX_HANDLE_VA_DISPLAY, decode.va_dpy);
+    MFXVideoCORE_SetFrameAllocator(decode.mfx_session, &frame_allocator);
+
+    /* initialize the decoder */
+    decoder = avcodec_find_decoder_by_name("h264_qsv");
+    if (!decoder) {
+        fprintf(stderr, "The QSV decoder is not present in libavcodec\n");
+        goto finish;
+    }
+
+    decoder_ctx = avcodec_alloc_context3(decoder);
+    if (!decoder_ctx) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+    decoder_ctx->codec_id = AV_CODEC_ID_H264;
+    if (video_st->codec->extradata_size) {
+        decoder_ctx->extradata = av_mallocz(video_st->codec->extradata_size +
+                                            AV_INPUT_BUFFER_PADDING_SIZE);
+        if (!decoder_ctx->extradata) {
+            ret = AVERROR(ENOMEM);
+            goto finish;
+        }
+        memcpy(decoder_ctx->extradata, video_st->codec->extradata,
+               video_st->codec->extradata_size);
+        decoder_ctx->extradata_size = video_st->codec->extradata_size;
+    }
+    decoder_ctx->refcounted_frames = 1;
+
+    decoder_ctx->opaque      = &decode;
+    decoder_ctx->get_buffer2 = get_buffer;
+    decoder_ctx->get_format  = get_format;
+
+    ret = avcodec_open2(decoder_ctx, NULL, NULL);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the decoder: ");
+        goto finish;
+    }
+
+    /* open the output stream */
+    ret = avio_open(&output_ctx, argv[2], AVIO_FLAG_WRITE);
+    if (ret < 0) {
+        fprintf(stderr, "Error opening the output context: ");
+        goto finish;
+    }
+
+    frame = av_frame_alloc();
+    if (!frame) {
+        ret = AVERROR(ENOMEM);
+        goto finish;
+    }
+
+    /* actual decoding */
+    while (ret >= 0) {
+        ret = av_read_frame(input_ctx, &pkt);
+        if (ret < 0)
+            break;
+
+        if (pkt.stream_index == video_st->index)
+            ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+        av_packet_unref(&pkt);
+    }
+
+    /* flush the decoder */
+    pkt.data = NULL;
+    pkt.size = 0;
+    ret = decode_packet(&decode, decoder_ctx, frame, &pkt, output_ctx);
+
+finish:
+    if (ret < 0) {
+        char buf[1024];
+        av_strerror(ret, buf, sizeof(buf));
+        fprintf(stderr, "%s\n", buf);
+    }
+
+    avformat_close_input(&input_ctx);
+
+    av_frame_free(&frame);
+
+    if (decode.mfx_session)
+        MFXClose(decode.mfx_session);
+    if (decode.va_dpy)
+        vaTerminate(decode.va_dpy);
+    if (dpy)
+        XCloseDisplay(dpy);
+
+    if (decoder_ctx)
+        av_freep(&decoder_ctx->hwaccel_context);
+    avcodec_free_context(&decoder_ctx);
+
+    avio_close(output_ctx);
+
+    return ret;
+}

doc/examples/remuxing.c

@@ -101,7 +101,7 @@ int main(int argc, char **argv)
         }
         out_stream->codec->codec_tag = 0;
         if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            out_stream->codec->flags |= CODEC_FLAG_GLOBAL_HEADER;
+            out_stream->codec->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
     }
     av_dump_format(ofmt_ctx, 0, out_filename, 1);
 
@@ -153,7 +153,7 @@ end:
 
     /* close output */
     if (ofmt_ctx && !(ofmt->flags & AVFMT_NOFILE))
-        avio_close(ofmt_ctx->pb);
+        avio_closep(&ofmt_ctx->pb);
     avformat_free_context(ofmt_ctx);
 
     if (ret < 0 && ret != AVERROR_EOF) {

doc/examples/resampling_audio.c

@@ -199,8 +199,7 @@ int main(int argc, char **argv)
             fmt, dst_ch_layout, dst_nb_channels, dst_rate, dst_filename);
 
 end:
-    if (dst_file)
-        fclose(dst_file);
+    fclose(dst_file);
 
     if (src_data)
         av_freep(&src_data[0]);

doc/examples/scaling_video.c

@@ -132,8 +132,7 @@ int main(int argc, char **argv)
            av_get_pix_fmt_name(dst_pix_fmt), dst_w, dst_h, dst_filename);
 
 end:
-    if (dst_file)
-        fclose(dst_file);
+    fclose(dst_file);
     av_freep(&src_data[0]);
     av_freep(&dst_data[0]);
     sws_freeContext(sws_ctx);

doc/examples/transcode_aac.c

@@ -41,18 +41,16 @@
 #include "libswresample/swresample.h"
 
 /** The output bit rate in kbit/s */
-#define OUTPUT_BIT_RATE 48000
+#define OUTPUT_BIT_RATE 96000
 /** The number of output channels */
 #define OUTPUT_CHANNELS 2
-/** The audio sample output format */
-#define OUTPUT_SAMPLE_FORMAT AV_SAMPLE_FMT_S16
 
 /**
  * Convert an error code into a text message.
  * @param error Error code to be converted
  * @return Corresponding error text (not thread-safe)
  */
-static char *const get_error_text(const int error)
+static const char *get_error_text(const int error)
 {
     static char error_buffer[255];
     av_strerror(error, error_buffer, sizeof(error_buffer));
@@ -169,7 +167,7 @@ static int open_output_file(const char *filename,
         goto cleanup;
     }
 
-    /** Save the encoder context for easiert access later. */
+    /** Save the encoder context for easier access later. */
     *output_codec_context = stream->codec;
 
     /**
@@ -179,15 +177,22 @@ static int open_output_file(const char *filename,
     (*output_codec_context)->channels       = OUTPUT_CHANNELS;
     (*output_codec_context)->channel_layout = av_get_default_channel_layout(OUTPUT_CHANNELS);
     (*output_codec_context)->sample_rate    = input_codec_context->sample_rate;
-    (*output_codec_context)->sample_fmt     = AV_SAMPLE_FMT_S16;
+    (*output_codec_context)->sample_fmt     = output_codec->sample_fmts[0];
     (*output_codec_context)->bit_rate       = OUTPUT_BIT_RATE;
 
+    /** Allow the use of the experimental AAC encoder */
+    (*output_codec_context)->strict_std_compliance = FF_COMPLIANCE_EXPERIMENTAL;
+
+    /** Set the sample rate for the container. */
+    stream->time_base.den = input_codec_context->sample_rate;
+    stream->time_base.num = 1;
+
     /**
      * Some container formats (like MP4) require global headers to be present
      * Mark the encoder so that it behaves accordingly.
      */
     if ((*output_format_context)->oformat->flags & AVFMT_GLOBALHEADER)
-        (*output_codec_context)->flags |= CODEC_FLAG_GLOBAL_HEADER;
+        (*output_codec_context)->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
     /** Open the encoder for the audio stream to use it later. */
     if ((error = avcodec_open2(*output_codec_context, output_codec, NULL)) < 0) {
@@ -199,7 +204,7 @@ static int open_output_file(const char *filename,
     return 0;
 
 cleanup:
-    avio_close((*output_format_context)->pb);
+    avio_closep(&(*output_format_context)->pb);
     avformat_free_context(*output_format_context);
     *output_format_context = NULL;
     return error < 0 ? error : AVERROR_EXIT;
@@ -271,10 +276,11 @@ static int init_resampler(AVCodecContext *input_codec_context,
 }
 
 /** Initialize a FIFO buffer for the audio samples to be encoded. */
-static int init_fifo(AVAudioFifo **fifo)
+static int init_fifo(AVAudioFifo **fifo, AVCodecContext *output_codec_context)
 {
     /** Create the FIFO buffer based on the specified output sample format. */
-    if (!(*fifo = av_audio_fifo_alloc(OUTPUT_SAMPLE_FORMAT, OUTPUT_CHANNELS, 1))) {
+    if (!(*fifo = av_audio_fifo_alloc(output_codec_context->sample_fmt,
+                                      output_codec_context->channels, 1))) {
         fprintf(stderr, "Could not allocate FIFO\n");
         return AVERROR(ENOMEM);
     }
@@ -306,7 +312,7 @@ static int decode_audio_frame(AVFrame *frame,
 
     /** Read one audio frame from the input file into a temporary packet. */
     if ((error = av_read_frame(input_format_context, &input_packet)) < 0) {
-        /** If we are the the end of the file, flush the decoder below. */
+        /** If we are at the end of the file, flush the decoder below. */
         if (error == AVERROR_EOF)
             *finished = 1;
         else {
@@ -537,6 +543,9 @@ static int init_output_frame(AVFrame **frame,
     return 0;
 }
 
+/** Global timestamp for the audio frames */
+static int64_t pts = 0;
+
 /** Encode one frame worth of audio to the output file. */
 static int encode_audio_frame(AVFrame *frame,
                               AVFormatContext *output_format_context,
@@ -548,6 +557,12 @@ static int encode_audio_frame(AVFrame *frame,
     int error;
     init_packet(&output_packet);
 
+    /** Set a timestamp based on the sample rate for the container. */
+    if (frame) {
+        frame->pts = pts;
+        pts += frame->nb_samples;
+    }
+
     /**
      * Encode the audio frame and store it in the temporary packet.
      * The output audio stream encoder is used to do this.
@@ -659,7 +674,7 @@ int main(int argc, char **argv)
                        &resample_context))
         goto cleanup;
     /** Initialize the FIFO buffer to store audio samples to be encoded. */
-    if (init_fifo(&fifo))
+    if (init_fifo(&fifo, output_codec_context))
         goto cleanup;
     /** Write the header of the output file container. */
     if (write_output_file_header(output_format_context))
@@ -743,7 +758,7 @@ cleanup:
     if (output_codec_context)
         avcodec_close(output_codec_context);
     if (output_format_context) {
-        avio_close(output_format_context->pb);
+        avio_closep(&output_format_context->pb);
         avformat_free_context(output_format_context);
     }
     if (input_codec_context)

doc/examples/transcoding.c

@@ -116,6 +116,10 @@ static int open_output_file(const char *filename)
                 || dec_ctx->codec_type == AVMEDIA_TYPE_AUDIO) {
             /* in this example, we choose transcoding to same codec */
             encoder = avcodec_find_encoder(dec_ctx->codec_id);
+            if (!encoder) {
+                av_log(NULL, AV_LOG_FATAL, "Necessary encoder not found\n");
+                return AVERROR_INVALIDDATA;
+            }
 
             /* In this example, we transcode to same properties (picture size,
              * sample rate etc.). These properties can be changed for output
@@ -157,7 +161,7 @@ static int open_output_file(const char *filename)
         }
 
         if (ofmt_ctx->oformat->flags & AVFMT_GLOBALHEADER)
-            enc_ctx->flags |= CODEC_FLAG_GLOBAL_HEADER;
+            enc_ctx->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
 
     }
     av_dump_format(ofmt_ctx, 0, filename, 1);
@@ -385,17 +389,9 @@ static int encode_write_frame(AVFrame *filt_frame, unsigned int stream_index, in
 
     /* prepare packet for muxing */
     enc_pkt.stream_index = stream_index;
-    enc_pkt.dts = av_rescale_q_rnd(enc_pkt.dts,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base,
-            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    enc_pkt.pts = av_rescale_q_rnd(enc_pkt.pts,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base,
-            AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-    enc_pkt.duration = av_rescale_q(enc_pkt.duration,
-            ofmt_ctx->streams[stream_index]->codec->time_base,
-            ofmt_ctx->streams[stream_index]->time_base);
+    av_packet_rescale_ts(&enc_pkt,
+                         ofmt_ctx->streams[stream_index]->codec->time_base,
+                         ofmt_ctx->streams[stream_index]->time_base);
 
     av_log(NULL, AV_LOG_DEBUG, "Muxing frame\n");
     /* mux encoded frame */
@@ -453,7 +449,7 @@ static int flush_encoder(unsigned int stream_index)
     int got_frame;
 
     if (!(ofmt_ctx->streams[stream_index]->codec->codec->capabilities &
-                CODEC_CAP_DELAY))
+                AV_CODEC_CAP_DELAY))
         return 0;
 
     while (1) {
@@ -509,14 +505,9 @@ int main(int argc, char **argv)
                 ret = AVERROR(ENOMEM);
                 break;
             }
-            packet.dts = av_rescale_q_rnd(packet.dts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ifmt_ctx->streams[stream_index]->codec->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            packet.pts = av_rescale_q_rnd(packet.pts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ifmt_ctx->streams[stream_index]->codec->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ifmt_ctx->streams[stream_index]->codec->time_base);
             dec_func = (type == AVMEDIA_TYPE_VIDEO) ? avcodec_decode_video2 :
                 avcodec_decode_audio4;
             ret = dec_func(ifmt_ctx->streams[stream_index]->codec, frame,
@@ -538,14 +529,9 @@ int main(int argc, char **argv)
             }
         } else {
             /* remux this frame without reencoding */
-            packet.dts = av_rescale_q_rnd(packet.dts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ofmt_ctx->streams[stream_index]->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
-            packet.pts = av_rescale_q_rnd(packet.pts,
-                    ifmt_ctx->streams[stream_index]->time_base,
-                    ofmt_ctx->streams[stream_index]->time_base,
-                    AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX);
+            av_packet_rescale_ts(&packet,
+                                 ifmt_ctx->streams[stream_index]->time_base,
+                                 ofmt_ctx->streams[stream_index]->time_base);
 
             ret = av_interleaved_write_frame(ofmt_ctx, &packet);
             if (ret < 0)
@@ -587,7 +573,7 @@ end:
     av_free(filter_ctx);
     avformat_close_input(&ifmt_ctx);
     if (ofmt_ctx && !(ofmt_ctx->oformat->flags & AVFMT_NOFILE))
-        avio_close(ofmt_ctx->pb);
+        avio_closep(&ofmt_ctx->pb);
     avformat_free_context(ofmt_ctx);
 
     if (ret < 0)

doc/faq.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg FAQ
 @titlepage
@@ -90,6 +91,56 @@ To build FFmpeg, you need to install the development package. It is usually
 called @file{libfoo-dev} or @file{libfoo-devel}. You can remove it after the
 build is finished, but be sure to keep the main package.
 
+@section How do I make @command{pkg-config} find my libraries?
+
+Somewhere along with your libraries, there is a @file{.pc} file (or several)
+in a @file{pkgconfig} directory. You need to set environment variables to
+point @command{pkg-config} to these files.
+
+If you need to @emph{add} directories to @command{pkg-config}'s search list
+(typical use case: library installed separately), add it to
+@code{$PKG_CONFIG_PATH}:
+
+@example
+export PKG_CONFIG_PATH=/opt/x264/lib/pkgconfig:/opt/opus/lib/pkgconfig
+@end example
+
+If you need to @emph{replace} @command{pkg-config}'s search list
+(typical use case: cross-compiling), set it in
+@code{$PKG_CONFIG_LIBDIR}:
+
+@example
+export PKG_CONFIG_LIBDIR=/home/me/cross/usr/lib/pkgconfig:/home/me/cross/usr/local/lib/pkgconfig
+@end example
+
+If you need to know the library's internal dependencies (typical use: static
+linking), add the @code{--static} option to @command{pkg-config}:
+
+@example
+./configure --pkg-config-flags=--static
+@end example
+
+@section How do I use @command{pkg-config} when cross-compiling?
+
+The best way is to install @command{pkg-config} in your cross-compilation
+environment. It will automatically use the cross-compilation libraries.
+
+You can also use @command{pkg-config} from the host environment by
+specifying explicitly @code{--pkg-config=pkg-config} to @command{configure}.
+In that case, you must point @command{pkg-config} to the correct directories
+using the @code{PKG_CONFIG_LIBDIR}, as explained in the previous entry.
+
+As an intermediate solution, you can place in your cross-compilation
+environment a script that calls the host @command{pkg-config} with
+@code{PKG_CONFIG_LIBDIR} set. That script can look like that:
+
+@example
+#!/bin/sh
+PKG_CONFIG_LIBDIR=/path/to/cross/lib/pkgconfig
+export PKG_CONFIG_LIBDIR
+exec /usr/bin/pkg-config "$@@"
+@end example
+
 @chapter Usage
 
 @section ffmpeg does not work; what is wrong?
@@ -298,7 +349,7 @@ FFmpeg has a @url{http://ffmpeg.org/ffmpeg-protocols.html#concat,
 @code{concat}} protocol designed specifically for that, with examples in the
 documentation.
 
-A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow to concatenate
+A few multimedia containers (MPEG-1, MPEG-2 PS, DV) allow one to concatenate
 video by merely concatenating the files containing them.
 
 Hence you may concatenate your multimedia files by first transcoding them to
@@ -416,6 +467,40 @@ point acceptable for your tastes. The most common options to do that are
 @option{-qscale} and @option{-qmax}, but you should peruse the documentation
 of the encoder you chose.
 
+@section I have a stretched video, why does scaling does not fix it?
+
+A lot of video codecs and formats can store the @emph{aspect ratio} of the
+video: this is the ratio between the width and the height of either the full
+image (DAR, display aspect ratio) or individual pixels (SAR, sample aspect
+ratio). For example, EGA screens at resolution 640×350 had 4:3 DAR and 35:48
+SAR.
+
+Most still image processing work with square pixels, i.e. 1:1 SAR, but a lot
+of video standards, especially from the analogic-numeric transition era, use
+non-square pixels.
+
+Most processing filters in FFmpeg handle the aspect ratio to avoid
+stretching the image: cropping adjusts the DAR to keep the SAR constant,
+scaling adjusts the SAR to keep the DAR constant.
+
+If you want to stretch, or “unstretch”, the image, you need to override the
+information with the
+@url{http://ffmpeg.org/ffmpeg-filters.html#setdar_002c-setsar, @code{setdar or setsar filters}}.
+
+Do not forget to examine carefully the original video to check whether the
+stretching comes from the image or from the aspect ratio information.
+
+For example, to fix a badly encoded EGA capture, use the following commands,
+either the first one to upscale to square pixels or the second one to set
+the correct aspect ratio or the third one to avoid transcoding (may not work
+depending on the format / codec / player / phase of the moon):
+
+@example
+ffmpeg -i ega_screen.nut -vf scale=640:480,setsar=1 ega_screen_scaled.nut
+ffmpeg -i ega_screen.nut -vf setdar=4/3 ega_screen_anamorphic.nut
+ffmpeg -i ega_screen.nut -aspect 4/3 -c copy ega_screen_overridden.nut
+@end example
+
 @chapter Development
 
 @section Are there examples illustrating how to use the FFmpeg libraries, particularly libavcodec and libavformat?

doc/fate.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Automated Testing Environment
 @titlepage
@@ -12,36 +13,36 @@
 
 @chapter Introduction
 
-  FATE is an extended regression suite on the client-side and a means
+FATE is an extended regression suite on the client-side and a means
 for results aggregation and presentation on the server-side.
 
-  The first part of this document explains how you can use FATE from
+The first part of this document explains how you can use FATE from
 your FFmpeg source directory to test your ffmpeg binary. The second
 part describes how you can run FATE to submit the results to FFmpeg's
 FATE server.
 
-  In any way you can have a look at the publicly viewable FATE results
+In any way you can have a look at the publicly viewable FATE results
 by visiting this website:
 
-  @url{http://fate.ffmpeg.org/}
+@url{http://fate.ffmpeg.org/}
 
-  This is especially recommended for all people contributing source
+This is especially recommended for all people contributing source
 code to FFmpeg, as it can be seen if some test on some platform broke
 with their recent contribution. This usually happens on the platforms
 the developers could not test on.
 
-  The second part of this document describes how you can run FATE to
+The second part of this document describes how you can run FATE to
 submit your results to FFmpeg's FATE server. If you want to submit your
 results be sure to check that your combination of CPU, OS and compiler
 is not already listed on the above mentioned website.
 
-  In the third part you can find a comprehensive listing of FATE makefile
+In the third part you can find a comprehensive listing of FATE makefile
 targets and variables.
 
 
 @chapter Using FATE from your FFmpeg source directory
 
-  If you want to run FATE on your machine you need to have the samples
+If you want to run FATE on your machine you need to have the samples
 in place. You can get the samples via the build target fate-rsync.
 Use this command from the top-level source directory:
 
@@ -50,11 +51,11 @@ make fate-rsync SAMPLES=fate-suite/
 make fate       SAMPLES=fate-suite/
 @end example
 
-  The above commands set the samples location by passing a makefile
+The above commands set the samples location by passing a makefile
 variable via command line. It is also possible to set the samples
 location at source configuration time by invoking configure with
-`--samples=<path to the samples directory>'. Afterwards you can
-invoke the makefile targets without setting the SAMPLES makefile
+@option{--samples=<path to the samples directory>}. Afterwards you can
+invoke the makefile targets without setting the @var{SAMPLES} makefile
 variable. This is illustrated by the following commands:
 
 @example
@@ -63,7 +64,7 @@ make fate-rsync
 make fate
 @end example
 
-  Yet another way to tell FATE about the location of the sample
+Yet another way to tell FATE about the location of the sample
 directory is by making sure the environment variable FATE_SAMPLES
 contains the path to your samples directory. This can be achieved
 by e.g. putting that variable in your shell profile or by setting
@@ -84,7 +85,7 @@ To use a custom wrapper to run the test, pass @option{--target-exec} to
 
 @chapter Submitting the results to the FFmpeg result aggregation server
 
-  To submit your results to the server you should run fate through the
+To submit your results to the server you should run fate through the
 shell script @file{tests/fate.sh} from the FFmpeg sources. This script needs
 to be invoked with a configuration file as its first argument.
 
@@ -92,23 +93,23 @@ to be invoked with a configuration file as its first argument.
 tests/fate.sh /path/to/fate_config
 @end example
 
-  A configuration file template with comments describing the individual
+A configuration file template with comments describing the individual
 configuration variables can be found at @file{doc/fate_config.sh.template}.
 
 @ifhtml
-  The mentioned configuration template is also available here:
+The mentioned configuration template is also available here:
 @verbatiminclude fate_config.sh.template
 @end ifhtml
 
-  Create a configuration that suits your needs, based on the configuration
-template. The `slot' configuration variable can be any string that is not
+Create a configuration that suits your needs, based on the configuration
+template. The @env{slot} configuration variable can be any string that is not
 yet used, but it is suggested that you name it adhering to the following
-pattern <arch>-<os>-<compiler>-<compiler version>. The configuration file
-itself will be sourced in a shell script, therefore all shell features may
-be used. This enables you to setup the environment as you need it for your
-build.
+pattern @samp{@var{arch}-@var{os}-@var{compiler}-@var{compiler version}}. The
+configuration file itself will be sourced in a shell script, therefore all
+shell features may be used. This enables you to setup the environment as you
+need it for your build.
 
-  For your first test runs the `fate_recv' variable should be empty or
+For your first test runs the @env{fate_recv} variable should be empty or
 commented out. This will run everything as normal except that it will omit
 the submission of the results to the server. The following files should be
 present in $workdir as specified in the configuration file:
@@ -121,29 +122,29 @@ present in $workdir as specified in the configuration file:
     @item version
 @end itemize
 
-  When you have everything working properly you can create an SSH key pair
+When you have everything working properly you can create an SSH key pair
 and send the public key to the FATE server administrator who can be contacted
 at the email address @email{fate-admin@@ffmpeg.org}.
 
-  Configure your SSH client to use public key authentication with that key
+Configure your SSH client to use public key authentication with that key
 when connecting to the FATE server. Also do not forget to check the identity
 of the server and to accept its host key. This can usually be achieved by
 running your SSH client manually and killing it after you accepted the key.
 The FATE server's fingerprint is:
 
-@table @option
+@table @samp
 @item RSA
    d3:f1:83:97:a4:75:2b:a6:fb:d6:e8:aa:81:93:97:51
 @item ECDSA
    76:9f:68:32:04:1e:d5:d4:ec:47:3f:dc:fc:18:17:86
 @end table
 
-  If you have problems connecting to the FATE server, it may help to try out
+If you have problems connecting to the FATE server, it may help to try out
 the @command{ssh} command with one or more @option{-v} options. You should
 get detailed output concerning your SSH configuration and the authentication
 process.
 
-  The only thing left is to automate the execution of the fate.sh script and
+The only thing left is to automate the execution of the fate.sh script and
 the synchronisation of the samples directory.
 
 
@@ -164,7 +165,7 @@ Run the FATE test suite (requires the fate-suite dataset).
 
 @section Makefile variables
 
-@table @option
+@table @env
 @item V
 Verbosity level, can be set to 0, 1 or 2.
     @itemize
@@ -182,20 +183,20 @@ Specify how many threads to use while running regression tests, it is
 quite useful to detect thread-related regressions.
 
 @item THREAD_TYPE
-Specify which threading strategy test, either @var{slice} or @var{frame},
-by default @var{slice+frame}
+Specify which threading strategy test, either @samp{slice} or @samp{frame},
+by default @samp{slice+frame}
 
 @item CPUFLAGS
 Specify CPU flags.
 
 @item TARGET_EXEC
 Specify or override the wrapper used to run the tests.
-The @var{TARGET_EXEC} option provides a way to run FATE wrapped in
+The @env{TARGET_EXEC} option provides a way to run FATE wrapped in
 @command{valgrind}, @command{qemu-user} or @command{wine} or on remote targets
 through @command{ssh}.
 
 @item GEN
-Set to @var{1} to generate the missing or mismatched references.
+Set to @samp{1} to generate the missing or mismatched references.
 @end table
 
 @section Examples

doc/fate_config.sh.template

@@ -1,5 +1,6 @@
 slot=                                    # some unique identifier
 repo=git://source.ffmpeg.org/ffmpeg.git  # the source repository
+#branch=release/2.6                       # the branch to test
 samples=                                 # path to samples directory
 workdir=                                 # directory in which to do all the work
 #fate_recv="ssh -T fate@fate.ffmpeg.org" # command to submit report

doc/ffmpeg-bitstream-filters.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Bitstream Filters Documentation
 @titlepage

doc/ffmpeg-codecs.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Codecs Documentation
 @titlepage

doc/ffmpeg-devices.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Devices Documentation
 @titlepage

doc/ffmpeg-filters.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Filters Documentation
 @titlepage

doc/ffmpeg-formats.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Formats Documentation
 @titlepage

doc/ffmpeg-protocols.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Protocols Documentation
 @titlepage

doc/ffmpeg-resampler.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Resampler Documentation
 @titlepage

doc/ffmpeg-scaler.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Scaler Documentation
 @titlepage

doc/ffmpeg-utils.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle FFmpeg Utilities Documentation
 @titlepage

doc/ffmpeg.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle ffmpeg Documentation
 @titlepage
@@ -11,7 +12,7 @@
 
 @chapter Synopsis
 
-ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_file}@} ... @{[@var{output_file_options}] @file{output_file}@} ...
+ffmpeg [@var{global_options}] @{[@var{input_file_options}] -i @file{input_url}@} ... @{[@var{output_file_options}] @file{output_url}@} ...
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -23,10 +24,10 @@ rates and resize video on the fly with a high quality polyphase filter.
 @command{ffmpeg} reads from an arbitrary number of input "files" (which can be regular
 files, pipes, network streams, grabbing devices, etc.), specified by the
 @code{-i} option, and writes to an arbitrary number of output "files", which are
-specified by a plain output filename. Anything found on the command line which
-cannot be interpreted as an option is considered to be an output filename.
+specified by a plain output url. Anything found on the command line which
+cannot be interpreted as an option is considered to be an output url.
 
-Each input or output file can, in principle, contain any number of streams of
+Each input or output url can, in principle, contain any number of streams of
 different types (video/audio/subtitle/attachment/data). The allowed number and/or
 types of streams may be limited by the container format. Selecting which
 streams from which inputs will go into which output is either done automatically
@@ -79,7 +80,7 @@ The format option may be needed for raw input files.
 The transcoding process in @command{ffmpeg} for each output can be described by
 the following diagram:
 
-@example
+@verbatim
  _______              ______________
 |       |            |              |
 | input |  demuxer   | encoded data |   decoder
@@ -98,7 +99,7 @@ the following diagram:
 |________|           |______________|
 
 
-@end example
+@end verbatim
 
 @command{ffmpeg} calls the libavformat library (containing demuxers) to read
 input files and get packets containing encoded data from them. When there are
@@ -123,7 +124,7 @@ Simple filtergraphs are those that have exactly one input and output, both of
 the same type. In the above diagram they can be represented by simply inserting
 an additional step between decoding and encoding:
 
-@example
+@verbatim
  _________                        ______________
 |         |                      |              |
 | decoded |                      | encoded data |
@@ -135,19 +136,19 @@ an additional step between decoding and encoding:
                 | frames   |
                 |__________|
 
-@end example
+@end verbatim
 
 Simple filtergraphs are configured with the per-stream @option{-filter} option
 (with @option{-vf} and @option{-af} aliases for video and audio respectively).
 A simple filtergraph for video can look for example like this:
 
-@example
+@verbatim
  _______        _____________        _______        ________
 |       |      |             |      |       |      |        |
 | input | ---> | deinterlace | ---> | scale | ---> | output |
 |_______|      |_____________|      |_______|      |________|
 
-@end example
+@end verbatim
 
 Note that some filters change frame properties but not frame contents. E.g. the
 @code{fps} filter in the example above changes number of frames, but does not
@@ -160,7 +161,7 @@ processing chain applied to one stream. This is the case, for example, when the
 more than one input and/or output, or when output stream type is different from
 input. They can be represented with the following diagram:
 
-@example
+@verbatim
  _________
 |         |
 | input 0 |\                    __________
@@ -178,7 +179,7 @@ input. They can be represented with the following diagram:
 | input 2 |/
 |_________|
 
-@end example
+@end verbatim
 
 Complex filtergraphs are configured with the @option{-filter_complex} option.
 Note that this option is global, since a complex filtergraph, by its nature,
@@ -197,14 +198,14 @@ step for the specified stream, so it does only demuxing and muxing. It is useful
 for changing the container format or modifying container-level metadata. The
 diagram above will, in this case, simplify to this:
 
-@example
+@verbatim
  _______              ______________            ________
 |       |            |              |          |        |
 | input |  demuxer   | encoded data |  muxer   | output |
 | file  | ---------> | packets      | -------> | file   |
 |_______|            |______________|          |________|
 
-@end example
+@end verbatim
 
 Since there is no decoding or encoding, it is very fast and there is no quality
 loss. However, it might not work in some cases because of many factors. Applying
@@ -242,8 +243,8 @@ Force input or output file format. The format is normally auto detected for inpu
 files and guessed from the file extension for output files, so this option is not
 needed in most cases.
 
-@item -i @var{filename} (@emph{input})
-input file name
+@item -i @var{url} (@emph{input})
+input file url
 
 @item -y (@emph{global})
 Overwrite output files without asking.
@@ -276,16 +277,18 @@ libx264, and the 138th audio, which will be encoded with libvorbis.
 When used as an input option (before @code{-i}), limit the @var{duration} of
 data read from the input file.
 
-When used as an output option (before an output filename), stop writing the
+When used as an output option (before an output url), stop writing the
 output after its duration reaches @var{duration}.
 
-@var{duration} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+@var{duration} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
 -to and -t are mutually exclusive and -t has priority.
 
 @item -to @var{position} (@emph{output})
 Stop writing the output at @var{position}.
-@var{position} may be a number in seconds, or in @code{hh:mm:ss[.xxx]} form.
+@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}.
 
 -to and -t are mutually exclusive and -t has priority.
 
@@ -294,17 +297,23 @@ Set the file size limit, expressed in bytes.
 
 @item -ss @var{position} (@emph{input/output})
 When used as an input option (before @code{-i}), seeks in this input file to
-@var{position}. Note the in most formats it is not possible to seek exactly, so
-@command{ffmpeg} will seek to the closest seek point before @var{position}.
+@var{position}. Note that in most formats it is not possible to seek exactly,
+so @command{ffmpeg} will seek to the closest seek point before @var{position}.
 When transcoding and @option{-accurate_seek} is enabled (the default), this
 extra segment between the seek point and @var{position} will be decoded and
 discarded. When doing stream copy or when @option{-noaccurate_seek} is used, it
 will be preserved.
 
-When used as an output option (before an output filename), decodes but discards
+When used as an output option (before an output url), decodes but discards
 input until the timestamps reach @var{position}.
 
-@var{position} may be either in seconds or in @code{hh:mm:ss[.xxx]} form.
+@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})
+
+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.
 
 @item -itsoffset @var{offset} (@emph{input})
 Set the input time offset.
@@ -319,7 +328,7 @@ the time duration specified in @var{offset}.
 @item -timestamp @var{date} (@emph{output})
 Set the recording timestamp in the container.
 
-@var{date} must be a time duration specification,
+@var{date} must be a date specification,
 see @ref{date syntax,,the Date section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 
 @item -metadata[:metadata_specifier] @var{key}=@var{value} (@emph{output,per-metadata})
@@ -339,7 +348,7 @@ ffmpeg -i in.avi -metadata title="my title" out.flv
 
 To set the language of the first audio stream:
 @example
-ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT
+ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT
 @end example
 
 @item -target @var{type} (@emph{output})
@@ -360,7 +369,7 @@ ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg
 @end example
 
 @item -dframes @var{number} (@emph{output})
-Set the number of data frames to record. This is an alias for @code{-frames:d}.
+Set the number of data frames to output. This is an alias for @code{-frames:d}.
 
 @item -frames[:@var{stream_specifier}] @var{framecount} (@emph{output,per-stream})
 Stop writing to the stream after @var{framecount} frames.
@@ -461,18 +470,24 @@ Technical note -- attachments are implemented as codec extradata, so this
 option can actually be used to extract extradata from any stream, not just
 attachments.
 
+@item -noautorotate
+Disable automatically rotating video based on file metadata.
+
 @end table
 
 @section Video Options
 
 @table @option
 @item -vframes @var{number} (@emph{output})
-Set the number of video frames to record. This is an alias for @code{-frames:v}.
+Set the number of video frames to output. This is an alias for @code{-frames:v}.
 @item -r[:@var{stream_specifier}] @var{fps} (@emph{input/output,per-stream})
 Set frame rate (Hz value, fraction or abbreviation).
 
 As an input option, ignore any timestamps stored in the file and instead
 generate timestamps assuming constant frame rate @var{fps}.
+This is not the same as the @option{-framerate} option used for some input formats
+like image2 or v4l2 (it used to be the same in older versions of FFmpeg).
+If in doubt use @option{-framerate} instead of the input option @option{-r}.
 
 As an output option, duplicate or drop input frames to achieve constant output
 frame rate @var{fps}.
@@ -683,13 +698,17 @@ is not specified, the value of the @var{DISPLAY} environment variable is used
 For DXVA2, this option should contain the number of the display adapter to use.
 If this option is not specified, the default adapter is used.
 @end table
+
+@item -hwaccels
+List all hardware acceleration methods supported in this build of ffmpeg.
+
 @end table
 
 @section Audio Options
 
 @table @option
 @item -aframes @var{number} (@emph{output})
-Set the number of audio frames to record. This is an alias for @code{-frames:a}.
+Set the number of audio frames to output. This is an alias for @code{-frames:a}.
 @item -ar[:@var{stream_specifier}] @var{freq} (@emph{input/output,per-stream})
 Set the audio sampling frequency. For output streams it is set by
 default to the frequency of the corresponding input stream. For input
@@ -821,8 +840,21 @@ To map all the streams except the second audio, use negative mappings
 ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT
 @end example
 
+To pick the English audio stream:
+@example
+ffmpeg -i INPUT -map 0:m:language:eng OUTPUT
+@end example
+
 Note that using this option disables the default mappings for this output file.
 
+@item -ignore_unknown
+Ignore input streams with unknown type instead of failing if copying
+such streams is attempted.
+
+@item -copy_unknown
+Allow input streams with unknown type to be copied instead of failing if copying
+such streams is attempted.
+
 @item -map_channel [@var{input_file_id}.@var{stream_specifier}.@var{channel_id}|-1][:@var{output_file_id}.@var{stream_specifier}]
 Map an audio channel from a given input to an output. If
 @var{output_file_id}.@var{stream_specifier} is not set, the audio channel will
@@ -986,6 +1018,13 @@ With -map you can select from which stream the timestamps should be
 taken. You can leave either video or audio unchanged and sync the
 remaining stream(s) to the unchanged one.
 
+@item -frame_drop_threshold @var{parameter}
+Frame drop threshold, which specifies how much behind video frames can
+be before they are dropped. In frame rate units, so 1.0 is one frame.
+The default is -1.1. One possible usecase is to avoid framedrops in case
+of noisy timestamps or to increase frame drop precision in case of exact
+timestamps.
+
 @item -async @var{samples_per_second}
 Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
 the parameter is the maximum samples per second by which the audio is changed.
@@ -1008,6 +1047,12 @@ processing (e.g. in case the format option @option{avoid_negative_ts}
 is enabled) the output timestamps may mismatch with the input
 timestamps even when this option is selected.
 
+@item -start_at_zero
+When used with @option{copyts}, shift input timestamps so they start at zero.
+
+This means that using e.g. @code{-ss 50} will make output timestamps start at
+50 seconds, regardless of what timestamp the input file started at.
+
 @item -copytb @var{mode}
 Specify how to set the encoder timebase when stream copying.  @var{mode} is an
 integer numeric value, and can assume one of the following values:
@@ -1049,7 +1094,7 @@ may be reassigned to a different value.
 For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for
 an output mpegts file:
 @example
-ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts
+ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts
 @end example
 
 @item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
@@ -1136,6 +1181,19 @@ This option enables or disables accurate seeking in input files with the
 transcoding. Use @option{-noaccurate_seek} to disable it, which may be useful
 e.g. when copying some streams and transcoding the others.
 
+@item -seek_timestamp (@emph{input})
+This option enables or disables seeking by timestamp in input files with the
+@option{-ss} option. It is disabled by default. If enabled, the argument
+to the @option{-ss} option is considered an actual timestamp, and is not
+offset by the start time of the file. This matters only for files which do
+not start from timestamp 0, such as transport streams.
+
+@item -thread_queue_size @var{size} (@emph{input})
+This option sets the maximum number of queued packets when reading from the
+file or device. With low latency / high rate live streams, packets may be
+discarded if they are not read in a timely manner; raising this value can
+avoid it.
+
 @item -override_ffserver (@emph{global})
 Overrides the input specifications from @command{ffserver}. Using this
 option you can map any input stream to @command{ffserver} and control
@@ -1146,6 +1204,11 @@ requested by @command{ffserver}.
 The option is intended for cases where features are needed that cannot be
 specified to @command{ffserver} but can be to @command{ffmpeg}.
 
+@item -sdp_file @var{file} (@emph{global})
+Print sdp information for an output stream to @var{file}.
+This allows dumping sdp information when at least one output isn't an
+rtp stream. (Requires at least one of the output formats to be rtp).
+
 @item -discard (@emph{input})
 Allows discarding specific streams or frames of streams at the demuxer.
 Not all demuxers support this.
@@ -1170,6 +1233,9 @@ Discard all frames excepts keyframes.
 Discard all frames.
 @end table
 
+@item -xerror (@emph{global})
+Stop and exit on error
+
 @end table
 
 As a special exception, you can use a bitmap subtitle stream as input: it
@@ -1195,7 +1261,10 @@ awkward to specify on the command line. Lines starting with the hash
 ('#') character are ignored and are used to provide comments. Check
 the @file{presets} directory in the FFmpeg source tree for examples.
 
-Preset files are specified with the @code{vpre}, @code{apre},
+There are two types of preset files: ffpreset and avpreset files.
+
+@subsection ffpreset files
+ffpreset files are specified with the @code{vpre}, @code{apre},
 @code{spre}, and @code{fpre} options. The @code{fpre} option takes the
 filename of the preset instead of a preset name as input and can be
 used for any kind of codec. For the @code{vpre}, @code{apre}, and
@@ -1220,67 +1289,31 @@ directories, where @var{codec_name} is the name of the codec to which
 the preset file options will be applied. For example, if you select
 the video codec with @code{-vcodec libvpx} and use @code{-vpre 1080p},
 then it will search for the file @file{libvpx-1080p.ffpreset}.
+
+@subsection avpreset files
+avpreset files are specified with the @code{pre} option. They work similar to
+ffpreset files, but they only allow encoder- specific options. Therefore, an
+@var{option}=@var{value} pair specifying an encoder cannot be used.
+
+When the @code{pre} option is specified, ffmpeg will look for files with the
+suffix .avpreset in the directories @file{$AVCONV_DATADIR} (if set), and
+@file{$HOME/.avconv}, and in the datadir defined at configuration time (usually
+@file{PREFIX/share/ffmpeg}), in that order.
+
+First ffmpeg searches for a file named @var{codec_name}-@var{arg}.avpreset in
+the above-mentioned directories, where @var{codec_name} is the name of the codec
+to which the preset file options will be applied. For example, if you select the
+video codec with @code{-vcodec libvpx} and use @code{-pre 1080p}, then it will
+search for the file @file{libvpx-1080p.avpreset}.
+
+If no such file is found, then ffmpeg will search for a file named
+@var{arg}.avpreset in the same directories.
+
 @c man end OPTIONS
 
-@chapter Tips
-@c man begin TIPS
-
-@itemize
-@item
-For streaming at very low bitrates, use a low frame rate
-and a small GOP size. This is especially true for RealVideo where
-the Linux player does not seem to be very fast, so it can miss
-frames. An example is:
-
-@example
-ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm
-@end example
-
-@item
-The parameter 'q' which is displayed while encoding is the current
-quantizer. The value 1 indicates that a very good quality could
-be achieved. The value 31 indicates the worst quality. If q=31 appears
-too often, it means that the encoder cannot compress enough to meet
-your bitrate. You must either increase the bitrate, decrease the
-frame rate or decrease the frame size.
-
-@item
-If your computer is not fast enough, you can speed up the
-compression at the expense of the compression ratio. You can use
-'-me zero' to speed up motion estimation, and '-g 0' to disable
-motion estimation completely (you have only I-frames, which means it
-is about as good as JPEG compression).
-
-@item
-To have very low audio bitrates, reduce the sampling frequency
-(down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).
-
-@item
-To have a constant quality (but a variable bitrate), use the option
-'-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
-quality).
-
-@end itemize
-@c man end TIPS
-
 @chapter Examples
 @c man begin EXAMPLES
 
-@section Preset files
-
-A preset file contains a sequence of @var{option=value} pairs, one for
-each line, specifying a sequence of options which can be specified also on
-the command line. Lines starting with the hash ('#') character are ignored and
-are used to provide comments. Empty lines are also ignored. Check the
-@file{presets} directory in the FFmpeg source tree for examples.
-
-Preset files are specified with the @code{pre} option, this option takes a
-preset name as input.  FFmpeg searches for a file named @var{preset_name}.avpreset in
-the directories @file{$AVCONV_DATADIR} (if set), and @file{$HOME/.ffmpeg}, and in
-the data directory defined at configuration time (usually @file{$PREFIX/share/ffmpeg})
-in that order.  For example, if the argument is @code{libx264-max}, it will
-search for the file @file{libx264-max.avpreset}.
-
 @section Video and Audio grabbing
 
 If you specify the input format and device then ffmpeg can grab video
@@ -1428,7 +1461,7 @@ combination with -ss to start extracting from a certain point in time.
 
 For creating a video from many images:
 @example
-ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi
+ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi
 @end example
 
 The syntax @code{foo-%03d.jpeg} specifies to use a decimal number
@@ -1443,18 +1476,18 @@ image2-specific @code{-pattern_type glob} option.
 For example, for creating a video from filenames matching the glob pattern
 @code{foo-*.jpeg}:
 @example
-ffmpeg -f image2 -pattern_type glob -i 'foo-*.jpeg' -r 12 -s WxH foo.avi
+ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi
 @end example
 
 @item
 You can put many streams of the same type in the output:
 
 @example
-ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut
+ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut
 @end example
 
-The resulting output file @file{test12.avi} will contain first four streams from
-the input file in reverse order.
+The resulting output file @file{test12.nut} will contain the first four streams
+from the input files in reverse order.
 
 @item
 To force CBR video output:

doc/ffplay.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle ffplay Documentation
 @titlepage
@@ -11,7 +12,7 @@
 
 @chapter Synopsis
 
-ffplay [@var{options}] [@file{input_file}]
+ffplay [@var{options}] [@file{input_url}]
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -37,14 +38,26 @@ Force displayed height.
 Set frame size (WxH or abbreviation), needed for videos which do
 not contain a header with the frame size like raw YUV.  This option
 has been deprecated in favor of private options, try -video_size.
+@item -fs
+Start in fullscreen mode.
 @item -an
 Disable audio.
 @item -vn
 Disable video.
+@item -sn
+Disable subtitles.
 @item -ss @var{pos}
-Seek to a given position in seconds.
+Seek to @var{pos}. Note that in most formats it is not possible to seek
+exactly, so @command{ffplay} will seek to the nearest seek point to
+@var{pos}.
+
+@var{pos} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 @item -t @var{duration}
-play <duration> seconds of audio/video
+Play @var{duration} seconds of audio/video.
+
+@var{duration} must be a time duration specification,
+see @ref{time duration syntax,,the Time duration section in the ffmpeg-utils(1) manual,ffmpeg-utils}.
 @item -bytes
 Seek by bytes.
 @item -nodisp
@@ -93,8 +106,8 @@ the input audio.
 Use the option "-filters" to show all the available filters (including
 sources and sinks).
 
-@item -i @var{input_file}
-Read @var{input_file}.
+@item -i @var{input_url}
+Read @var{input_url}.
 @end table
 
 @section Advanced options
@@ -109,15 +122,10 @@ duration, the codec parameters, the current position in the stream and
 the audio/video synchronisation drift. It is on by default, to
 explicitly disable it you need to specify @code{-nostats}.
 
-@item -bug
-Work around bugs.
 @item -fast
 Non-spec-compliant optimizations.
 @item -genpts
 Generate pts.
-@item -rtp_tcp
-Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful
-if you are streaming with the RTSP protocol.
 @item -sync @var{type}
 Set the master clock to audio (@code{type=audio}), video
 (@code{type=video}) or external (@code{type=ext}). Default is audio. The
@@ -125,23 +133,20 @@ master clock is used to control audio-video synchronization. Most media
 players use audio as master clock, but in some cases (streaming or high
 quality broadcast) it is necessary to change that. This option is mainly
 used for debugging purposes.
-@item -threads @var{count}
-Set the thread count.
-@item -ast @var{audio_stream_number}
-Select the desired audio stream number, counting from 0. The number
-refers to the list of all the input audio streams. If it is greater
-than the number of audio streams minus one, then the last one is
-selected, if it is negative the audio playback is disabled.
-@item -vst @var{video_stream_number}
-Select the desired video stream number, counting from 0. The number
-refers to the list of all the input video streams. If it is greater
-than the number of video streams minus one, then the last one is
-selected, if it is negative the video playback is disabled.
-@item -sst @var{subtitle_stream_number}
-Select the desired subtitle stream number, counting from 0. The number
-refers to the list of all the input subtitle streams. If it is greater
-than the number of subtitle streams minus one, then the last one is
-selected, if it is negative the subtitle rendering is disabled.
+@item -ast @var{audio_stream_specifier}
+Select the desired audio stream using the given stream specifier. The stream
+specifiers are described in the @ref{Stream specifiers} chapter. If this option
+is not specified, the "best" audio stream is selected in the program of the
+already selected video stream.
+@item -vst @var{video_stream_specifier}
+Select the desired video stream using the given stream specifier. The stream
+specifiers are described in the @ref{Stream specifiers} chapter. If this option
+is not specified, the "best" video stream is selected.
+@item -sst @var{subtitle_stream_specifier}
+Select the desired subtitle stream using the given stream specifier. The stream
+specifiers are described in the @ref{Stream specifiers} chapter. If this option
+is not specified, the "best" subtitle stream is selected in the program of the
+already selected video or audio stream.
 @item -autoexit
 Exit when video is done playing.
 @item -exitonkeydown
@@ -164,8 +169,20 @@ Force a specific video decoder.
 Force a specific subtitle decoder.
 
 @item -autorotate
-Automatically rotate the video according to presentation metadata. Set by
-default, use -noautorotate to disable.
+Automatically rotate the video according to file metadata. Enabled by
+default, use @option{-noautorotate} to disable it.
+
+@item -framedrop
+Drop video frames if video is out of sync. Enabled by default if the master
+clock is not set to video. Use this option to enable frame dropping for all
+master clock sources, use @option{-noframedrop} to disable it.
+
+@item -infbuf
+Do not limit the input buffer size, read as much data as possible from the
+input as soon as possible. Enabled by default for realtime streams, where data
+may be dropped if not read in time. Use this option to enable infinite buffers
+for all inputs, use @option{-noinfbuf} to disable it.
+
 @end table
 
 @section While playing

doc/ffprobe.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle ffprobe Documentation
 @titlepage
@@ -11,7 +12,7 @@
 
 @chapter Synopsis
 
-ffprobe [@var{options}] [@file{input_file}]
+ffprobe [@var{options}] [@file{input_url}]
 
 @chapter Description
 @c man begin DESCRIPTION
@@ -23,8 +24,8 @@ For example it can be used to check the format of the container used
 by a multimedia stream and the format and type of each media stream
 contained in it.
 
-If a filename is specified in input, ffprobe will try to open and
-probe the file content. If the file cannot be opened or recognized as
+If a url is specified in input, ffprobe will try to open and
+probe the url content. If the url cannot be opened or recognized as
 a multimedia file, a positive exit code is returned.
 
 ffprobe may be employed both as a standalone application or in
@@ -119,6 +120,10 @@ Show payload data, as a hexadecimal and ASCII dump. Coupled with
 
 The dump is printed as the "data" field. It may contain newlines.
 
+@item -show_data_hash @var{algorithm}
+Show a hash of payload data, for packets with @option{-show_packets} and for
+codec extradata with @option{-show_streams}.
+
 @item -show_error
 Show information about the error found when trying to probe the input.
 
@@ -180,7 +185,7 @@ format : stream=codec_type
 
 To show all the tags in the stream and format sections:
 @example
-format_tags : format_tags
+stream_tags : format_tags
 @end example
 
 To show only the @code{title} tag (if available) in the stream
@@ -317,12 +322,18 @@ Show information related to program and library versions. This is the
 equivalent of setting both @option{-show_program_version} and
 @option{-show_library_versions} options.
 
+@item -show_pixel_formats
+Show information about all pixel formats supported by FFmpeg.
+
+Pixel format information for each format is printed within a section
+with name "PIXEL_FORMAT".
+
 @item -bitexact
 Force bitexact output, useful to produce output which is not dependent
 on the specific build.
 
-@item -i @var{input_file}
-Read @var{input_file}.
+@item -i @var{input_url}
+Read @var{input_url}.
 
 @end table
 @c man end
@@ -436,17 +447,17 @@ writer).
 It can assume one of the following values:
 @table @option
 @item c
-Perform C-like escaping. Strings containing a newline ('\n'), carriage
-return ('\r'), a tab ('\t'), a form feed ('\f'), the escaping
-character ('\') or the item separator character @var{SEP} are escaped using C-like fashioned
-escaping, so that a newline is converted to the sequence "\n", a
-carriage return to "\r", '\' to "\\" and the separator @var{SEP} is
-converted to "\@var{SEP}".
+Perform C-like escaping. Strings containing a newline (@samp{\n}), carriage
+return (@samp{\r}), a tab (@samp{\t}), a form feed (@samp{\f}), the escaping
+character (@samp{\}) or the item separator character @var{SEP} are escaped
+using C-like fashioned escaping, so that a newline is converted to the
+sequence @samp{\n}, a carriage return to @samp{\r}, @samp{\} to @samp{\\} and
+the separator @var{SEP} is converted to @samp{\@var{SEP}}.
 
 @item csv
 Perform CSV-like escaping, as described in RFC4180.  Strings
-containing a newline ('\n'), a carriage return ('\r'), a double quote
-('"'), or @var{SEP} are enclosed in double-quotes.
+containing a newline (@samp{\n}), a carriage return (@samp{\r}), a double quote
+(@samp{"}), or @var{SEP} are enclosed in double-quotes.
 
 @item none
 Perform no escaping.
@@ -474,7 +485,7 @@ The description of the accepted options follows.
 Separator character used to separate the chapter, the section name, IDs and
 potential tags in the printed field key.
 
-Default value is '.'.
+Default value is @samp{.}.
 
 @item hierarchical, h
 Specify if the section name specification should be hierarchical. If
@@ -496,21 +507,22 @@ The following conventions are adopted:
 @item
 all key and values are UTF-8
 @item
-'.' is the subgroup separator
+@samp{.} is the subgroup separator
 @item
-newline, '\t', '\f', '\b' and the following characters are escaped
+newline, @samp{\t}, @samp{\f}, @samp{\b} and the following characters are
+escaped
 @item
-'\' is the escape character
+@samp{\} is the escape character
 @item
-'#' is the comment indicator
+@samp{#} is the comment indicator
 @item
-'=' is the key/value separator
+@samp{=} is the key/value separator
 @item
-':' is not used but usually parsed as key/value separator
+@samp{:} is not used but usually parsed as key/value separator
 @end itemize
 
 This writer accepts options as a list of @var{key}=@var{value} pairs,
-separated by ":".
+separated by @samp{:}.
 
 The description of the accepted options follows.
 

doc/ffprobe.xsd

@@ -10,8 +10,10 @@
         <xsd:sequence>
             <xsd:element name="program_version"  type="ffprobe:programVersionType"  minOccurs="0" maxOccurs="1" />
             <xsd:element name="library_versions" type="ffprobe:libraryVersionsType" minOccurs="0" maxOccurs="1" />
+            <xsd:element name="pixel_formats"    type="ffprobe:pixelFormatsType"    minOccurs="0" maxOccurs="1" />
             <xsd:element name="packets"  type="ffprobe:packetsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="frames"   type="ffprobe:framesType"  minOccurs="0" maxOccurs="1" />
+            <xsd:element name="packets_and_frames" type="ffprobe:packetsAndFramesType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="programs" type="ffprobe:programsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="streams"  type="ffprobe:streamsType" minOccurs="0" maxOccurs="1" />
             <xsd:element name="chapters" type="ffprobe:chaptersType" minOccurs="0" maxOccurs="1" />
@@ -35,7 +37,22 @@
         </xsd:sequence>
     </xsd:complexType>
 
+    <xsd:complexType name="packetsAndFramesType">
+        <xsd:sequence>
+            <xsd:choice minOccurs="0" maxOccurs="unbounded">
+                <xsd:element name="packet" type="ffprobe:packetType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="frame" type="ffprobe:frameType" minOccurs="0" maxOccurs="unbounded"/>
+                <xsd:element name="subtitle" type="ffprobe:subtitleType" minOccurs="0" maxOccurs="unbounded"/>
+            </xsd:choice>
+        </xsd:sequence>
+    </xsd:complexType>
+
     <xsd:complexType name="packetType">
+      <xsd:sequence>
+        <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
+      </xsd:sequence>
+
       <xsd:attribute name="codec_type"    type="xsd:string" use="required" />
       <xsd:attribute name="stream_index"  type="xsd:int" use="required" />
       <xsd:attribute name="pts"           type="xsd:long"  />
@@ -50,14 +67,27 @@
       <xsd:attribute name="pos"           type="xsd:long"  />
       <xsd:attribute name="flags"         type="xsd:string" use="required" />
       <xsd:attribute name="data"          type="xsd:string" />
+      <xsd:attribute name="data_hash"     type="xsd:string" />
+    </xsd:complexType>
+
+    <xsd:complexType name="packetSideDataListType">
+        <xsd:sequence>
+            <xsd:element name="side_data" type="ffprobe:packetSideDataType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="packetSideDataType">
+        <xsd:attribute name="side_data_type"              type="xsd:string"/>
+        <xsd:attribute name="side_data_size"              type="xsd:int"   />
     </xsd:complexType>
 
     <xsd:complexType name="frameType">
       <xsd:sequence>
             <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+            <xsd:element name="side_data_list" type="ffprobe:frameSideDataListType"   minOccurs="0" maxOccurs="1" />
       </xsd:sequence>
 
       <xsd:attribute name="media_type"    type="xsd:string" use="required"/>
+      <xsd:attribute name="stream_index"  type="xsd:int"  />
       <xsd:attribute name="key_frame"     type="xsd:int"    use="required"/>
       <xsd:attribute name="pts"           type="xsd:long" />
       <xsd:attribute name="pts_time"      type="xsd:float"/>
@@ -91,6 +121,16 @@
       <xsd:attribute name="repeat_pict"            type="xsd:int"   />
     </xsd:complexType>
 
+    <xsd:complexType name="frameSideDataListType">
+        <xsd:sequence>
+            <xsd:element name="side_data" type="ffprobe:frameSideDataType" minOccurs="1" maxOccurs="unbounded"/>
+        </xsd:sequence>
+    </xsd:complexType>
+    <xsd:complexType name="frameSideDataType">
+        <xsd:attribute name="side_data_type"              type="xsd:string"/>
+        <xsd:attribute name="side_data_size"              type="xsd:int"   />
+    </xsd:complexType>
+
     <xsd:complexType name="subtitleType">
       <xsd:attribute name="media_type"         type="xsd:string" fixed="subtitle" use="required"/>
       <xsd:attribute name="pts"                type="xsd:long" />
@@ -131,6 +171,7 @@
       <xsd:sequence>
         <xsd:element name="disposition" type="ffprobe:streamDispositionType" minOccurs="0" maxOccurs="1"/>
         <xsd:element name="tag" type="ffprobe:tagType" minOccurs="0" maxOccurs="unbounded"/>
+        <xsd:element name="side_data_list" type="ffprobe:packetSideDataListType"   minOccurs="0" maxOccurs="1" />
       </xsd:sequence>
 
       <xsd:attribute name="index"            type="xsd:int" use="required"/>
@@ -142,10 +183,13 @@
       <xsd:attribute name="codec_tag"        type="xsd:string" use="required"/>
       <xsd:attribute name="codec_tag_string" type="xsd:string" use="required"/>
       <xsd:attribute name="extradata"        type="xsd:string" />
+      <xsd:attribute name="extradata_hash"   type="xsd:string" />
 
       <!-- video attributes -->
       <xsd:attribute name="width"                type="xsd:int"/>
       <xsd:attribute name="height"               type="xsd:int"/>
+      <xsd:attribute name="coded_width"          type="xsd:int"/>
+      <xsd:attribute name="coded_height"         type="xsd:int"/>
       <xsd:attribute name="has_b_frames"         type="xsd:int"/>
       <xsd:attribute name="sample_aspect_ratio"  type="xsd:string"/>
       <xsd:attribute name="display_aspect_ratio" type="xsd:string"/>
@@ -153,7 +197,11 @@
       <xsd:attribute name="level"                type="xsd:int"/>
       <xsd:attribute name="color_range"          type="xsd:string"/>
       <xsd:attribute name="color_space"          type="xsd:string"/>
+      <xsd:attribute name="color_transfer"       type="xsd:string"/>
+      <xsd:attribute name="color_primaries"      type="xsd:string"/>
+      <xsd:attribute name="chroma_location"      type="xsd:string"/>
       <xsd:attribute name="timecode"             type="xsd:string"/>
+      <xsd:attribute name="refs"                 type="xsd:int"/>
 
       <!-- audio attributes -->
       <xsd:attribute name="sample_fmt"       type="xsd:string"/>
@@ -172,6 +220,7 @@
       <xsd:attribute name="duration"         type="xsd:float"/>
       <xsd:attribute name="bit_rate"         type="xsd:int"/>
       <xsd:attribute name="max_bit_rate"     type="xsd:int"/>
+      <xsd:attribute name="bits_per_raw_sample" type="xsd:int"/>
       <xsd:attribute name="nb_frames"        type="xsd:int"/>
       <xsd:attribute name="nb_read_frames"   type="xsd:int"/>
       <xsd:attribute name="nb_read_packets"  type="xsd:int"/>
@@ -224,8 +273,8 @@
     <xsd:complexType name="programVersionType">
       <xsd:attribute name="version"          type="xsd:string" use="required"/>
       <xsd:attribute name="copyright"        type="xsd:string" use="required"/>
-      <xsd:attribute name="build_date"       type="xsd:string" use="required"/>
-      <xsd:attribute name="build_time"       type="xsd:string" use="required"/>
+      <xsd:attribute name="build_date"       type="xsd:string"/>
+      <xsd:attribute name="build_time"       type="xsd:string"/>
       <xsd:attribute name="compiler_ident"   type="xsd:string" use="required"/>
       <xsd:attribute name="configuration"    type="xsd:string" use="required"/>
     </xsd:complexType>
@@ -263,4 +312,45 @@
           <xsd:element name="library_version" type="ffprobe:libraryVersionType" minOccurs="0" maxOccurs="unbounded"/>
         </xsd:sequence>
     </xsd:complexType>
+
+    <xsd:complexType name="pixelFormatFlagsType">
+      <xsd:attribute name="big_endian" type="xsd:int" use="required"/>
+      <xsd:attribute name="palette"    type="xsd:int" use="required"/>
+      <xsd:attribute name="bitstream"  type="xsd:int" use="required"/>
+      <xsd:attribute name="hwaccel"    type="xsd:int" use="required"/>
+      <xsd:attribute name="planar"     type="xsd:int" use="required"/>
+      <xsd:attribute name="rgb"        type="xsd:int" use="required"/>
+      <xsd:attribute name="pseudopal"  type="xsd:int" use="required"/>
+      <xsd:attribute name="alpha"      type="xsd:int" use="required"/>
+    </xsd:complexType>
+
+    <xsd:complexType name="pixelFormatComponentType">
+      <xsd:attribute name="index"      type="xsd:int" use="required"/>
+      <xsd:attribute name="bit_depth"  type="xsd:int" use="required"/>
+    </xsd:complexType>
+
+    <xsd:complexType name="pixelFormatComponentsType">
+      <xsd:sequence>
+        <xsd:element name="component" type="ffprobe:pixelFormatComponentType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>
+
+    <xsd:complexType name="pixelFormatType">
+      <xsd:sequence>
+        <xsd:element name="flags" type="ffprobe:pixelFormatFlagsType" minOccurs="0" maxOccurs="1"/>
+        <xsd:element name="components" type="ffprobe:pixelFormatComponentsType" minOccurs="0" maxOccurs="1"/>
+      </xsd:sequence>
+
+      <xsd:attribute name="name"                type="xsd:string" use="required"/>
+      <xsd:attribute name="nb_components"       type="xsd:int"    use="required"/>
+      <xsd:attribute name="log2_chroma_w"       type="xsd:int"/>
+      <xsd:attribute name="log2_chroma_h"       type="xsd:int"/>
+      <xsd:attribute name="bits_per_pixel"      type="xsd:int"/>
+    </xsd:complexType>
+
+    <xsd:complexType name="pixelFormatsType">
+      <xsd:sequence>
+        <xsd:element name="pixel_format" type="ffprobe:pixelFormatType" minOccurs="0" maxOccurs="unbounded"/>
+      </xsd:sequence>
+    </xsd:complexType>
 </xsd:schema>

doc/ffserver.conf

@@ -1,11 +1,11 @@
 # Port on which the server is listening. You must select a different
 # port from your standard HTTP web server if it is running on the same
 # computer.
-Port 8090
+HTTPPort 8090
 
 # Address on which the server is bound. Only useful if you have
 # several network interfaces.
-BindAddress 0.0.0.0
+HTTPBindAddress 0.0.0.0
 
 # Number of simultaneous HTTP connections that can be handled. It has
 # to be defined *before* the MaxClients parameter, since it defines the
@@ -82,6 +82,7 @@ Feed feed1.ffm
 # ra         : RealNetworks-compatible stream. Audio only.
 # mpjpeg     : Multipart JPEG (works with Netscape without any plugin)
 # jpeg       : Generate a single JPEG image.
+# mjpeg      : Generate a M-JPEG stream.
 # asf        : ASF compatible streaming (Windows Media Player format).
 # swf        : Macromedia Flash compatible stream
 # avi        : AVI format (MPEG-4 video, MPEG audio sound)

doc/ffserver.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle ffserver Documentation
 @titlepage
@@ -66,12 +67,12 @@ http://@var{ffserver_ip_address}:@var{http_port}/@var{feed_name}
 
 where @var{ffserver_ip_address} is the IP address of the machine where
 @command{ffserver} is installed, @var{http_port} is the port number of
-the HTTP server (configured through the @option{Port} option), and
+the HTTP server (configured through the @option{HTTPPort} option), and
 @var{feed_name} is the name of the corresponding feed defined in the
 configuration file.
 
 Each feed is associated to a file which is stored on disk. This stored
-file is used to allow to send pre-recorded data to a player as fast as
+file is used to send pre-recorded data to a player as fast as
 possible when new content is added in real-time to the stream.
 
 A "live-stream" or "stream" is a resource published by
@@ -101,7 +102,7 @@ http://@var{ffserver_ip_address}:@var{rtsp_port}/@var{stream_name}[@var{options}
 the configuration file. @var{options} is a list of options specified
 after the URL which affects how the stream is served by
 @command{ffserver}. @var{http_port} and @var{rtsp_port} are the HTTP
-and RTSP ports configured with the options @var{Port} and
+and RTSP ports configured with the options @var{HTTPPort} and
 @var{RTSPPort} respectively.
 
 In case the stream is associated to a feed, the encoding parameters
@@ -117,7 +118,8 @@ Multiple streams can be connected to the same feed.
 
 For example, you can have a situation described by the following
 graph:
-@example
+
+@verbatim
                _________       __________
               |         |     |          |
 ffmpeg 1 -----| feed 1  |-----| stream 1 |
@@ -142,7 +144,8 @@ ffmpeg 2 -----| feed 3  |-----| stream 4 |
               |         |     |          |
               | file 1  |-----| stream 5 |
               |_________|     |__________|
-@end example
+
+@end verbatim
 
 @anchor{FFM}
 @section FFM, FFM2 formats
@@ -203,11 +206,9 @@ WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to
 transfer the entire file before starting to play.
 The same is true of AVI files.
 
-@section What happens next?
-
-You should edit the ffserver.conf file to suit your needs (in terms of
-frame rates etc). Then install ffserver and ffmpeg, write a script to start
-them up, and off you go.
+You should edit the @file{ffserver.conf} file to suit your needs (in
+terms of frame rates etc). Then install @command{ffserver} and
+@command{ffmpeg}, write a script to start them up, and off you go.
 
 @section What else can it do?
 
@@ -354,20 +355,29 @@ allow everybody else.
 
 @section Global options
 @table @option
+@item HTTPPort @var{port_number}
 @item Port @var{port_number}
 @item RTSPPort @var{port_number}
 
-Set TCP port number on which the HTTP/RTSP server is listening. You
-must select a different port from your standard HTTP web server if it
-is running on the same computer.
+@var{HTTPPort} sets the HTTP server listening TCP port number,
+@var{RTSPPort} sets the RTSP server listening TCP port number.
+
+@var{Port} is the equivalent of @var{HTTPPort} and is deprecated.
+
+You must select a different port from your standard HTTP web server if
+it is running on the same computer.
 
 If not specified, no corresponding server will be created.
 
+@item HTTPBindAddress @var{ip_address}
 @item BindAddress @var{ip_address}
 @item RTSPBindAddress @var{ip_address}
 Set address on which the HTTP/RTSP server is bound. Only useful if you
 have several network interfaces.
 
+@var{BindAddress} is the equivalent of @var{HTTPBindAddress} and is
+deprecated.
+
 @item MaxHTTPConnections @var{n}
 Set number of simultaneous HTTP connections that can be handled. It
 has to be defined @emph{before} the @option{MaxClients} parameter,
@@ -401,6 +411,12 @@ ignored, and the log is written to standard output.
 Set no-daemon mode. This option is currently ignored since now
 @command{ffserver} will always work in no-daemon mode, and is
 deprecated.
+
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the all streams or not.
+Each stream may overwrite this setting for its own. Default is @var{UseDefaults}.
+The lastest occurrence overrides previous if multiple definitions.
 @end table
 
 @section Feed section
@@ -564,6 +580,11 @@ deprecated in favor of @option{Metadata}.
 @item Metadata @var{key} @var{value}
 Set metadata value on the output stream.
 
+@item UseDefaults
+@item NoDefaults
+Control whether default codec options are used for the stream or not.
+Default is @var{UseDefaults} unless disabled globally.
+
 @item NoAudio
 @item NoVideo
 Suppress audio/video.
@@ -582,8 +603,9 @@ Set sampling frequency for audio. When using low bitrates, you should
 lower this frequency to 22050 or 11025. The supported frequencies
 depend on the selected audio codec.
 
-@item AVOptionAudio @var{option} @var{value} (@emph{encoding,audio})
-Set generic option for audio stream.
+@item AVOptionAudio [@var{codec}:]@var{option} @var{value} (@emph{encoding,audio})
+Set generic or private option for audio stream.
+Private option must be prefixed with codec name or codec must be defined before.
 
 @item AVPresetAudio @var{preset} (@emph{encoding,audio})
 Set preset for audio stream.
@@ -660,8 +682,9 @@ Set video @option{qdiff} encoding option.
 @item DarkMask @var{float} (@emph{encoding,video})
 Set @option{lumi_mask}/@option{dark_mask} encoding options.
 
-@item AVOptionVideo @var{option} @var{value} (@emph{encoding,video})
-Set generic option for video stream.
+@item AVOptionVideo [@var{codec}:]@var{option} @var{value} (@emph{encoding,video})
+Set generic or private option for video stream.
+Private option must be prefixed with codec name or codec must be defined before.
 
 @item AVPresetVideo @var{preset} (@emph{encoding,video})
 Set preset for video stream.

doc/fftools-common-opts.texi

@@ -3,7 +3,7 @@ representing a number as input, which may be followed by one of the SI
 unit prefixes, for example: 'K', 'M', or 'G'.
 
 If 'i' is appended to the SI unit prefix, the complete prefix will be
-interpreted as a unit prefix for binary multiplies, which are based on
+interpreted as a unit prefix for binary multiples, which are based on
 powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
 prefix multiplies the value by 8. This allows using, for example:
 'KB', 'MiB', 'G' and 'B' as number suffixes.
@@ -36,8 +36,10 @@ Possible forms of stream specifiers are:
 Matches the stream with this index. E.g. @code{-threads:1 4} would set the
 thread count for the second stream to 4.
 @item @var{stream_type}[:@var{stream_index}]
-@var{stream_type} is one of following: 'v' for video, 'a' for audio, 's' for subtitle,
-'d' for data, and 't' for attachments. If @var{stream_index} is given, then it matches
+@var{stream_type} is one of following: 'v' or 'V' for video, 'a' for audio, 's'
+for subtitle, 'd' for data, and 't' for attachments. 'v' matches all video
+streams, 'V' only matches video streams which are not attached pictures, video
+thumbnails or cover arts.  If @var{stream_index} is given, then it matches
 stream number @var{stream_index} of this type. Otherwise, it matches all
 streams of this type.
 @item p:@var{program_id}[:@var{stream_index}]
@@ -46,6 +48,16 @@ in the program with the id @var{program_id}. Otherwise, it matches all streams i
 program.
 @item #@var{stream_id} or i:@var{stream_id}
 Match the stream by stream id (e.g. PID in MPEG-TS container).
+@item m:@var{key}[:@var{value}]
+Matches streams with the metadata tag @var{key} having the specified value. If
+@var{value} is not given, matches streams that contain the given tag with any
+value.
+@item u
+Matches streams with usable configuration, the codec must be defined and the
+essential information such as video dimension or audio sample rate must be present.
+
+Note that in @command{ffmpeg}, matching by metadata will only work properly for
+input files.
 @end table
 
 @section Generic options
@@ -96,7 +108,10 @@ Print detailed information about the filter name @var{filter_name}. Use the
 Show version.
 
 @item -formats
-Show available formats.
+Show available formats (including devices).
+
+@item -devices
+Show available devices.
 
 @item -codecs
 Show all codecs known to libavcodec.
@@ -131,6 +146,22 @@ Show channel names and standard channel layouts.
 @item -colors
 Show recognized color names.
 
+@item -sources @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
+Show autodetected sources of the intput device.
+Some devices may provide system-dependent source names that cannot be autodetected.
+The returned list cannot be assumed to be always complete.
+@example
+ffmpeg -sources pulse,server=192.168.0.4
+@end example
+
+@item -sinks @var{device}[,@var{opt1}=@var{val1}[,@var{opt2}=@var{val2}]...]
+Show autodetected sinks of the output device.
+Some devices may provide system-dependent sink names that cannot be autodetected.
+The returned list cannot be assumed to be always complete.
+@example
+ffmpeg -sinks pulse,server=192.168.0.4
+@end example
+
 @item -loglevel [repeat+]@var{loglevel} | -v [repeat+]@var{loglevel}
 Set the logging level used by the library.
 Adding "repeat+" indicates that repeated log output should not be compressed
@@ -139,28 +170,29 @@ omitted. "repeat" can also be used alone.
 If "repeat" is used alone, and with no prior loglevel set, the default
 loglevel will be used. If multiple loglevel parameters are given, using
 'repeat' will not change the loglevel.
-@var{loglevel} is a number or a string containing one of the following values:
+@var{loglevel} is a string or a number containing one of the following values:
 @table @samp
-@item quiet
+@item quiet, -8
 Show nothing at all; be silent.
-@item panic
+@item panic, 0
 Only show fatal errors which could lead the process to crash, such as
 and assert failure. This is not currently used for anything.
-@item fatal
+@item fatal, 8
 Only show fatal errors. These are errors after which the process absolutely
 cannot continue after.
-@item error
+@item error, 16
 Show all errors, including ones which can be recovered from.
-@item warning
+@item warning, 24
 Show all warnings and errors. Any message related to possibly
 incorrect or unexpected events will be shown.
-@item info
+@item info, 32
 Show informative messages during processing. This is in addition to
 warnings and errors. This is the default value.
-@item verbose
+@item verbose, 40
 Same as @code{info}, except more verbose.
-@item debug
+@item debug, 48
 Show everything, including debugging information.
+@item trace, 56
 @end table
 
 By default the program logs to stderr, if coloring is supported by the
@@ -178,21 +210,29 @@ directory.
 This file can be useful for bug reports.
 It also implies @code{-loglevel verbose}.
 
-Setting the environment variable @code{FFREPORT} to any value has the
+Setting the environment variable @env{FFREPORT} to any value has the
 same effect. If the value is a ':'-separated key=value sequence, these
-options will affect the report; options values must be escaped if they
+options will affect the report; option values must be escaped if they
 contain special characters or the options delimiter ':' (see the
-``Quoting and escaping'' section in the ffmpeg-utils manual). The
-following option is recognized:
+``Quoting and escaping'' section in the ffmpeg-utils manual).
+
+The following options are recognized:
 @table @option
 @item file
 set the file name to use for the report; @code{%p} is expanded to the name
 of the program, @code{%t} is expanded to a timestamp, @code{%%} is expanded
 to a plain @code{%}
 @item level
-set the log level
+set the log verbosity level using a numerical value (see @code{-loglevel}).
 @end table
 
+For example, to output a report to a file named @file{ffreport.log}
+using a log level of @code{32} (alias for log level @code{info}):
+
+@example
+FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
+@end example
+
 Errors in parsing the environment variable are not fatal, and will not
 appear in the report.
 
@@ -227,10 +267,14 @@ Possible flags for this option are:
 @item sse4.1
 @item sse4.2
 @item avx
+@item avx2
 @item xop
+@item fma3
 @item fma4
 @item 3dnow
 @item 3dnowext
+@item bmi1
+@item bmi2
 @item cmov
 @end table
 @item ARM
@@ -241,6 +285,13 @@ Possible flags for this option are:
 @item vfp
 @item vfpv3
 @item neon
+@item setend
+@end table
+@item AArch64
+@table @samp
+@item armv8
+@item vfp
+@item neon
 @end table
 @item PowerPC
 @table @samp
@@ -260,8 +311,41 @@ Possible flags for this option are:
 @end table
 
 @item -opencl_bench
-Benchmark all available OpenCL devices and show the results. This option
-is only available when FFmpeg has been compiled with @code{--enable-opencl}.
+This option is used to benchmark all available OpenCL devices and print the
+results. This option is only available when FFmpeg has been compiled with
+@code{--enable-opencl}.
+
+When FFmpeg is configured with @code{--enable-opencl}, the options for the
+global OpenCL context are set via @option{-opencl_options}. See the
+"OpenCL Options" section in the ffmpeg-utils manual for the complete list of
+supported options. Amongst others, these options include the ability to select
+a specific platform and device to run the OpenCL code on. By default, FFmpeg
+will run on the first device of the first platform. While the options for the
+global OpenCL context provide flexibility to the user in selecting the OpenCL
+device of their choice, most users would probably want to select the fastest
+OpenCL device for their system.
+
+This option assists the selection of the most efficient configuration by
+identifying the appropriate device for the user's system. The built-in
+benchmark is run on all the OpenCL devices and the performance is measured for
+each device. The devices in the results list are sorted based on their
+performance with the fastest device listed first. The user can subsequently
+invoke @command{ffmpeg} using the device deemed most appropriate via
+@option{-opencl_options} to obtain the best performance for the OpenCL
+accelerated code.
+
+Typical usage to use the fastest OpenCL device involve the following steps.
+
+Run the command:
+@example
+ffmpeg -opencl_bench
+@end example
+Note down the platform ID (@var{pidx}) and device ID (@var{didx}) of the first
+i.e. fastest device in the list.
+Select the platform and device using the command:
+@example
+ffmpeg -opencl_options platform_idx=@var{pidx}:device_idx=@var{didx} ...
+@end example
 
 @item -opencl_options options (@emph{global})
 Set OpenCL environment options. This option is only available when

doc/filter_design.txt

@@ -98,7 +98,7 @@ Buffer references ownership and permissions
     The AVFilterLink structure has a few AVFilterBufferRef fields. The
     cur_buf and out_buf were used with the deprecated
     start_frame/draw_slice/end_frame API and should no longer be used.
-    src_buf, cur_buf_copy and partial_buf are used by libavfilter internally
+    src_buf and partial_buf are used by libavfilter internally
     and must not be accessed by filters.
 
   Reference permissions

doc/formats.texi

@@ -23,7 +23,7 @@ Reduce buffering.
 
 @item probesize @var{integer} (@emph{input})
 Set probing size in bytes, i.e. the size of the data to analyze to get
-stream information. A higher value will allow to detect more
+stream information. A higher value will enable detecting more
 information in case it is dispersed into the stream, but will increase
 latency. Must be an integer not lesser than 32. It is 5000000 by default.
 
@@ -37,6 +37,8 @@ Possible values:
 @table @samp
 @item ignidx
 Ignore index.
+@item fastseek
+Enable fast, but inaccurate seeks for some formats.
 @item genpts
 Generate PTS.
 @item nofillin
@@ -55,6 +57,10 @@ Do not merge side data.
 Enable RTP MP4A-LATM payload.
 @item nobuffer
 Reduce the latency introduced by optional buffering
+@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.
 @end table
 
 @item seek2any @var{integer} (@emph{input})
@@ -63,7 +69,7 @@ Default is 0.
 
 @item analyzeduration @var{integer} (@emph{input})
 Specify how many microseconds are analyzed to probe the input. A
-higher value will allow to detect more accurate information, but will
+higher value will enable detecting more accurate information, but will
 increase latency. It defaults to 5,000,000 microseconds = 5 seconds.
 
 @item cryptokey @var{hexadecimal string} (@emph{input})
@@ -121,6 +127,25 @@ Consider all spec non compliancies as errors.
 Consider things that a sane encoder should not do as an error.
 @end table
 
+@item max_interleave_delta @var{integer} (@emph{output})
+Set maximum buffering duration for interleaving. The duration is
+expressed in microseconds, and defaults to 1000000 (1 second).
+
+To ensure all the streams are interleaved correctly, libavformat will
+wait until it has at least one packet for each stream before actually
+writing any packets to the output file. When some streams are
+"sparse" (i.e. there are large gaps between successive packets), this
+can result in excessive buffering.
+
+This field specifies the maximum difference between the timestamps of the
+first and the last packet in the muxing queue, above which libavformat
+will output a packet regardless of whether it has queued a packet for all
+the streams.
+
+If set to 0, libavformat will continue buffering packets until it has
+a packet for each stream, regardless of the maximum timestamp
+difference between the buffered packets.
+
 @item use_wallclock_as_timestamps @var{integer} (@emph{input})
 Use wallclock as timestamps.
 
@@ -168,6 +193,22 @@ The offset is added by the muxer to the output timestamps.
 Specifying a positive offset means that the corresponding streams are
 delayed bt the time duration specified in @var{offset}. Default value
 is @code{0} (meaning that no offset is applied).
+
+@item format_whitelist @var{list} (@emph{input})
+"," separated List of allowed demuxers. By default all are allowed.
+
+@item dump_separator @var{string} (@emph{input})
+Separator used to separate the fields printed on the command line about the
+Stream parameters.
+For example to separate the fields with newlines and indention:
+@example
+ffprobe -dump_separator "
+                          "  -i ~/videos/matrixbench_mpeg2.mpg
+@end example
+
+@item max_streams @var{integer} (@emph{input})
+Specifies the maximum number of streams. This can be used to reject files that
+would require too many resources due to a large number of streams.
 @end table
 
 @c man end FORMAT OPTIONS

doc/general.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle General Documentation
 @titlepage
@@ -108,6 +109,14 @@ Go to @url{http://www.wavpack.com/} and follow the instructions for
 installing the library. Then pass @code{--enable-libwavpack} to configure to
 enable it.
 
+@section OpenH264
+
+FFmpeg can make use of the OpenH264 library for H.264 encoding.
+
+Go to @url{http://www.openh264.org/} and follow the instructions for
+installing the library. Then pass @code{--enable-libopenh264} to configure to
+enable it.
+
 @section x264
 
 FFmpeg can make use of the x264 library for H.264 encoding.
@@ -130,12 +139,20 @@ Go to @url{http://x265.org/developers.html} and follow the instructions
 for installing the library. Then pass @code{--enable-libx265} to configure
 to enable it.
 
-@float note
+@float NOTE
 x265 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 kvazaar
+
+FFmpeg can make use of the kvazaar library for HEVC encoding.
+
+Go to @url{https://github.com/ultravideo/kvazaar} and follow the
+instructions for installing the library. Then pass
+@code{--enable-libkvazaar} to configure to enable it.
+
 @section libilbc
 
 iLBC is a narrowband speech codec that has been made freely available
@@ -143,7 +160,7 @@ by Google as part of the WebRTC project. libilbc is a packaging friendly
 copy of the iLBC codec. FFmpeg can make use of the libilbc library for
 iLBC encoding and decoding.
 
-Go to @url{https://github.com/dekkers/libilbc} and follow the instructions for
+Go to @url{https://github.com/TimothyGu/libilbc} and follow the instructions for
 installing the library. Then pass @code{--enable-libilbc} to configure to
 enable it.
 
@@ -170,8 +187,8 @@ included in compat/avisynth/, which allows the user to enable support
 without needing to search for these headers themselves.
 
 For Windows, supported AviSynth variants are
-@url{http://avisynth.nl, AviSynth 2.5 or 2.6} for 32-bit builds and
-@url{http://avs-plus.net, AviSynth+ 0.1} for 32-bit and 64-bit builds.
+@url{http://avisynth.nl, AviSynth 2.6 RC1 or higher} for 32-bit builds and
+@url{http://avs-plus.net, AviSynth+ r1718 or higher} for 32-bit and 64-bit builds.
 
 For Linux and OS X, the supported AviSynth variant is
 @url{https://github.com/avxsynth/avxsynth, AvxSynth}.
@@ -183,6 +200,17 @@ end user having AviSynth or AvxSynth installed - they'll only need to be
 installed to use AviSynth scripts (obviously).
 @end float
 
+@section Intel QuickSync Video
+
+FFmpeg can use Intel QuickSync Video (QSV) for accelerated encoding and decoding
+of multiple codecs. To use QSV, FFmpeg must be linked against the @code{libmfx}
+dispatcher, which loads the actual decoding libraries.
+
+The dispatcher is open source and can be downloaded from
+@url{https://github.com/lu-zero/mfx_dispatch.git}. FFmpeg needs to be configured
+with the @code{--enable-libmfx} option and @code{pkg-config} needs to be able to
+locate the dispatcher's @code{.pc} files.
+
 
 @chapter Supported File Formats, Codecs or Features
 
@@ -198,6 +226,10 @@ library:
 @item 4xm                       @tab   @tab X
     @tab 4X Technologies format, used in some games.
 @item 8088flex TMV              @tab   @tab X
+@item AAX                       @tab   @tab X
+    @tab Audible Enhanced Audio format, used in audiobooks.
+@item AA                        @tab   @tab X
+    @tab Audible Format 2, 3, and 4, used in audiobooks.
 @item ACT Voice                 @tab   @tab X
     @tab contains G.729 audio
 @item Adobe Filmstrip           @tab X @tab X
@@ -213,6 +245,7 @@ library:
     @tab Audio format used on the Nintendo Gamecube.
 @item AFC                       @tab   @tab X
     @tab Audio format used on the Nintendo Gamecube.
+@item APNG                      @tab X @tab X
 @item ASF                       @tab X @tab X
 @item AST                       @tab X @tab X
     @tab Audio format used on the Nintendo Wii.
@@ -233,6 +266,8 @@ library:
     @tab Used in Z and Z95 games.
 @item Brute Force & Ignorance   @tab   @tab X
     @tab Used in the game Flash Traffic: City of Angels.
+@item BFSTM                     @tab   @tab X
+    @tab Audio format used on the Nintendo WiiU (based on BRSTM).
 @item BRSTM                     @tab   @tab X
     @tab Audio format used on the Nintendo Wii.
 @item BWF                       @tab X @tab X
@@ -243,6 +278,10 @@ library:
     @tab Used in the game Cyberia from Interplay.
 @item Delphine Software International CIN @tab   @tab X
     @tab Multimedia format used by Delphine Software games.
+@item Digital Speech Standard (DSS) @tab   @tab X
+@item Canopus HQ                @tab   @tab X
+@item Canopus HQA               @tab   @tab X
+@item Canopus HQX               @tab   @tab X
 @item CD+G                      @tab   @tab X
     @tab Video format used by CD+G karaoke disks
 @item Phantom Cine              @tab   @tab X
@@ -259,6 +298,7 @@ library:
 @item Deluxe Paint Animation    @tab   @tab X
 @item DFA                       @tab   @tab X
     @tab This format is used in Chronomaster game
+@item DirectDraw Surface        @tab   @tab X
 @item DSD Stream File (DSF)     @tab   @tab X
 @item DV video                  @tab X @tab X
 @item DXA                       @tab   @tab X
@@ -454,6 +494,8 @@ library:
 @item Sony Wave64 (W64)         @tab X @tab X
 @item SoX native format         @tab X @tab X
 @item SUN AU format             @tab X @tab X
+@item SUP raw PGS subtitles     @tab   @tab X
+@item TDSC                      @tab   @tab X
 @item Text files                @tab   @tab X
 @item THP                       @tab   @tab X
     @tab Used on the Nintendo GameCube.
@@ -461,6 +503,7 @@ library:
     @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.
 @item True Audio                @tab   @tab X
 @item VC-1 test bitstream       @tab X @tab X
+@item Vidvox Hap                @tab X @tab X
 @item Vivo                      @tab   @tab X
 @item WAV                       @tab X @tab X
 @item WavPack                   @tab X @tab X
@@ -495,6 +538,7 @@ following image formats are supported:
 @item Alias PIX    @tab X @tab X
     @tab Alias/Wavefront PIX image format
 @item animated GIF @tab X @tab X
+@item APNG         @tab X @tab X
 @item BMP          @tab X @tab X
     @tab Microsoft BMP image
 @item BRender PIX  @tab   @tab X
@@ -646,15 +690,17 @@ following image formats are supported:
     @tab Sorenson H.263 used in Flash
 @item Forward Uncompressed   @tab     @tab  X
 @item Fraps                  @tab     @tab  X
+@item Go2Meeting             @tab     @tab  X
+    @tab fourcc: G2M2, G2M3
 @item Go2Webinar             @tab     @tab  X
     @tab fourcc: G2M4
 @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
 @item H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10  @tab  E  @tab  X
-    @tab encoding supported through external library libx264
+    @tab encoding supported through external library libx264 and OpenH264
 @item HEVC                   @tab  X  @tab  X
-    @tab encoding supported through the external library libx265
+    @tab encoding supported through external library libx265 and libkvazaar
 @item HNM version 4          @tab     @tab  X
 @item HuffYUV                @tab  X  @tab  X
 @item HuffYUV FFmpeg variant @tab  X  @tab  X
@@ -819,7 +865,7 @@ following image formats are supported:
 @item Name @tab Encoding @tab Decoding @tab Comments
 @item 8SVX exponential       @tab     @tab  X
 @item 8SVX fibonacci         @tab     @tab  X
-@item AAC+                   @tab  E  @tab  X
+@item AAC+                   @tab  E  @tab  IX
     @tab encoding supported through external library libaacplus
 @item AAC                    @tab  E  @tab  X
     @tab encoding supported through external library libfaac and libvo-aacenc
@@ -859,7 +905,7 @@ following image formats are supported:
 @item ADPCM MS IMA           @tab  X  @tab  X
 @item ADPCM Nintendo Gamecube AFC  @tab     @tab  X
 @item ADPCM Nintendo Gamecube DTK  @tab     @tab  X
-@item ADPCM Nintendo Gamecube THP  @tab     @tab  X
+@item ADPCM Nintendo THP  @tab     @tab  X
 @item ADPCM QT IMA           @tab  X  @tab  X
 @item ADPCM SEGA CRI ADX     @tab  X  @tab  X
     @tab Used in Sega Dreamcast games.
@@ -888,10 +934,12 @@ following image formats are supported:
     @tab decoding supported through external library libcelt
 @item Delphine Software International CIN audio  @tab     @tab  X
     @tab Codec used in Delphine Software International games.
+@item Digital Speech Standard - Standard Play mode (DSS SP) @tab     @tab  X
 @item Discworld II BMV Audio @tab     @tab  X
 @item COOK                   @tab     @tab  X
     @tab All versions except 5.1 are supported.
 @item DCA (DTS Coherent Acoustics)  @tab  X  @tab  X
+    @tab supported extensions: XCh, XLL (partially)
 @item DPCM id RoQ            @tab  X  @tab  X
     @tab Used in Quake III, Jedi Knight 2 and other computer games.
 @item DPCM Interplay         @tab     @tab  X
@@ -935,8 +983,8 @@ following image formats are supported:
 @item Musepack SV8           @tab     @tab  X
 @item Nellymoser Asao        @tab  X  @tab  X
 @item On2 AVC (Audio for Video Codec) @tab     @tab  X
-@item Opus                   @tab  E  @tab  E
-    @tab supported through external library libopus
+@item Opus                   @tab  E  @tab  X
+    @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 signed 8-bit planar  @tab  X  @tab  X
@@ -1030,6 +1078,7 @@ performance on systems without hardware floating point support).
 @item PJS (Phoenix)    @tab   @tab X @tab   @tab X
 @item RealText         @tab   @tab X @tab   @tab X
 @item SAMI             @tab   @tab X @tab   @tab X
+@item Spruce format (STL) @tab   @tab X @tab   @tab X
 @item SSA/ASS          @tab X @tab X @tab X @tab X
 @item SubRip (SRT)     @tab X @tab X @tab X @tab X
 @item SubViewer v1     @tab   @tab X @tab   @tab X
@@ -1055,6 +1104,7 @@ performance on systems without hardware floating point support).
 @item HLS          @tab X
 @item HTTP         @tab X
 @item HTTPS        @tab X
+@item Icecast      @tab X
 @item MMSH         @tab X
 @item MMST         @tab X
 @item pipe         @tab X

doc/git-howto.texi

@@ -1,9 +1,10 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
-@settitle Using git to develop FFmpeg
+@settitle Using Git to develop FFmpeg
 
 @titlepage
-@center @titlefont{Using git to develop FFmpeg}
+@center @titlefont{Using Git to develop FFmpeg}
 @end titlepage
 
 @top
@@ -12,9 +13,9 @@
 
 @chapter Introduction
 
-This document aims in giving some quick references on a set of useful git
+This document aims in giving some quick references on a set of useful Git
 commands. You should always use the extensive and detailed documentation
-provided directly by git:
+provided directly by Git:
 
 @example
 git --help
@@ -31,22 +32,21 @@ man git-<command>
 shows information about the subcommand <command>.
 
 Additional information could be found on the
-@url{http://gitref.org, Git Reference} website
+@url{http://gitref.org, Git Reference} website.
 
 For more information about the Git project, visit the
-
-@url{http://git-scm.com/, Git website}
+@url{http://git-scm.com/, Git website}.
 
 Consult these resources whenever you have problems, they are quite exhaustive.
 
 What follows now is a basic introduction to Git and some FFmpeg-specific
-guidelines to ease the contribution to the project
+guidelines to ease the contribution to the project.
 
 @chapter Basics Usage
 
-@section Get GIT
+@section Get Git
 
-You can get git from @url{http://git-scm.com/}
+You can get Git from @url{http://git-scm.com/}
 Most distribution and operating system provide a package for it.
 
 
@@ -74,6 +74,7 @@ git config --global core.autocrlf false
 @end example
 
 
+@anchor{Updating the source tree to the latest revision}
 @section Updating the source tree to the latest revision
 
 @example
@@ -106,7 +107,7 @@ git add [-A] <filename/dirname>
 git rm [-r] <filename/dirname>
 @end example
 
-GIT needs to get notified of all changes you make to your working
+Git needs to get notified of all changes you make to your working
 directory that makes files appear or disappear.
 Line moves across files are automatically tracked.
 
@@ -126,8 +127,8 @@ will show all local modifications in your working directory as unified diff.
 git log <filename(s)>
 @end example
 
-You may also use the graphical tools like gitview or gitk or the web
-interface available at http://source.ffmpeg.org/
+You may also use the graphical tools like @command{gitview} or @command{gitk}
+or the web interface available at @url{http://source.ffmpeg.org/}.
 
 @section Checking source tree status
 
@@ -148,6 +149,7 @@ git diff --check
 to double check your changes before committing them to avoid trouble later
 on. All experienced developers do this on each and every commit, no matter
 how small.
+
 Every one of them has been saved from looking like a fool by this many times.
 It's very easy for stray debug output or cosmetic modifications to slip in,
 please avoid problems through this extra level of scrutiny.
@@ -170,14 +172,14 @@ 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 and email address
 
 @example
 git config --global user.name "My Name"
 git config --global user.email my@@email.invalid
 @end example
 
-Use @var{--global} to set the global configuration for all your git checkouts.
+Use @option{--global} to set the global configuration for all your Git checkouts.
 
 Git will select the changes to the files for commit. Optionally you can use
 the interactive or the patch mode to select hunk by hunk what should be
@@ -208,7 +210,7 @@ include filenames in log messages, Git provides that information.
 
 Possibly make the commit message have a terse, descriptive first line, an
 empty line and then a full description. The first line will be used to name
-the patch by git format-patch.
+the patch by @command{git format-patch}.
 
 @section Preparing a patchset
 
@@ -324,12 +326,14 @@ faulty commit disappear from the history.
 @section Pushing changes to remote trees
 
 @example
-git push
+git push origin master --dry-run
 @end example
 
-Will push the changes to the default remote (@var{origin}).
+Will simulate a push of the local master branch to the default remote
+(@var{origin}). And list which branches and ranges or commits would have been
+pushed.
 Git will prevent you from pushing changes if the local and remote trees are
-out of sync. Refer to and to sync the local tree.
+out of sync. Refer to @ref{Updating the source tree to the latest revision}.
 
 @example
 git remote add <name> <url>
@@ -348,23 +352,24 @@ branches matching the local ones.
 
 @section Finding a specific svn revision
 
-Since version 1.7.1 git supports @var{:/foo} syntax for specifying commits
+Since version 1.7.1 Git supports @samp{:/foo} syntax for specifying commits
 based on a regular expression. see man gitrevisions
 
 @example
 git show :/'as revision 23456'
 @end example
 
-will show the svn changeset @var{r23456}. With older git versions searching in
+will show the svn changeset @samp{r23456}. With older Git versions searching in
 the @command{git log} output is the easiest option (especially if a pager with
 search capabilities is used).
+
 This commit can be checked out with
 
 @example
 git checkout -b svn_23456 :/'as revision 23456'
 @end example
 
-or for git < 1.7.1 with
+or for Git < 1.7.1 with
 
 @example
 git checkout -b svn_23456 $SHA1
@@ -373,7 +378,7 @@ git checkout -b svn_23456 $SHA1
 where @var{$SHA1} is the commit hash from the @command{git log} output.
 
 
-@chapter pre-push checklist
+@chapter Pre-push checklist
 
 Once you have a set of commits that you feel are ready for pushing,
 work through the following checklist to doublecheck everything is in
@@ -384,7 +389,7 @@ Apply your common sense, but if in doubt, err on the side of caution.
 First, make sure that the commits and branches you are going to push
 match what you want pushed and that nothing is missing, extraneous or
 wrong. You can see what will be pushed by running the git push command
-with --dry-run first. And then inspecting the commits listed with
+with @option{--dry-run} first. And then inspecting the commits listed with
 @command{git log -p 1234567..987654}. The @command{git status} command
 may help in finding local changes that have been forgotten to be added.
 
@@ -393,7 +398,7 @@ Next let the code pass through a full run of our testsuite.
 @itemize
 @item @command{make distclean}
 @item @command{/path/to/ffmpeg/configure}
-@item @command{make check}
+@item @command{make fate}
 @item if fate fails due to missing samples run @command{make fate-rsync} and retry
 @end itemize
 
@@ -411,5 +416,5 @@ recommended.
 
 @chapter Server Issues
 
-Contact the project admins @email{root@@ffmpeg.org} if you have technical
-problems with the GIT server.
+Contact the project admins at @email{root@@ffmpeg.org} if you have technical
+problems with the Git server.

doc/indevs.texi

@@ -1,7 +1,7 @@
 @chapter Input Devices
 @c man begin INPUT DEVICES
 
-Input devices are configured elements in FFmpeg which allow to access
+Input devices are configured elements in FFmpeg which enable accessing
 the data coming from a multimedia device attached to your system.
 
 When you configure your FFmpeg build, all the supported input devices
@@ -51,6 +51,18 @@ ffmpeg -f alsa -i hw:0 alsaout.wav
 For more information see:
 @url{http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html}
 
+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
 @section avfoundation
 
 AVFoundation input device.
@@ -58,38 +70,204 @@ AVFoundation input device.
 AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as on iOS.
 The older QTKit framework has been marked deprecated since OSX version 10.7.
 
-The filename passed as input is parsed to contain either a device name or index.
-The device index can also be given by using -video_device_index.
-A given device index will override any given device name.
-If the desired device consists of numbers only, use -video_device_index to identify it.
-The default device will be chosen if an empty string  or the device name "default" is given.
-The available devices can be enumerated by using -list_devices.
-The pixel format can be set using -pixel_format.
-Available formats:
- monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
+The input filename has to be given in the following syntax:
+@example
+-i "[[VIDEO]:[AUDIO]]"
+@end example
+The first entry selects the video input while the latter selects the audio input.
+The stream has to be specified by the device name or the device index as shown by the device list.
+Alternatively, the video and/or audio input device can be chosen by index using the
+@option{
+    -video_device_index <INDEX>
+}
+and/or
+@option{
+    -audio_device_index <INDEX>
+}
+, overriding any
+device name or index given in the input filename.
+
+All available devices can be enumerated by using @option{-list_devices true}, listing
+all device names and corresponding indices.
+
+There are two device name aliases:
+@table @code
+
+@item default
+Select the AVFoundation default device of the corresponding type.
+
+@item none
+Do not record the corresponding media type.
+This is equivalent to specifying an empty device name or index.
+
+@end table
+
+@subsection Options
+
+AVFoundation supports the following options:
+
+@table @option
+
+@item -list_devices <TRUE|FALSE>
+If set to true, a list of all available input devices is given showing all
+device names and indices.
+
+@item -video_device_index <INDEX>
+Specify the video device by its index. Overrides anything given in the input filename.
+
+@item -audio_device_index <INDEX>
+Specify the audio device by its index. Overrides anything given in the input filename.
+
+@item -pixel_format <FORMAT>
+Request the video device to use a specific pixel format.
+If the specified format is not supported, a list of available formats is given
+und the first one in this list is used instead. Available pixel formats are:
+@code{monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
  bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
- yuv420p, nv12, yuyv422, gray
+ yuv420p, nv12, yuyv422, gray}
 
+@item -framerate
+Set the grabbing frame rate. Default is @code{ntsc}, corresponding to a
+frame rate of @code{30000/1001}.
+
+@item -video_size
+Set the video frame size.
+
+@item -capture_cursor
+Capture the mouse pointer. Default is 0.
+
+@item -capture_mouse_clicks
+Capture the screen mouse clicks. Default is 0.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+Print the list of AVFoundation supported devices and exit:
 @example
-ffmpeg -f avfoundation -i "0" out.mpg
+$ ffmpeg -f avfoundation -list_devices true -i ""
 @end example
 
+@item
+Record video from video device 0 and audio from audio device 0 into out.avi:
 @example
-ffmpeg -f avfoundation -video_device_index 0 -i "" out.mpg
+$ ffmpeg -f avfoundation -i "0:0" out.avi
 @end example
 
+@item
+Record video from video device 2 and audio from audio device 1 into out.avi:
 @example
-ffmpeg -f avfoundation -pixel_format bgr0 -i "default" out.mpg
+$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
 @end example
 
+@item
+Record video from the system default video device using the pixel format bgr0 and do not record any audio into out.avi:
 @example
-ffmpeg -f avfoundation -list_devices true -i ""
+$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
 @end example
 
+@end itemize
+
 @section bktr
 
 BSD video input device.
 
+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate.
+
+@item video_size
+Set the video frame size. Default is @code{vga}.
+
+@item standard
+
+Available values are:
+@table @samp
+@item pal
+
+@item ntsc
+
+@item secam
+
+@item paln
+
+@item palm
+
+@item ntscj
+
+@end table
+
+@end table
+
+@section decklink
+
+The decklink input device provides capture capabilities for Blackmagic
+DeckLink devices.
+
+To enable this input device, you need the Blackmagic DeckLink SDK and you
+need to configure with the appropriate @code{--extra-cflags}
+and @code{--extra-ldflags}.
+On Windows, you need to run the IDL files through @command{widl}.
+
+DeckLink is very picky about the formats it supports. Pixel format is
+uyvy422 or v210, framerate and video size must be determined for your device with
+@command{-list_formats 1}. Audio sample rate is always 48 kHz and the number
+of channels can be 2, 8 or 16.
+
+@subsection Options
+
+@table @option
+
+@item list_devices
+If set to @option{true}, print a list of devices and exit.
+Defaults to @option{false}.
+
+@item list_formats
+If set to @option{true}, print a list of supported formats and exit.
+Defaults to @option{false}.
+
+@item bm_v210
+If set to @samp{1}, video is captured in 10 bit v210 instead
+of uyvy422. Not all Blackmagic devices support this option.
+
+@end table
+
+@subsection Examples
+
+@itemize
+
+@item
+List input devices:
+@example
+ffmpeg -f decklink -list_devices 1 -i dummy
+@end example
+
+@item
+List supported formats:
+@example
+ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
+@end example
+
+@item
+Capture video clip at 1080i50 (format 11):
+@example
+ffmpeg -f decklink -i 'Intensity Pro@@11' -acodec copy -vcodec copy output.avi
+@end example
+
+@item
+Capture video clip at 1080i50 10 bit:
+@example
+ffmpeg -bm_v210 1 -f decklink -i 'UltraStudio Mini Recorder@@11' -acodec copy -vcodec copy output.avi
+@end example
+
+@end itemize
+
 @section dshow
 
 Windows DirectShow input device.
@@ -107,7 +285,7 @@ The input name should be in the format:
 @end example
 
 where @var{TYPE} can be either @var{audio} or @var{video},
-and @var{NAME} is the device's name.
+and @var{NAME} is the device's name or alternative name..
 
 @subsection Options
 
@@ -140,11 +318,11 @@ If set to @option{true}, print a list of selected device's options
 and exit.
 
 @item video_device_number
-Set video device number for devices with same name (starts at 0,
+Set video device number for devices with the same name (starts at 0,
 defaults to 0).
 
 @item audio_device_number
-Set audio device number for devices with same name (starts at 0,
+Set audio device number for devices with the same name (starts at 0,
 defaults to 0).
 
 @item pixel_format
@@ -160,6 +338,85 @@ Setting this value too low can degrade performance.
 See also
 @url{http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx}
 
+@item video_pin_name
+Select video capture pin to use by name or alternative name.
+
+@item audio_pin_name
+Select audio capture pin to use by name or alternative name.
+
+@item crossbar_video_input_pin_number
+Select video input pin number for crossbar device. This will be
+routed to the crossbar device's Video Decoder output pin.
+Note that changing this value can affect future invocations
+(sets a new default) until system reboot occurs.
+
+@item crossbar_audio_input_pin_number
+Select audio input pin number for crossbar device. This will be
+routed to the crossbar device's Audio Decoder output pin.
+Note that changing this value can affect future invocations
+(sets a new default) until system reboot occurs.
+
+@item show_video_device_dialog
+If set to @option{true}, before capture starts, popup a display dialog
+to the end user, allowing them to change video filter properties
+and configurations manually.
+Note that for crossbar devices, adjusting values in this dialog
+may be needed at times to toggle between PAL (25 fps) and NTSC (29.97)
+input frame rates, sizes, interlacing, etc.  Changing these values can
+enable different scan rates/frame rates and avoiding green bars at
+the bottom, flickering scan lines, etc.
+Note that with some devices, changing these properties can also affect future
+invocations (sets new defaults) until system reboot occurs.
+
+@item show_audio_device_dialog
+If set to @option{true}, before capture starts, popup a display dialog
+to the end user, allowing them to change audio filter properties
+and configurations manually.
+
+@item show_video_crossbar_connection_dialog
+If set to @option{true}, before capture starts, popup a display
+dialog to the end user, allowing them to manually
+modify crossbar pin routings, when it opens a video device.
+
+@item show_audio_crossbar_connection_dialog
+If set to @option{true}, before capture starts, popup a display
+dialog to the end user, allowing them to manually
+modify crossbar pin routings, when it opens an audio device.
+
+@item show_analog_tv_tuner_dialog
+If set to @option{true}, before capture starts, popup a display
+dialog to the end user, allowing them to manually
+modify TV channels and frequencies.
+
+@item show_analog_tv_tuner_audio_dialog
+If set to @option{true}, before capture starts, popup a display
+dialog to the end user, allowing them to manually
+modify TV audio (like mono vs. stereo, Language A,B or C).
+
+@item audio_device_load
+Load an audio capture filter device from file instead of searching
+it by name. It may load additional parameters too, if the filter
+supports the serialization of its properties to.
+To use this an audio capture source has to be specified, but it can
+be anything even fake one.
+
+@item audio_device_save
+Save the currently used audio capture filter device and its
+parameters (if the filter supports it) to a file.
+If a file with the same name exists it will be overwritten.
+
+@item video_device_load
+Load a video capture filter device from file instead of searching
+it by name. It may load additional parameters too, if the filter
+supports the serialization of its properties to.
+To use this a video capture source has to be specified, but it can
+be anything even fake one.
+
+@item video_device_save
+Save the currently used video capture filter device and its
+parameters (if the filter supports it) to a file.
+If a file with the same name exists it will be overwritten.
+
 @end table
 
 @subsection Examples
@@ -196,12 +453,46 @@ Print the list of supported options in selected device and exit:
 $ ffmpeg -list_options true -f dshow -i video="Camera"
 @end example
 
+@item
+Specify pin names to capture by name or alternative name, specify alternative device name:
+@example
+$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#@{65e8773d-8f56-11d0-a3b9-00a0c9223196@}\@{ca465100-deb0-4d59-818f-8c477184adf6@}":audio="Microphone"
+@end example
+
+@item
+Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties at startup:
+@example
+$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
+     -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
+@end example
+
 @end itemize
 
 @section dv1394
 
 Linux DV 1394 input device.
 
+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate. Default is 25.
+
+@item standard
+
+Available values are:
+@table @samp
+@item pal
+
+@item ntsc
+
+@end table
+
+Default value is @code{ntsc}.
+
+@end table
+
 @section fbdev
 
 Linux framebuffer input device.
@@ -214,18 +505,27 @@ console. It is accessed through a file device node, usually
 For more detailed information read the file
 Documentation/fb/framebuffer.txt included in the Linux source tree.
 
+See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+
 To record from the framebuffer device @file{/dev/fb0} with
 @command{ffmpeg}:
 @example
-ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi
+ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
 @end example
 
 You can take a single screenshot image with the command:
 @example
-ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg
+ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
 @end example
 
-See also @url{http://linux-fbdev.sourceforge.net/}, and fbset(1).
+@subsection Options
+
+@table @option
+
+@item framerate
+Set the frame rate. Default is 25.
+
+@end table
 
 @section gdigrab
 
@@ -330,7 +630,7 @@ not work and result in undefined behavior.
 The values @option{auto}, @option{dv} and @option{hdv} are supported.
 
 @item dvbuffer
-Set maxiumum size of buffer for incoming data, in frames. For DV, this
+Set maximum size of buffer for incoming data, in frames. For DV, this
 is an exact value. For HDV, it is not frame exact, since HDV does
 not have a fixed frame size.
 
@@ -411,6 +711,15 @@ $ jack_connect metro:120_bpm ffmpeg:input_1
 For more information read:
 @url{http://jackaudio.org/}
 
+@subsection Options
+
+@table @option
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
 @section lavfi
 
 Libavfilter input virtual device.
@@ -435,6 +744,14 @@ generated by the device.
 The first unlabelled output is automatically assigned to the "out0"
 label, but all the others need to be specified explicitly.
 
+The suffix "+subcc" can be appended to the output label to create an extra
+stream with the closed captions packets attached to that output
+(experimental; only for EIA-608 / CEA-708 for now).
+The subcc streams are created after all the normal streams, in the order of
+the corresponding stream.
+For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the
+stream #43 is subcc for stream #7 and stream #44 is subcc for stream #19.
+
 If not specified defaults to the filename specified for the input
 device.
 
@@ -443,6 +760,9 @@ Set the filename of the filtergraph to be read and sent to the other
 filters. Syntax of the filtergraph is the same as the one specified by
 the option @var{graph}.
 
+@item dumpgraph
+Dump graph to stderr.
+
 @end table
 
 @subsection Examples
@@ -481,12 +801,63 @@ Read an audio stream and a video stream and play it back with
 ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
 @end example
 
+@item
+Dump decoded frames to images and closed captions to a file (experimental):
+@example
+ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
+@end example
+
 @end itemize
 
+@section libcdio
+
+Audio-CD input device based on libcdio.
+
+To enable this input device during configuration you need libcdio
+installed on your system. It requires the configure option
+@code{--enable-libcdio}.
+
+This device allows playing and grabbing from an Audio-CD.
+
+For example to copy with @command{ffmpeg} the entire Audio-CD in @file{/dev/sr0},
+you may run the command:
+@example
+ffmpeg -f libcdio -i /dev/sr0 cd.wav
+@end example
+
+@subsection Options
+@table @option
+@item speed
+Set drive reading speed. Default value is 0.
+
+The speed is specified CD-ROM speed units. The speed is set through
+the libcdio @code{cdio_cddap_speed_set} function. On many CD-ROM
+drives, specifying a value too large will result in using the fastest
+speed.
+
+@item paranoia_mode
+Set paranoia recovery mode flags. It accepts one of the following values:
+
+@table @samp
+@item disable
+@item verify
+@item overlap
+@item neverskip
+@item full
+@end table
+
+Default value is @samp{disable}.
+
+For more information about the available recovery modes, consult the
+paranoia project documentation.
+@end table
+
 @section libdc1394
 
 IIDC1394 input device, based on libdc1394 and libraw1394.
 
+Requires the configure option @code{--enable-libdc1394}.
+
 @section openal
 
 The OpenAL input device provides audio capture on all systems with a
@@ -593,6 +964,19 @@ ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
 For more information about OSS see:
 @url{http://manuals.opensound.com/usersguide/dsp.html}
 
+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
+
 @section pulse
 
 PulseAudio input device.
@@ -633,6 +1017,10 @@ Specify the number of bytes per frame, by default it is set to 1024.
 @item fragment_size
 Specify the minimal buffering fragment in PulseAudio, it will affect the
 audio latency. By default it is unset.
+
+@item wallclock
+Set the initial PTS using the current time. Default is 1.
+
 @end table
 
 @subsection Examples
@@ -668,6 +1056,22 @@ ffmpeg -f qtkit -i "default" out.mpg
 ffmpeg -f qtkit -list_devices true -i ""
 @end example
 
+@subsection Options
+
+@table @option
+
+@item frame_rate
+Set frame rate. Default is 30.
+
+@item list_devices
+If set to @code{true}, print a list of devices and exit. Default is
+@code{false}.
+
+@item video_device_index
+Select the video device by index for devices with the same name (starts at 0).
+
+@end table
+
 @section sndio
 
 sndio input device.
@@ -685,6 +1089,18 @@ command:
 ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
 @end example
 
+@subsection Options
+
+@table @option
+
+@item sample_rate
+Set the sample rate in Hz. Default is 48000.
+
+@item channels
+Set the number of channels. Default is 2.
+
+@end table
+
 @section video4linux2, v4l2
 
 Video4Linux2 input video device.
@@ -717,6 +1133,12 @@ conversion into the real time clock.
 Some usage examples of the video4linux2 device with @command{ffmpeg}
 and @command{ffplay}:
 @itemize
+@item
+List supported formats for a video4linux2 device:
+@example
+ffplay -f video4linux2 -list_formats all /dev/video0
+@end example
+
 @item
 Grab and show the input of a video4linux2 device:
 @example
@@ -801,6 +1223,10 @@ Force conversion from monotonic to absolute timestamps.
 @end table
 
 Default value is @code{default}.
+
+@item use_libv4l2
+Use libv4l2 (v4l-utils) conversion functions. Default is 0.
+
 @end table
 
 @section vfwcap
@@ -811,10 +1237,30 @@ The filename passed as input is the capture driver number, ranging from
 0 to 9. You may use "list" as filename to print a list of drivers. Any
 other filename will be interpreted as device number 0.
 
+@subsection Options
+
+@table @option
+
+@item video_size
+Set the video frame size.
+
+@item framerate
+Set the grabbing frame rate. Default value is @code{ntsc},
+corresponding to a frame rate of @code{30000/1001}.
+
+@end table
+
 @section x11grab
 
 X11 video input device.
 
+To enable this input device during configuration you need libxcb
+installed on your system. It will be automatically detected during
+configuration.
+
+Alternatively, the configure option @option{--enable-x11grab} exists
+for legacy Xlib users.
+
 This device allows one to capture a region of an X11 display.
 
 The filename passed as input has the syntax:
@@ -831,10 +1277,12 @@ omitted, and defaults to "localhost". The environment variable
 area with respect to the top-left border of the X11 screen. They
 default to 0.
 
-Check the X11 documentation (e.g. man X) for more detailed information.
+Check the X11 documentation (e.g. @command{man X}) for more detailed
+information.
 
-Use the @command{dpyinfo} program for getting basic information about the
-properties of your X11 display (e.g. grep for "name" or "dimensions").
+Use the @command{xdpyinfo} program for getting basic information about
+the properties of your X11 display (e.g. grep for "name" or
+"dimensions").
 
 For example to grab from @file{:0.0} using @command{ffmpeg}:
 @example
@@ -883,6 +1331,10 @@ If @var{show_region} is specified with @code{1}, then the grabbing
 region will be indicated on screen. With this option, it is easy to
 know what is being grabbed if only a portion of the screen is grabbed.
 
+@item region_border
+Set the region border thickness if @option{-show_region 1} is used.
+Range is 1 to 128 and default is 3 (XCB-based x11grab only).
+
 For example:
 @example
 ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
@@ -895,6 +1347,22 @@ ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_siz
 
 @item video_size
 Set the video frame size. Default value is @code{vga}.
+
+@item use_shm
+Use the MIT-SHM extension for shared memory. Default value is @code{1}.
+It may be necessary to disable it for remote displays (legacy x11grab
+only).
 @end table
 
+@subsection @var{grab_x} @var{grab_y} AVOption
+
+The syntax is:
+@example
+-grab_x @var{x_offset} -grab_y @var{y_offset}
+@end example
+
+Set the grabbing region coordinates. They are expressed as offset from the top left
+corner of the X11 window. The default value is 0.
+
+
 @c man end INPUT DEVICES

doc/issue_tracker.txt

@@ -22,7 +22,7 @@ a mail for every change to every issue.
 (the above does all work already after light testing)
 
 The subscription URL for the ffmpeg-trac list is:
-http(s)://ffmpeg.org/mailman/listinfo/ffmpeg-trac
+http(s)://lists.ffmpeg.org/mailman/listinfo/ffmpeg-trac
 The URL of the webinterface of the tracker is:
 http(s)://trac.ffmpeg.org
 

doc/libavcodec.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libavcodec Documentation
 @titlepage

doc/libavdevice.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libavdevice Documentation
 @titlepage

doc/libavfilter.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libavfilter Documentation
 @titlepage

doc/libavformat.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libavformat Documentation
 @titlepage

doc/libavutil.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libavutil Documentation
 @titlepage

doc/libswresample.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libswresample Documentation
 @titlepage

doc/libswscale.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Libswscale Documentation
 @titlepage

doc/metadata.texi

@@ -12,10 +12,10 @@ A file consists of a header and a number of metadata tags divided into sections,
 each on its own line.
 
 @item
-The header is a ';FFMETADATA' string, followed by a version number (now 1).
+The header is a @samp{;FFMETADATA} string, followed by a version number (now 1).
 
 @item
-Metadata tags are of the form 'key=value'
+Metadata tags are of the form @samp{key=value}
 
 @item
 Immediately after header follows global metadata
@@ -26,26 +26,30 @@ metadata.
 
 @item
 A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
-brackets ('[', ']') and ends with next section or end of file.
+brackets (@samp{[}, @samp{]}) and ends with next section or end of file.
 
 @item
 At the beginning of a chapter section there may be an optional timebase to be
-used for start/end values. It must be in form 'TIMEBASE=num/den', where num and
-den are integers. If the timebase is missing then start/end times are assumed to
+used for start/end values. It must be in form
+@samp{TIMEBASE=@var{num}/@var{den}}, where @var{num} and @var{den} are
+integers. If the timebase is missing then start/end times are assumed to
 be in milliseconds.
+
 Next a chapter section must contain chapter start and end times in form
-'START=num', 'END=num', where num is a positive integer.
+@samp{START=@var{num}}, @samp{END=@var{num}}, where @var{num} is a positive
+integer.
 
 @item
-Empty lines and lines starting with ';' or '#' are ignored.
+Empty lines and lines starting with @samp{;} or @samp{#} are ignored.
 
 @item
-Metadata keys or values containing special characters ('=', ';', '#', '\' and a
-newline) must be escaped with a backslash '\'.
+Metadata keys or values containing special characters (@samp{=}, @samp{;},
+@samp{#}, @samp{\} and a newline) must be escaped with a backslash @samp{\}.
 
 @item
-Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of
-the tag (in the example above key is 'foo ', value is ' bar').
+Note that whitespace in metadata (e.g. @samp{foo = bar}) is considered to be
+a part of the tag (in the example above key is @samp{foo }, value is
+@samp{ bar}).
 @end enumerate
 
 A ffmetadata file might look like this:

doc/mips.txt

@@ -47,12 +47,16 @@ Files that have MIPS copyright notice in them:
 * libavutil/mips/
       float_dsp_mips.c
       libm_mips.h
+      softfloat_tables.h
 * libavcodec/
       fft_fixed_32.c
       fft_init_table.c
       fft_table.h
       mdct_fixed_32.c
 * libavcodec/mips/
+      aacdec_fixed.c
+      aacsbr_fixed.c
+      aacsbr_template.c
       aaccoder_mips.c
       aacpsy_mips.h
       ac3dsp_mips.c

doc/multithreading.txt

@@ -54,7 +54,7 @@ thread.
 If the codec allocates writable tables in its init(), add an init_thread_copy()
 which re-allocates them for other threads.
 
-Add CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
+Add AV_CODEC_CAP_FRAME_THREADS to the codec capabilities. There will be very little
 speed gain at this point but it should work.
 
 If there are inter-frame dependencies, so the codec calls

doc/muxers.texi

@@ -194,15 +194,19 @@ can not be smaller than one centi second.
 Apple HTTP Live Streaming muxer that segments MPEG-TS according to
 the HTTP Live Streaming (HLS) specification.
 
-It creates a playlist file and numbered segment files. The output
-filename specifies the playlist filename; the segment filenames
-receive the same basename as the playlist, a sequential number and
-a .ts extension.
+It creates a playlist file, and one or more segment files. The output filename
+specifies the playlist filename.
+
+By default, the muxer creates a file for each segment produced. These files
+have the same name as the playlist, followed by a sequential number and a
+.ts extension.
 
 For example, to convert an input file with @command{ffmpeg}:
 @example
 ffmpeg -i in.nut out.m3u8
 @end example
+This example will produce the playlist, @file{out.m3u8}, and segment files:
+@file{out0.ts}, @file{out1.ts}, @file{out2.ts}, etc.
 
 See also the @ref{segment} muxer, which provides a more generic and
 flexible implementation of a segmenter, and can be used to perform HLS
@@ -220,6 +224,11 @@ Set the segment length in seconds. Default value is 2.
 Set the maximum number of playlist entries. If set to 0 the list file
 will contain all the segments. Default value is 5.
 
+@item hls_ts_options @var{options_list}
+Set output format options using a :-separated list of key=value
+parameters. Values containing @code{:} special characters must be
+escaped.
+
 @item hls_wrap @var{wrap}
 Set the number after which the segment filename number (the number
 specified in each segment file) wraps. If set to 0 the number will be
@@ -233,6 +242,9 @@ to @var{wrap}.
 Start the playlist sequence number from @var{number}. Default value is
 0.
 
+@item hls_allow_cache @var{allowcache}
+Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.
+
 @item hls_base_url @var{baseurl}
 Append @var{baseurl} to every entry in the playlist.
 Useful to generate playlists with absolute paths.
@@ -241,6 +253,86 @@ Note that the playlist sequence number must be unique for each segment
 and it is not to be confused with the segment filename sequence number
 which can be cyclic, for example if the @option{wrap} option is
 specified.
+
+@item hls_segment_filename @var{filename}
+Set the segment filename. Unless hls_flags single_file is set @var{filename}
+is used as a string format with the segment number:
+@example
+ffmpeg in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
+@end example
+This example will produce the playlist, @file{out.m3u8}, and segment files:
+@file{file000.ts}, @file{file001.ts}, @file{file002.ts}, etc.
+
+@item hls_key_info_file @var{key_info_file}
+Use the information in @var{key_info_file} for segment encryption. The first
+line of @var{key_info_file} specifies the key URI written to the playlist. The
+key URL is used to access the encryption key during playback. The second line
+specifies the path to the key file used to obtain the key during the encryption
+process. The key file is read as a single packed array of 16 octets in binary
+format. The optional third line specifies the initialization vector (IV) as a
+hexadecimal string to be used instead of the segment sequence number (default)
+for encryption. Changes to @var{key_info_file} will result in segment
+encryption with the new key/IV and an entry in the playlist for the new key
+URI/IV.
+
+Key info file format:
+@example
+@var{key URI}
+@var{key file path}
+@var{IV} (optional)
+@end example
+
+Example key URIs:
+@example
+http://server/file.key
+/path/to/file.key
+file.key
+@end example
+
+Example key file paths:
+@example
+file.key
+/path/to/file.key
+@end example
+
+Example IV:
+@example
+0123456789ABCDEF0123456789ABCDEF
+@end example
+
+Key info file example:
+@example
+http://server/file.key
+/path/to/file.key
+0123456789ABCDEF0123456789ABCDEF
+@end example
+
+Example shell script:
+@example
+#!/bin/sh
+BASE_URL=$@{1:-'.'@}
+openssl rand 16 > file.key
+echo $BASE_URL/file.key > file.keyinfo
+echo file.key >> file.keyinfo
+echo $(openssl rand -hex 16) >> file.keyinfo
+ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
+  -hls_key_info_file file.keyinfo out.m3u8
+@end example
+
+@item hls_flags single_file
+If this flag is set, the muxer will store all segments in a single MPEG-TS
+file, and will use byte ranges in the playlist. HLS playlists generated with
+this way will have the version number 4.
+For example:
+@example
+ffmpeg -i in.nut -hls_flags single_file out.m3u8
+@end example
+Will produce the playlist, @file{out.m3u8}, and a single segment file,
+@file{out.ts}.
+
+@item hls_flags delete_segments
+Segment files removed from the playlist are deleted after a period of time
+equal to the duration of the segment plus the duration of the playlist.
 @end table
 
 @anchor{ico}
@@ -345,8 +437,7 @@ ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
 
 @table @option
 @item start_number
-Start the sequence from the specified number. Default value is 1. Must
-be a non-negative number.
+Start the sequence from the specified number. Default value is 0.
 
 @item update
 If set to 1, the filename will always be interpreted as just a
@@ -536,7 +627,6 @@ a short portion of the file. With this option set, there is no initial
 mdat atom, and the moov atom only describes the tracks but has
 a zero duration.
 
-Files written with this option set do not work in QuickTime.
 This option is implicitly set when writing ismv (Smooth Streaming) files.
 @item -movflags separate_moof
 Write a separate moof (movie fragment) atom for each track. Normally,
@@ -551,6 +641,22 @@ This operation can take a while, and will not work in various situations such
 as fragmented output, thus it is not enabled by default.
 @item -movflags rtphint
 Add RTP hinting tracks to the output file.
+@item -movflags disable_chpl
+Disable Nero chapter markers (chpl atom).  Normally, both Nero chapters
+and a QuickTime chapter track are written to the file. With this option
+set, only the QuickTime chapter track will be written. Nero chapters can
+cause failures when the file is reprocessed with certain tagging programs, like
+mp3Tag 2.61a and iTunes 11.3, most likely other versions are affected as well.
+@item -movflags omit_tfhd_offset
+Do not write any absolute base_data_offset in tfhd atoms. This avoids
+tying fragments to absolute byte positions in the file/streams.
+@item -movflags default_base_moof
+Similarly to the omit_tfhd_offset, this flag avoids writing the
+absolute base_data_offset field in tfhd atoms, but does so by using
+the new default-base-is-moof flag instead. This flag is new from
+14496-12:2012. This may make the fragments easier to parse in certain
+circumstances (avoiding basing track fragment location calculations
+on the implicit end of the previous track fragment).
 @end table
 
 @subsection Example
@@ -561,31 +667,47 @@ point on IIS with this muxer. Example:
 ffmpeg -re @var{<normal input/transcoding options>} -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)
 @end example
 
+@subsection Audible AAX
+
+Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation secret.
+@example
+ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
+@end example
+
 @section mp3
 
-The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
-optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
-@code{id3v2_version} option controls which one is used. Setting
-@code{id3v2_version} to 0 will disable the ID3v2 header completely. The legacy
-ID3v1 tag is not written by default, but may be enabled with the
-@code{write_id3v1} option.
+The MP3 muxer writes a raw MP3 stream with the following optional features:
+@itemize @bullet
+@item
+An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and
+2.4 are supported, the @code{id3v2_version} private option controls which one is
+used (3 or 4). Setting @code{id3v2_version} to 0 disables the ID3v2 header
+completely.
 
-The muxer may also write a Xing frame at the beginning, which contains the
-number of frames in the file. It is useful for computing duration of VBR files.
-The Xing frame is written if the output stream is seekable and if the
-@code{write_xing} option is set to 1 (the default).
-
-The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures
-are supplied to the muxer in form of a video stream with a single packet. There
-can be any number of those streams, each will correspond to a single APIC frame.
-The stream metadata tags @var{title} and @var{comment} map to APIC
-@var{description} and @var{picture type} respectively. See
+The muxer supports writing attached pictures (APIC frames) to the ID3v2 header.
+The pictures are supplied to the muxer in form of a video stream with a single
+packet. There can be any number of those streams, each will correspond to a
+single APIC frame.  The stream metadata tags @var{title} and @var{comment} map
+to APIC @var{description} and @var{picture type} respectively. See
 @url{http://id3.org/id3v2.4.0-frames} for allowed picture types.
 
 Note that the APIC frames must be written at the beginning, so the muxer will
 buffer the audio frames until it gets all the pictures. It is therefore advised
 to provide the pictures as soon as possible to avoid excessive buffering.
 
+@item
+A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by
+default, but will be written only if the output is seekable. The
+@code{write_xing} private option can be used to disable it.  The frame contains
+various information that may be useful to the decoder, like the audio duration
+or encoder delay.
+
+@item
+A legacy ID3v1 tag at the end of the file (disabled by default). It may be
+enabled with the @code{write_id3v1} private option, but as its capabilities are
+very limited, its usage is not recommended.
+@end itemize
+
 Examples:
 
 Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
@@ -630,6 +752,9 @@ Set the transport_stream_id (default 0x0001). This identifies a
 transponder in DVB.
 @item -mpegts_service_id @var{number}
 Set the service_id (default 0x0001) also known as program in DVB.
+@item -mpegts_service_type @var{number}
+Set the program service_type (default @var{digital_tv}), see below
+a list of pre defined values.
 @item -mpegts_pmt_start_pid @var{number}
 Set the first PID for PMT (default 0x1000, max 0x1f00).
 @item -mpegts_start_pid @var{number}
@@ -641,6 +766,10 @@ Set a constant muxrate (default VBR).
 @item -pcr_period @var{numer}
 Override the default PCR retransmission time (default 20ms), ignored
 if variable muxrate is selected.
+@item pat_period @var{number}
+Maximal time in seconds between PAT/PMT tables.
+@item sdt_period @var{number}
+Maximal time in seconds between SDT tables.
 @item -pes_payload_size @var{number}
 Set minimum PES packet payload in bytes.
 @item -mpegts_flags @var{flags}
@@ -664,6 +793,27 @@ ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
 @end example
 @end table
 
+Option mpegts_service_type accepts the following values:
+
+@table @option
+@item hex_value
+Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.
+@item digital_tv
+Digital TV service.
+@item digital_radio
+Digital Radio service.
+@item teletext
+Teletext service.
+@item advanced_codec_digital_radio
+Advanced Codec Digital Radio service.
+@item mpeg2_digital_hdtv
+MPEG2 Digital HDTV service.
+@item advanced_codec_digital_sdtv
+Advanced Codec Digital SDTV service.
+@item advanced_codec_digital_hdtv
+Advanced Codec Digital HDTV service.
+@end table
+
 Option mpegts_flags may take a set of such flags:
 
 @table @option
@@ -671,6 +821,8 @@ Option mpegts_flags may take a set of such flags:
 Reemit PAT/PMT before writing the next packet.
 @item latm
 Use LATM packetization for AAC.
+@item pat_pmt_at_frames
+Reemit PAT and PMT at each video frame.
 @end table
 
 @subsection Example
@@ -687,6 +839,21 @@ ffmpeg -i file.mpg -c copy \
      -y out.ts
 @end example
 
+@section mxf, mxf_d10
+
+MXF muxer.
+
+@subsection Options
+
+The muxer options are:
+
+@table @option
+@item store_user_comments @var{bool}
+Set if user comments should be stored if available or never.
+IRT D-10 does not allow user comments. The default is thus to write them for
+mxf but not for mxf_d10
+@end table
+
 @section null
 
 Null muxer.
@@ -721,7 +888,7 @@ Change the syncpoint usage in nut:
     sensitive and seeking is not possible. Also in general the overhead from
     syncpoints is negligible. Note, -@code{write_index} 0 can be used to disable
     all growing data tables, allowing to mux endless streams with limited memory
-    and wihout these disadvantages.
+    and without these disadvantages.
 @item @var{timestamped} extend the syncpoint with a wallclock field.
 @end table
 The @var{none} and @var{timestamped} flags are experimental.
@@ -746,6 +913,11 @@ is 1 second. A value of 0 will fill all segments, making pages as large as
 possible. A value of 1 will effectively use 1 packet-per-page in most
 situations, giving a small seek granularity at the cost of additional container
 overhead.
+@item -serial_offset @var{value}
+Serial value from which to set the streams serial number.
+Setting it to different and sufficiently large values ensures that the produced
+ogg files can be safely chained.
+
 @end table
 
 @anchor{segment}
@@ -754,8 +926,9 @@ overhead.
 Basic stream segmenter.
 
 This muxer outputs streams to a number of separate files of nearly
-fixed duration. Output filename pattern can be set in a fashion similar to
-@ref{image2}.
+fixed duration. Output filename pattern can be set in a fashion
+similar to @ref{image2}, or by using a @code{strftime} template if
+the @option{strftime} option is enabled.
 
 @code{stream_segment} is a variant of the muxer used to write to
 streaming output formats, i.e. which do not require global headers,
@@ -798,6 +971,11 @@ reference stream. The default value is @code{auto}.
 Override the inner container format, by default it is guessed by the filename
 extension.
 
+@item segment_format_options @var{options_list}
+Set output format options using a :-separated list of key=value
+parameters. Values containing the @code{:} special character must be
+escaped.
+
 @item segment_list @var{name}
 Generate also a listfile named @var{name}. If not specified no
 listfile is generated.
@@ -814,15 +992,8 @@ Allow caching (only affects M3U8 list files).
 Allow live-friendly file generation.
 @end table
 
-@item segment_list_type @var{type}
-Select the listing format.
-@table @option
-@item @var{flat} use a simple flat list of entries.
-@item @var{hls} use a m3u8-like structure.
-@end table
-
 @item segment_list_size @var{size}
-Update the list file so that it contains at most the last @var{size}
+Update the list file so that it contains at most @var{size}
 segments. If 0 the list file will contain all the segments. Default
 value is 0.
 
@@ -830,6 +1001,9 @@ value is 0.
 Prepend @var{prefix} to each entry. Useful to generate absolute paths.
 By default no prefix is applied.
 
+@item segment_list_type @var{type}
+Select the listing format.
+
 The following values are recognized:
 @table @samp
 @item flat
@@ -930,6 +1104,18 @@ Wrap around segment index once it reaches @var{limit}.
 @item segment_start_number @var{number}
 Set the sequence number of the first segment. Defaults to @code{0}.
 
+@item strftime @var{1|0}
+Use the @code{strftime} function to define the name of the new
+segments to write. If this is selected, the output segment name must
+contain a @code{strftime} function template. Default value is
+@code{0}.
+
+@item break_non_keyframes @var{1|0}
+If enabled, allow segments to start on frames other than keyframes. This
+improves behavior on some players when the time between keyframes is
+inconsistent, but may make things worse on others, and can cause some oddities
+during seeking. Defaults to @code{0}.
+
 @item reset_timestamps @var{1|0}
 Reset timestamps at the begin of each segment, so that each segment
 will start with near-zero timestamps. It is meant to ease the playback
@@ -945,7 +1131,7 @@ argument must be a time duration specification, and defaults to 0.
 
 @itemize
 @item
-To remux the content of file @file{in.mkv} to a list of segments
+Remux the content of file @file{in.mkv} to a list of segments
 @file{out-000.nut}, @file{out-001.nut}, etc., and write the list of
 generated segments to @file{out.list}:
 @example
@@ -953,14 +1139,20 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nu
 @end example
 
 @item
-As the example above, but segment the input file according to the split
-points specified by the @var{segment_times} option:
+Segment input and set output format options for the output segments:
+@example
+ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
+@end example
+
+@item
+Segment the input file according to the split points specified by the
+@var{segment_times} option:
 @example
 ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut
 @end example
 
 @item
-As the example above, but use the @command{ffmpeg} @option{force_key_frames}
+Use the @command{ffmpeg} @option{force_key_frames}
 option to force key frames in the input at the specified location, together
 with the segment option @option{segment_time_delta} to account for
 possible roundings operated when setting key frame times.
@@ -979,7 +1171,7 @@ ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_fr
 @end example
 
 @item
-To convert the @file{in.mkv} to TS segments using the @code{libx264}
+Convert the @file{in.mkv} to TS segments using the @code{libx264}
 and @code{libfaac} encoders:
 @example
 ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts
@@ -1104,7 +1296,17 @@ is the @option{global_header} flag.
 
 WebM DASH Manifest muxer.
 
-This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML.
+This muxer implements the WebM DASH Manifest specification to generate the DASH
+manifest XML. It also supports manifest generation for DASH live streams.
+
+For more information see:
+
+@itemize @bullet
+@item
+WebM DASH Specification: @url{https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification}
+@item
+ISO DASH Specification: @url{http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip}
+@end itemize
 
 @subsection Options
 
@@ -1115,6 +1317,32 @@ This muxer supports the following options:
 This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
 unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
 audio and video streams. Any number of adaptation sets can be added using this option.
+
+@item live
+Set this to 1 to create a live stream DASH Manifest. Default: 0.
+
+@item chunk_start_index
+Start index of the first chunk. This will go in the @samp{startNumber} attribute
+of the @samp{SegmentTemplate} element in the manifest. Default: 0.
+
+@item chunk_duration_ms
+Duration of each chunk in milliseconds. This will go in the @samp{duration}
+attribute of the @samp{SegmentTemplate} element in the manifest. Default: 1000.
+
+@item utc_timing_url
+URL of the page that will return the UTC timestamp in ISO format. This will go
+in the @samp{value} attribute of the @samp{UTCTiming} element in the manifest.
+Default: None.
+
+@item time_shift_buffer_depth
+Smallest time (in seconds) shifting buffer for which any Representation is
+guaranteed to be available. This will go in the @samp{timeShiftBufferDepth}
+attribute of the @samp{MPD} element. Default: 60.
+
+@item minimum_update_period
+Minimum update period (in seconds) of the manifest. This will go in the
+@samp{minimumUpdatePeriod} attribute of the @samp{MPD} element. Default: 0.
+
 @end table
 
 @subsection Example
@@ -1130,4 +1358,47 @@ ffmpeg -f webm_dash_manifest -i video1.webm \
        manifest.xml
 @end example
 
+@section webm_chunk
+
+WebM Live Chunk Muxer.
+
+This muxer writes out WebM headers and chunks as separate files which can be
+consumed by clients that support WebM Live streams via DASH.
+
+@subsection Options
+
+This muxer supports the following options:
+
+@table @option
+@item chunk_start_index
+Index of the first chunk (defaults to 0).
+
+@item header
+Filename of the header where the initialization data will be written.
+
+@item audio_chunk_duration
+Duration of each audio chunk in milliseconds (defaults to 5000).
+@end table
+
+@subsection Example
+@example
+ffmpeg -f v4l2 -i /dev/video0 \
+       -f alsa -i hw:0 \
+       -map 0:0 \
+       -c:v libvpx-vp9 \
+       -s 640x360 -keyint_min 30 -g 30 \
+       -f webm_chunk \
+       -header webm_live_video_360.hdr \
+       -chunk_start_index 1 \
+       webm_live_video_360_%d.chk \
+       -map 1:0 \
+       -c:a libvorbis \
+       -b:a 128k \
+       -f webm_chunk \
+       -header webm_live_audio_128.hdr \
+       -chunk_start_index 1 \
+       -audio_chunk_duration 1000 \
+       webm_live_audio_128_%d.chk
+@end example
+
 @c man end MUXERS

doc/nut.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle NUT
 

doc/optimization.txt

@@ -191,6 +191,11 @@ __asm__() block.
 Use external asm (nasm/yasm) or inline asm (__asm__()), do not use intrinsics.
 The latter requires a good optimizing compiler which gcc is not.
 
+When debugging a x86 external asm compilation issue, if lost in the macro
+expansions, add DBG=1 to your make command-line: the input file will be
+preprocessed, stripped of the debug/empty lines, then compiled, showing the
+actual lines causing issues.
+
 Inline asm vs. external asm
 ---------------------------
 Both inline asm (__asm__("..") in a .c file, handled by a compiler such as gcc)

doc/platform.texi

@@ -1,4 +1,5 @@
 \input texinfo @c -*- texinfo -*-
+@documentencoding UTF-8
 
 @settitle Platform Specific Information
 @titlepage
@@ -35,6 +36,9 @@ to your project LDFLAGS:
 -Wl,-Bsymbolic
 @end example
 
+If your target platform requires position independent binaries, you should
+pass the correct linking flag (e.g. @code{-pie}) to @code{--extra-ldexeflags}.
+
 @section BSD
 
 BSD make will not build FFmpeg, you need to install and use GNU Make
@@ -93,9 +97,9 @@ the FFmpeg Windows Help Forum at @url{http://ffmpeg.zeranoe.com/forum/}.
 
 @section Native Windows compilation using MinGW or MinGW-w64
 
-FFmpeg can be built to run natively on Windows using the MinGW or MinGW-w64
-toolchains. Install the latest versions of MSYS and MinGW or MinGW-w64 from
-@url{http://www.mingw.org/} or @url{http://mingw-w64.sourceforge.net/}.
+FFmpeg can be built to run natively on Windows using the MinGW-w64
+toolchain. Install the latest versions of MSYS2 and MinGW-w64 from
+@url{http://msys2.github.io/} and/or @url{http://mingw-w64.sourceforge.net/}.
 You can find detailed installation instructions in the download section and
 the FAQ.
 
@@ -103,7 +107,7 @@ Notes:
 
 @itemize
 
-@item Building natively using MSYS can be sped up by disabling implicit rules
+@item Building natively using MSYS2 can be sped up by disabling implicit rules
 in the Makefile by calling @code{make -r} instead of plain @code{make}. This
 speed up is close to non-existent for normal one-off builds and is only
 noticeable when running make for a second time (for example during
@@ -130,13 +134,12 @@ You will need the following prerequisites:
 (if using MSVC 2012 or earlier)
 @item @uref{http://code.google.com/p/msinttypes/, msinttypes}
 (if using MSVC 2012 or earlier)
-@item @uref{http://www.mingw.org/, MSYS}
+@item @uref{http://msys2.github.io/, MSYS2}
 @item @uref{http://yasm.tortall.net/, YASM}
-@item @uref{http://gnuwin32.sourceforge.net/packages/bc.htm, bc for Windows} if
-you want to run @uref{fate.html, FATE}.
+(Also available via MSYS2's package manager.)
 @end itemize
 
-To set up a proper environment in MSYS, you need to run @code{msys.bat} from
+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
@@ -172,12 +175,6 @@ Notes:
 
 @itemize
 
-@item It is possible that coreutils' @code{link.exe} conflicts with MSVC's linker.
-You can find out by running @code{which link} to see which @code{link.exe} you
-are using. If it is located at @code{/bin/link.exe}, then you have the wrong one
-in your @code{PATH}. Either move or remove that copy, or make sure MSVC's
-@code{link.exe} takes precedence in your @code{PATH} over coreutils'.
-
 @item If you wish to build with zlib support, you will have to grab a compatible
 zlib binary from somewhere, with an MSVC import lib, or if you wish to link
 statically, you can follow the instructions below to build a compatible
@@ -275,12 +272,12 @@ llrint() in its C library.
 Install your Cygwin with all the "Base" packages, plus the
 following "Devel" ones:
 @example
-binutils, gcc4-core, make, git, mingw-runtime, texi2html
+binutils, gcc4-core, make, git, mingw-runtime, texinfo
 @end example
 
 In order to run FATE you will also need the following "Utils" packages:
 @example
-bc, diffutils
+diffutils
 @end example
 
 If you want to build FFmpeg with additional libraries, download Cygwin

doc/print_options.c

@@ -26,6 +26,10 @@
 #include <string.h>
 #include <float.h>
 
+// print_options is build for the host, os_support.h isn't needed and is setup
+// for the target. without this build breaks on mingw
+#define AVFORMAT_OS_SUPPORT_H
+
 #include "libavformat/avformat.h"
 #include "libavformat/options_table.h"
 #include "libavcodec/avcodec.h"

doc/protocols.texi

@@ -19,6 +19,18 @@ supported protocols.
 
 A description of the currently available protocols follows.
 
+@section async
+
+Asynchronous data filling wrapper for input stream.
+
+Fill data in a background thread, to decouple I/O operation from demux thread.
+
+@example
+async:@var{URL}
+async:http://host/resource
+async:cache:http://host/resource
+@end example
+
 @section bluray
 
 Read BluRay playlist.
@@ -63,7 +75,7 @@ cache:@var{URL}
 
 Physical concatenation protocol.
 
-Allow to read and seek from many resource in sequence as if they were
+Read and seek from many resources in sequence as if they were
 a unique resource.
 
 A URL accepted by this protocol has the syntax:
@@ -117,7 +129,7 @@ ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP/////////////
 
 File access protocol.
 
-Allow to read from or write to a file.
+Read from or write to a file.
 
 A file URL can have the form:
 @example
@@ -155,7 +167,7 @@ time, which is valuable for files on slow medium.
 
 FTP (File Transfer Protocol).
 
-Allow to read from or write to remote resources using FTP protocol.
+Read from or write to remote resources using FTP protocol.
 
 Following syntax is required.
 @example
@@ -255,7 +267,7 @@ Export the MIME type.
 If set to 1 request ICY (SHOUTcast) metadata from the server. If the server
 supports this, the metadata has to be retrieved by the application by reading
 the @option{icy_metadata_headers} and @option{icy_metadata_packet} options.
-The default is 0.
+The default is 1.
 
 @item icy_metadata_headers
 If the server supports ICY metadata, this contains the ICY-specific HTTP reply
@@ -277,6 +289,43 @@ Set initial byte offset.
 
 @item end_offset
 Try to limit the request to bytes preceding this offset.
+
+@item method
+When used as a client option it sets the HTTP method for the request.
+
+When used as a server option it sets the HTTP method that is going to be
+expected from the client(s).
+If the expected and the received HTTP method do not match the client will
+be given a Bad Request response.
+When unset the HTTP method is not checked for now. This will be replaced by
+autodetection in the future.
+
+@item listen
+If set to 1 enables experimental HTTP server. This can be used to send data when
+used as an output option, or read data from a client with HTTP POST when used as
+an input option.
+If set to 2 enables experimental mutli-client HTTP server. This is not yet implemented
+in ffmpeg.c or ffserver.c and thus must not be used as a command line option.
+@example
+# Server side (sending):
+ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://@var{server}:@var{port}
+
+# Client side (receiving):
+ffmpeg -i http://@var{server}:@var{port} -c copy somefile.ogg
+
+# Client can also be done with wget:
+wget http://@var{server}:@var{port} -O somefile.ogg
+
+# Server side (receiving):
+ffmpeg -listen 1 -i http://@var{server}:@var{port} -c copy somefile.ogg
+
+# Client side (sending):
+ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://@var{server}:@var{port}
+
+# Client can also be done with wget:
+wget --post-file=somefile.ogg http://@var{server}:@var{port}
+@end example
+
 @end table
 
 @subsection HTTP Cookies
@@ -293,6 +342,50 @@ The required syntax to play a stream specifying a cookie is:
 ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
 @end example
 
+@section Icecast
+
+Icecast protocol (stream to Icecast servers)
+
+This protocol accepts the following options:
+
+@table @option
+@item ice_genre
+Set the stream genre.
+
+@item ice_name
+Set the stream name.
+
+@item ice_description
+Set the stream description.
+
+@item ice_url
+Set the stream website URL.
+
+@item ice_public
+Set if the stream should be public.
+The default is 0 (not public).
+
+@item user_agent
+Override the User-Agent header. If not specified a string of the form
+"Lavf/<version>" will be used.
+
+@item password
+Set the Icecast mountpoint password.
+
+@item content_type
+Set the stream content type. This must be set if it is different from
+audio/mpeg.
+
+@item legacy_icecast
+This enables support for Icecast versions < 2.4.0, that do not support the
+HTTP PUT method but the SOURCE method.
+
+@end table
+
+@example
+icecast://[@var{username}[:@var{password}]@@]@var{server}:@var{port}/@var{mountpoint}
+@end example
+
 @section mmst
 
 MMS (Microsoft Media Server) protocol over TCP.
@@ -330,7 +423,7 @@ be seekable, so they will fail with the MD5 output protocol.
 
 UNIX pipe access protocol.
 
-Allow to read and write from UNIX pipes.
+Read and write from UNIX pipes.
 
 The accepted syntax is:
 @example
@@ -539,7 +632,7 @@ firewalls.
 
 @section libsmbclient
 
-libsmbclient permits to manipulate CIFS/SMB network resources.
+libsmbclient permits one to manipulate CIFS/SMB network resources.
 
 Following syntax is required.
 
@@ -570,7 +663,7 @@ For more information see: @url{http://www.samba.org/}.
 
 Secure File Transfer Protocol via libssh
 
-Allow to read from or write to remote resources using SFTP protocol.
+Read from or write to remote resources using SFTP protocol.
 
 Following syntax is required.
 
@@ -706,7 +799,7 @@ port will be used for the local RTP and RTCP ports.
 
 @item
 If @option{localrtcpport} (the local RTCP port) is not set it will be
-set to the the local RTP port value plus 1.
+set to the local RTP port value plus 1.
 @end enumerate
 
 @section rtsp
@@ -1011,7 +1104,9 @@ subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
 @end example
 
 Play an AVI file directly from a TAR archive:
+@example
 subfile,,start,183241728,end,366490624,,:archive.tar
+@end example
 
 @section tcp
 
@@ -1037,8 +1132,8 @@ Set raise error timeout, expressed in microseconds.
 This option is only relevant in read mode: if no data arrived in more
 than this time interval, raise error.
 
-@item listen_timeout=@var{microseconds}
-Set listen timeout, expressed in microseconds.
+@item listen_timeout=@var{milliseconds}
+Set listen timeout, expressed in milliseconds.
 @end table
 
 The following example shows how to setup a listening TCP connection

doc/scaler.texi

@@ -122,6 +122,22 @@ a_dither).
 
 @end table
 
+@item alphablend
+Set the alpha blending to use when the input has alpha but the output does not.
+Default value is @samp{none}.
+
+@table @samp
+@item uniform_color
+Blend onto a uniform background color
+
+@item checkerboard
+Blend onto a checkerboard
+
+@item none
+No blending
+
+@end table
+
 @end table
 
 @c man end SCALER OPTIONS

doc/snow.txt

@@ -618,7 +618,6 @@ flip wavelet?
 try to use the wavelet transformed predicted image (motion compensated image) as context for coding the residual coefficients
 try the MV length as context for coding the residual coefficients
 use extradata for stuff which is in the keyframes now?
-the MV median predictor is patented IIRC
 implement per picture halfpel interpolation
 try different range coder state transition tables for different contexts